Merge "Merge upstream SHA 04cb763"
am: 9a337512d9
* commit '9a337512d97e37afc142dee4fd50a41b741a87d2': (797 commits)
Add tests for verifying transport feedback for audio and video.
Eliminate defines in talk/
Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
Remove assert which was incorrectly added to TcpPort::OnSentPacket.
Reland Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
Update with new default boringssl no-aes cipher suites. Re-enable tests.
Revert of Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket. (patchset #3 id:40001 of https://codereview.webrtc.org/1577873003/ )
Re-land: "Use an explicit identifier in Config"
Connect TurnPort and TCPPort to AsyncPacketSocket::SignalSentPacket.
Revert of Delete remnants of non-square pixel support from cricket::VideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/1586613002/ )
Remove libfuzzer trybot from default trybot set.
Add ramp-up tests for transport sequence number with and w/o audio.
Delete remnants of non-square pixel support from cricket::VideoFrame.
Fix IPAddress::ToSensitiveString() to avoid dependency on inet_ntop().
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
Re-enable tests that failed under Linux_Msan.
Revert of Use an explicit identifier in Config (patchset #4 id:60001 of https://codereview.webrtc.org/1538643004/ )
Roll chromium_revision 346fea9..099be58 (369082:369139)
Disable WebRtcVideoChannel2BaseTest.SendManyResizeOnce for TSan
Add build_protobuf variable.
...
diff --git a/.gitignore b/.gitignore
index b29171c..7090f4a 100644
--- a/.gitignore
+++ b/.gitignore
@@ -44,11 +44,9 @@
/chromium/.last_sync_chromium
/chromium/_bad_scm
/chromium/src
-/google_apis
/gyp-mac-tool
/links
/links.db
-/net
/out
/resources/**/*.aecdump
/resources/**/*.bin
@@ -79,7 +77,6 @@
/testing
/third_party/WebKit/Tools/Scripts
/third_party/android_platform
-/third_party/android_testrunner
/third_party/android_tools
/third_party/appurify-python
/third_party/asan
@@ -87,6 +84,7 @@
/third_party/binutils
/third_party/boringssl
/third_party/BUILD.gn
+/third_party/catapult
/third_party/clang_format
/third_party/class-dump
/third_party/colorama
@@ -94,6 +92,7 @@
/third_party/directxsdk
/third_party/drmemory
/third_party/expat
+/third_party/ffmpeg
/third_party/gaeunit
/third_party/gflags/src
/third_party/google-visualization-python
@@ -105,6 +104,7 @@
/third_party/junit
/third_party/junit-jar
/third_party/libc++
+/third_party/libc++-static
/third_party/libc++abi
/third_party/libevent
/third_party/libjingle
@@ -123,6 +123,7 @@
/third_party/nss
/third_party/oauth2
/third_party/ocmock
+/third_party/openh264
/third_party/openmax_dl
/third_party/opus
/third_party/proguard
@@ -139,7 +140,6 @@
/third_party/zlib
/tools/android
/tools/clang
-/tools/find_depot_tools.py
/tools/generate_library_loader
/tools/gn
/tools/grit
@@ -148,21 +148,15 @@
/tools/memory
/tools/protoc_wrapper
/tools/python
-/tools/relocation_packer
/tools/sanitizer_options
/tools/swarming_client
+/tools/telemetry
/tools/tsan_suppressions
/tools/valgrind
/tools/vim
/tools/win
/tools/xdisplaycheck
/tools/whitespace.txt
-/webrtc/examples/android/media_demo/bin
-/webrtc/examples/android/media_demo/gen
-/webrtc/examples/android/media_demo/libs
-/webrtc/examples/android/media_demo/local.properties
-/webrtc/examples/android/media_demo/obj
-/webrtc/examples/android/media_demo/proguard-project.txt
/webrtc/examples/android/opensl_loopback/bin
/webrtc/examples/android/opensl_loopback/gen
/webrtc/examples/android/opensl_loopback/libs
@@ -172,9 +166,5 @@
/webrtc/modules/audio_device/android/test/bin/
/webrtc/modules/audio_device/android/test/gen/
/webrtc/modules/audio_device/android/test/libs/
-/webrtc/video_engine/test/android/bin
-/webrtc/video_engine/test/android/gen
-/webrtc/video_engine/test/android/libs
-/webrtc/video_engine/test/android/obj
/x86-generic_out/
/xcodebuild
diff --git a/.gn b/.gn
index d078116..f849ef7 100644
--- a/.gn
+++ b/.gn
@@ -35,6 +35,7 @@
"//build/config/linux/pkg_config.gni",
"//build/config/mac/mac_sdk.gni",
"//build/config/posix/BUILD.gn",
+ "//build/config/sysroot.gni",
"//build/config/win/visual_studio_version.gni",
"//build/gn_helpers.py",
"//build/gypi_to_gn.py",
@@ -42,6 +43,7 @@
"//build/toolchain/mac/BUILD.gn",
"//build/toolchain/win/BUILD.gn",
"//third_party/boringssl/BUILD.gn",
+ "//third_party/openh264/BUILD.gn",
"//third_party/opus/BUILD.gn",
"//webrtc/modules/video_render/BUILD.gn",
]
diff --git a/AUTHORS b/AUTHORS
index 0d9ff5a..6f11ee6 100644
--- a/AUTHORS
+++ b/AUTHORS
@@ -1,6 +1,7 @@
# Names should be added to this file like so:
# Name or Organization <email address>
+Andrew MacDonald <andrew@webrtc.org>
Anil Kumar <an1kumar@gmail.com>
Ben Strong <bstrong@gmail.com>
Bob Withers <bwit@pobox.com>
@@ -22,6 +23,7 @@
Paul Kapustin <pkapustin@gmail.com>
Rafael Lopez Diez <rafalopezdiez@gmail.com>
Ralph Giles <giles@ghostscript.com>
+Riku Voipio <riku.voipio@linaro.org>
Robert Nagy <robert.nagy@gmail.com>
Ryan Yoakum <ryoakum@skobalt.com>
Sarah Thompson <sarah@telergy.com>
@@ -30,8 +32,10 @@
Steve Reid <sreid@sea-to-sky.net>
Vicken Simonian <vsimon@gmail.com>
Victor Costan <costan@gmail.com>
+Alexander Brauckmann <a.brauckmann@gmail.com>
&yet LLC
+Agora IO
ARM Holdings
BroadSoft Inc.
Google Inc.
diff --git a/DEPS b/DEPS
index 3ceff79..dcf56bc 100644
--- a/DEPS
+++ b/DEPS
@@ -6,7 +6,7 @@
vars = {
'extra_gyp_flag': '-Dextra_gyp_flag=0',
'chromium_git': 'https://chromium.googlesource.com',
- 'chromium_revision': '657e8d9a9139da0735d73fc0fc6d90a05f980319',
+ 'chromium_revision': '099be58b08dadb64b1dc9f359ae097e978df5416',
}
# NOTE: Prefer revision numbers to tags for svn deps. Use http rather than
@@ -24,7 +24,7 @@
deps_os = {
'win': {
'src/third_party/winsdk_samples/src':
- Var('chromium_git') + '/external/webrtc/deps/third_party/winsdk_samples_v71@c0cbedd854cb610a53226d9817416c4ab9a7d1e9', # from svn revision 7951
+ Var('chromium_git') + '/external/webrtc/deps/third_party/winsdk_samples_v71@e71b549167a665d7424d6f1dadfbff4b4aad1589',
},
}
@@ -102,6 +102,7 @@
'--recursive',
'--num_threads=10',
'--no_auth',
+ '--quiet',
'--bucket', 'chromium-webrtc-resources',
'src/resources'],
},
diff --git a/OWNERS b/OWNERS
index 3ecf2a2..5812db5 100644
--- a/OWNERS
+++ b/OWNERS
@@ -1,4 +1,3 @@
-andrew@webrtc.org
henrika@webrtc.org
mflodman@webrtc.org
niklas.enbom@webrtc.org
diff --git a/PRESUBMIT.py b/PRESUBMIT.py
index e7ceac9..08dd68d 100755
--- a/PRESUBMIT.py
+++ b/PRESUBMIT.py
@@ -14,6 +14,100 @@
import sys
+# Directories that will be scanned by cpplint by the presubmit script.
+CPPLINT_DIRS = [
+ 'webrtc/audio',
+ 'webrtc/call',
+ 'webrtc/common_video',
+ 'webrtc/examples',
+ 'webrtc/modules/remote_bitrate_estimator',
+ 'webrtc/modules/rtp_rtcp',
+ 'webrtc/modules/video_coding',
+ 'webrtc/modules/video_processing',
+ 'webrtc/sound',
+ 'webrtc/tools',
+ 'webrtc/video',
+]
+
+# List of directories of "supported" native APIs. That means changes to headers
+# will be done in a compatible way following this scheme:
+# 1. Non-breaking changes are made.
+# 2. The old APIs as marked as deprecated (with comments).
+# 3. Deprecation is announced to discuss-webrtc@googlegroups.com and
+# webrtc-users@google.com (internal list).
+# 4. (later) The deprecated APIs are removed.
+# Directories marked as DEPRECATED should not be used. They're only present in
+# the list to support legacy downstream code.
+NATIVE_API_DIRS = (
+ 'talk/app/webrtc',
+ 'webrtc',
+ 'webrtc/base', # DEPRECATED.
+ 'webrtc/common_audio/include', # DEPRECATED.
+ 'webrtc/modules/audio_coding/include',
+ 'webrtc/modules/audio_conference_mixer/include', # DEPRECATED.
+ 'webrtc/modules/audio_device/include',
+ 'webrtc/modules/audio_processing/include',
+ 'webrtc/modules/bitrate_controller/include',
+ 'webrtc/modules/include',
+ 'webrtc/modules/remote_bitrate_estimator/include',
+ 'webrtc/modules/rtp_rtcp/include',
+ 'webrtc/modules/rtp_rtcp/source', # DEPRECATED.
+ 'webrtc/modules/utility/include',
+ 'webrtc/modules/video_coding/codecs/h264/include',
+ 'webrtc/modules/video_coding/codecs/i420/include',
+ 'webrtc/modules/video_coding/codecs/vp8/include',
+ 'webrtc/modules/video_coding/codecs/vp9/include',
+ 'webrtc/modules/video_coding/include',
+ 'webrtc/system_wrappers/include', # DEPRECATED.
+ 'webrtc/voice_engine/include',
+)
+
+
+def _VerifyNativeApiHeadersListIsValid(input_api, output_api):
+ """Ensures the list of native API header directories is up to date."""
+ non_existing_paths = []
+ native_api_full_paths = [
+ input_api.os_path.join(input_api.PresubmitLocalPath(),
+ *path.split('/')) for path in NATIVE_API_DIRS]
+ for path in native_api_full_paths:
+ if not os.path.isdir(path):
+ non_existing_paths.append(path)
+ if non_existing_paths:
+ return [output_api.PresubmitError(
+ 'Directories to native API headers have changed which has made the '
+ 'list in PRESUBMIT.py outdated.\nPlease update it to the current '
+ 'location of our native APIs.',
+ non_existing_paths)]
+ return []
+
+
+def _CheckNativeApiHeaderChanges(input_api, output_api):
+ """Checks to remind proper changing of native APIs."""
+ files = []
+ for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
+ if f.LocalPath().endswith('.h'):
+ for path in NATIVE_API_DIRS:
+ if os.path.dirname(f.LocalPath()) == path:
+ files.append(f)
+
+ if files:
+ return [output_api.PresubmitNotifyResult(
+ 'You seem to be changing native API header files. Please make sure '
+ 'you:\n'
+ ' 1. Make compatible changes that don\'t break existing clients.\n'
+ ' 2. Mark the old APIs as deprecated.\n'
+ ' 3. Create a timeline and plan for when the deprecated method will '
+ 'be removed (preferably 3 months or so).\n'
+ ' 4. Update/inform existing downstream code owners to stop using the '
+ 'deprecated APIs: \n'
+ 'send announcement to discuss-webrtc@googlegroups.com and '
+ 'webrtc-users@google.com.\n'
+ ' 5. (after ~3 months) remove the deprecated API.\n'
+ 'Related files:',
+ files)]
+ return []
+
+
def _CheckNoIOStreamInHeaders(input_api, output_api):
"""Checks to make sure no .h files include <iostream>."""
files = []
@@ -54,6 +148,14 @@
'use FRIEND_TEST_ALL_PREFIXES() instead.\n' + '\n'.join(problems))]
+def _IsLintWhitelisted(whitelist_dirs, file_path):
+ """ Checks if a file is whitelisted for lint check."""
+ for path in whitelist_dirs:
+ if os.path.dirname(file_path).startswith(path):
+ return True
+ return False
+
+
def _CheckApprovedFilesLintClean(input_api, output_api,
source_file_filter=None):
"""Checks that all new or whitelisted .cc and .h files pass cpplint.py.
@@ -68,12 +170,9 @@
# pylint: disable=W0212
cpplint._cpplint_state.ResetErrorCounts()
- # Justifications for each filter:
- #
- # - build/header_guard : WebRTC coding style says they should be prefixed
- # with WEBRTC_, which is not possible to configure in
- # cpplint.py.
- cpplint._SetFilters('-build/header_guard')
+ # Create a platform independent whitelist for the CPPLINT_DIRS.
+ whitelist_dirs = [input_api.os_path.join(*path.split('/'))
+ for path in CPPLINT_DIRS]
# Use the strictest verbosity level for cpplint.py (level 1) which is the
# default when running cpplint.py from command line.
@@ -83,7 +182,7 @@
files = []
for f in input_api.AffectedSourceFiles(source_file_filter):
# Note that moved/renamed files also count as added.
- if f.Action() == 'A':
+ if f.Action() == 'A' or _IsLintWhitelisted(whitelist_dirs, f.LocalPath()):
files.append(f.AbsoluteLocalPath())
for file_name in files:
@@ -256,6 +355,14 @@
def _CommonChecks(input_api, output_api):
"""Checks common to both upload and commit."""
results = []
+ # Filter out files that are in objc or ios dirs from being cpplint-ed since
+ # they do not follow C++ lint rules.
+ black_list = input_api.DEFAULT_BLACK_LIST + (
+ r".*\bobjc[\\\/].*",
+ )
+ source_file_filter = lambda x: input_api.FilterSourceFile(x, None, black_list)
+ results.extend(_CheckApprovedFilesLintClean(
+ input_api, output_api, source_file_filter))
results.extend(input_api.canned_checks.RunPylint(input_api, output_api,
black_list=(r'^.*gviz_api\.py$',
r'^.*gaeunit\.py$',
@@ -305,7 +412,7 @@
input_api, output_api))
results.extend(input_api.canned_checks.CheckChangeTodoHasOwner(
input_api, output_api))
- results.extend(_CheckApprovedFilesLintClean(input_api, output_api))
+ results.extend(_CheckNativeApiHeaderChanges(input_api, output_api))
results.extend(_CheckNoIOStreamInHeaders(input_api, output_api))
results.extend(_CheckNoFRIEND_TEST(input_api, output_api))
results.extend(_CheckGypChanges(input_api, output_api))
@@ -325,6 +432,7 @@
def CheckChangeOnCommit(input_api, output_api):
results = []
results.extend(_CommonChecks(input_api, output_api))
+ results.extend(_VerifyNativeApiHeadersListIsValid(input_api, output_api))
results.extend(input_api.canned_checks.CheckOwners(input_api, output_api))
results.extend(input_api.canned_checks.CheckChangeWasUploaded(
input_api, output_api))
diff --git a/WATCHLISTS b/WATCHLISTS
index c89a28e..ba7af53 100644
--- a/WATCHLISTS
+++ b/WATCHLISTS
@@ -16,14 +16,14 @@
},
'all_webrtc': {
# NOTE: if you like this you might like webrtc-reviews@webrtc.org!
- 'filepath': 'webrtc/.*',
+ 'filepath': '^webrtc/.*',
},
'root_files': {
# webrtc/build/ and non-recursive contents of ./ and webrtc/
- 'filepath': '^[^/]*$|webrtc/[^/]*$|webrtc/build/.*',
+ 'filepath': '^[^/]*$|^webrtc/[^/]*$|^webrtc/build/.*',
},
'documented_interfaces': {
- 'filepath': 'webrtc/[^/]*\.h$|'\
+ 'filepath': '^webrtc/[^/]*\.h$|'\
'webrtc/voice_engine/include/.*',
},
'build_files': {
@@ -32,9 +32,14 @@
'java_files': {
'filepath': '\.java$|\.xml$',
},
- 'video_engine': {
- 'filepath': 'webrtc/video_engine/.*|'\
- 'webrtc/video/.*',
+ 'audio': {
+ 'filepath': 'webrtc/audio/.*',
+ },
+ 'call': {
+ 'filepath': 'webrtc/call/.*',
+ },
+ 'video': {
+ 'filepath': 'webrtc/video/.*',
},
'voice_engine': {
'filepath': 'webrtc/voice_engine/.*',
@@ -89,59 +94,93 @@
'WATCHLISTS': {
'this_file': [''],
'all_webrtc': ['tterriberry@mozilla.com'],
- 'root_files': ['andrew@webrtc.org',
- 'niklas.enbom@webrtc.org',
- 'yujie.mao@webrtc.org',
+ 'root_files': ['niklas.enbom@webrtc.org',
+ 'peah@webrtc.org',
'qiang.lu@intel.com',
- 'peah@webrtc.org'],
+ 'yujie.mao@webrtc.org'],
'documented_interfaces': ['interface-changes@webrtc.org',
'rwolff@gocast.it'],
'common_audio': ['aluebs@webrtc.org',
'andrew@webrtc.org',
+ 'audio-team@agora.io',
'bjornv@webrtc.org',
+ 'minyue@webrtc.org',
'peah@webrtc.org'],
- 'video_engine': ['andresp@webrtc.org',
- 'mflodman@webrtc.org',
- 'perkj@webrtc.org',
- 'stefan@webrtc.org',
- 'yujie.mao@webrtc.org',
- 'solenberg@webrtc.org'],
- 'voice_engine': ['henrika@webrtc.org',
+ 'audio': ['solenberg@webrtc.org',
+ 'tina.legrand@webrtc.org'],
+ 'call': ['mflodman@webrtc.org',
+ 'pbos@webrtc.org',
+ 'solenberg@webrtc.org',
+ 'stefan@webrtc.org'],
+ 'video': ['andresp@webrtc.org',
+ 'mflodman@webrtc.org',
+ 'pbos@webrtc.org',
+ 'perkj@webrtc.org',
+ 'solenberg@webrtc.org',
+ 'stefan@webrtc.org',
+ 'video-team@agora.io',
+ 'yujie.mao@webrtc.org',
+ 'zhengzhonghou@agora.io'],
+ 'voice_engine': ['andrew@webrtc.org',
+ 'audio-team@agora.io',
+ 'henrika@webrtc.org',
'henrik.lundin@webrtc.org',
- 'solenberg@webrtc.org',
- 'peah@webrtc.org'],
+ 'minyue@webrtc.org',
+ 'peah@webrtc.org',
+ 'solenberg@webrtc.org'],
'video_capture': ['mflodman@webrtc.org',
- 'perkj@webrtc.org'],
+ 'perkj@webrtc.org',
+ 'sdk-team@agora.io',
+ 'zhengzhonghou@agora.io'],
'video_render': ['mflodman@webrtc.org',
- 'perkj@webrtc.org'],
- 'audio_device': ['henrika@webrtc.org',
- 'peah@webrtc.org'],
- 'audio_coding': ['tina.legrand@webrtc.org',
+ 'perkj@webrtc.org',
+ 'sdk-team@agora.io',
+ 'zhengzhonghou@agora.io'],
+ 'audio_device': ['audio-team@agora.io',
+ 'henrika@webrtc.org',
+ 'peah@webrtc.org',
+ 'sdk-team@agora.io'],
+ 'audio_coding': ['audio-team@agora.io',
'henrik.lundin@webrtc.org',
'kwiberg@webrtc.org',
- 'peah@webrtc.org'],
- 'neteq': ['henrik.lundin@webrtc.org'],
+ 'minyue@webrtc.org',
+ 'peah@webrtc.org',
+ 'tina.legrand@webrtc.org'],
+ 'neteq': ['audio-team@agora.io',
+ 'henrik.lundin@webrtc.org',
+ 'minyue@webrtc.org'],
'audio_processing': ['aluebs@webrtc.org',
'andrew@webrtc.org',
+ 'audio-team@agora.io',
'bjornv@webrtc.org',
- 'kwiberg@webrtc.org',
'henrik.lundin@webrtc.org',
+ 'kwiberg@webrtc.org',
+ 'minyue@webrtc.org',
'peah@webrtc.org',
'solenberg@webrtc.org'],
- 'video_coding': ['stefan@webrtc.org',
- 'mflodman@webrtc.org'],
- 'video_processing': ['stefan@webrtc.org'],
- 'bitrate_controller': ['stefan@webrtc.org',
- 'mflodman@webrtc.org'],
- 'remote_bitrate_estimator': ['stefan@webrtc.org',
- 'mflodman@webrtc.org'],
- 'pacing': ['stefan@webrtc.org',
- 'mflodman@webrtc.org'],
+ 'video_coding': ['mflodman@webrtc.org',
+ 'stefan@webrtc.org',
+ 'video-team@agora.io',
+ 'zhengzhonghou@agora.io'],
+ 'video_processing': ['stefan@webrtc.org',
+ 'video-team@agora.io',
+ 'zhengzhonghou@agora.io'],
+ 'bitrate_controller': ['mflodman@webrtc.org',
+ 'stefan@webrtc.org',
+ 'zhuangzesen@agora.io'],
+ 'remote_bitrate_estimator': ['mflodman@webrtc.org',
+ 'stefan@webrtc.org',
+ 'zhuangzesen@agora.io'],
+ 'pacing': ['mflodman@webrtc.org',
+ 'stefan@webrtc.org',
+ 'zhuangzesen@agora.io'],
'rtp_rtcp': ['mflodman@webrtc.org',
- 'stefan@webrtc.org'],
- 'system_wrappers': ['mflodman@webrtc.org',
+ 'stefan@webrtc.org',
+ 'zhuangzesen@agora.io'],
+ 'system_wrappers': ['fengyue@agora.io',
'henrika@webrtc.org',
- 'andrew@webrtc.org',
- 'peah@webrtc.org'],
+ 'mflodman@webrtc.org',
+ 'peah@webrtc.org',
+ 'zhengzhonghou@agora.io'],
},
}
diff --git a/all.gyp b/all.gyp
index 40dbc13..0b11c8f 100644
--- a/all.gyp
+++ b/all.gyp
@@ -24,7 +24,6 @@
'conditions': [
['include_examples==1', {
'dependencies': [
- 'webrtc/libjingle_examples.gyp:*',
'webrtc/webrtc_examples.gyp:*',
],
}],
diff --git a/chromium/.gclient b/chromium/.gclient
index 7f46333..9d5dfcf 100644
--- a/chromium/.gclient
+++ b/chromium/.gclient
@@ -10,7 +10,6 @@
'src/chrome/tools/test/reference_build/chrome_win': None,
'src/native_client': None,
'src/third_party/cld_2/src': None,
- 'src/third_party/ffmpeg': None,
'src/third_party/hunspell_dictionaries': None,
'src/third_party/liblouis/src': None,
'src/third_party/pdfium': None,
diff --git a/infra/config/cq.cfg b/infra/config/cq.cfg
index 80f6f60..4c49280 100644
--- a/infra/config/cq.cfg
+++ b/infra/config/cq.cfg
@@ -30,6 +30,8 @@
builders { name: "android_rel" }
builders { name: "android_arm64_rel" }
builders { name: "android_clang_dbg" }
+ builders { name: "android_compile_x86_dbg" }
+ builders { name: "android_compile_x64_dbg" }
builders { name: "android_gn_dbg" }
builders { name: "android_gn_rel" }
builders { name: "ios_arm64_dbg" }
@@ -38,27 +40,34 @@
builders { name: "ios_rel" }
builders { name: "ios32_sim_dbg" }
builders { name: "ios64_sim_dbg" }
- builders { name: "linux_compile_dbg" }
builders { name: "linux_asan" }
+ builders { name: "linux_baremetal" }
+ builders { name: "linux_compile_dbg" }
builders { name: "linux_gn_dbg" }
builders { name: "linux_gn_rel" }
+ # Disabled, see http://crbug.com/577566 for details.
+ #builders { name: "linux_libfuzzer_rel" }
builders { name: "linux_msan" }
builders { name: "linux_rel" }
builders { name: "linux_tsan2" }
- builders { name: "mac_compile_dbg" }
- builders { name: "mac_compile_x64_dbg" }
builders { name: "mac_asan" }
+ builders { name: "mac_baremetal" }
+ builders { name: "mac_compile_dbg" }
builders { name: "mac_rel" }
- builders { name: "mac_x64_gn_dbg" }
- builders { name: "mac_x64_gn_rel" }
- builders { name: "mac_x64_rel" }
+ builders { name: "mac_gn_dbg" }
+ builders { name: "mac_gn_rel" }
builders { name: "presubmit" }
+ builders { name: "win_baremetal" }
+ builders { name: "win_clang_dbg" }
+ builders { name: "win_clang_rel" }
builders { name: "win_compile_dbg" }
+ builders { name: "win_drmemory_light" }
builders { name: "win_rel" }
+ builders { name: "win_x64_clang_dbg" }
+ builders { name: "win_x64_clang_rel" }
builders { name: "win_x64_gn_dbg" }
builders { name: "win_x64_gn_rel" }
builders { name: "win_x64_rel" }
- builders { name: "win_drmemory_light" }
}
}
}
diff --git a/resources/audio_coding/neteq4_network_stats.dat.sha1 b/resources/audio_coding/neteq4_network_stats.dat.sha1
index 72a9499..f51a02a 100644
--- a/resources/audio_coding/neteq4_network_stats.dat.sha1
+++ b/resources/audio_coding/neteq4_network_stats.dat.sha1
@@ -1 +1 @@
-e5e2d0ff26d16339cf0f37a3512bfa2d390a9a9a
\ No newline at end of file
+2cf380a05ee07080bd72471e8ec7777a39644ec9
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_opus_network_stats.dat.sha1 b/resources/audio_coding/neteq4_opus_network_stats.dat.sha1
new file mode 100644
index 0000000..6a9e7ee
--- /dev/null
+++ b/resources/audio_coding/neteq4_opus_network_stats.dat.sha1
@@ -0,0 +1 @@
+cc9fa62d0a8f46ffebc782aea2610dda67bb5558
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_opus_ref.pcm.sha1 b/resources/audio_coding/neteq4_opus_ref.pcm.sha1
new file mode 100644
index 0000000..5cecc50
--- /dev/null
+++ b/resources/audio_coding/neteq4_opus_ref.pcm.sha1
@@ -0,0 +1 @@
+301895f1aaa9cd9eae0f5d04d179d63491d744cc
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_opus_ref_win_32.pcm.sha1 b/resources/audio_coding/neteq4_opus_ref_win_32.pcm.sha1
new file mode 100644
index 0000000..b7cf990
--- /dev/null
+++ b/resources/audio_coding/neteq4_opus_ref_win_32.pcm.sha1
@@ -0,0 +1 @@
+fbad99878c7a26958e755190027c976692708334
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_opus_ref_win_64.pcm.sha1 b/resources/audio_coding/neteq4_opus_ref_win_64.pcm.sha1
new file mode 100644
index 0000000..b7cf990
--- /dev/null
+++ b/resources/audio_coding/neteq4_opus_ref_win_64.pcm.sha1
@@ -0,0 +1 @@
+fbad99878c7a26958e755190027c976692708334
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_opus_rtcp_stats.dat.sha1 b/resources/audio_coding/neteq4_opus_rtcp_stats.dat.sha1
new file mode 100644
index 0000000..05570b8
--- /dev/null
+++ b/resources/audio_coding/neteq4_opus_rtcp_stats.dat.sha1
@@ -0,0 +1 @@
+e37c797e3de6a64dda88c9ade7a013d022a2e1e0
\ No newline at end of file
diff --git a/resources/audio_coding/neteq4_rtcp_stats.dat.sha1 b/resources/audio_coding/neteq4_rtcp_stats.dat.sha1
index ae63c76..1fa337b 100644
--- a/resources/audio_coding/neteq4_rtcp_stats.dat.sha1
+++ b/resources/audio_coding/neteq4_rtcp_stats.dat.sha1
@@ -1 +1 @@
-948753a2087fbb5b74a3ea0b1aef8593c9c30b10
\ No newline at end of file
+b8880bf9fed2487efbddcb8d94b9937a29ae521d
\ No newline at end of file
diff --git a/resources/audio_coding/neteq_network_stats.dat.sha1 b/resources/audio_coding/neteq_network_stats.dat.sha1
deleted file mode 100644
index e02ad84..0000000
--- a/resources/audio_coding/neteq_network_stats.dat.sha1
+++ /dev/null
@@ -1 +0,0 @@
-e3c189b500d92fd0f10cb4c770c298cee7008749
\ No newline at end of file
diff --git a/resources/audio_coding/neteq_network_stats_win_32.dat.sha1 b/resources/audio_coding/neteq_network_stats_win_32.dat.sha1
deleted file mode 100644
index a197fc1..0000000
--- a/resources/audio_coding/neteq_network_stats_win_32.dat.sha1
+++ /dev/null
@@ -1 +0,0 @@
-343061419a64ca99323fc0d25a43149a5d40cf07
\ No newline at end of file
diff --git a/resources/audio_coding/neteq_opus.rtp.sha1 b/resources/audio_coding/neteq_opus.rtp.sha1
new file mode 100644
index 0000000..ff5b8fe
--- /dev/null
+++ b/resources/audio_coding/neteq_opus.rtp.sha1
@@ -0,0 +1 @@
+21c8f8aaf9518a629d6c6def87fe6ea1305d5c91
\ No newline at end of file
diff --git a/resources/audio_coding/neteq_rtcp_stats.dat.sha1 b/resources/audio_coding/neteq_rtcp_stats.dat.sha1
deleted file mode 100644
index 42d37ac..0000000
--- a/resources/audio_coding/neteq_rtcp_stats.dat.sha1
+++ /dev/null
@@ -1 +0,0 @@
-759d57e87517e0290144add9ba53d5c6dec1c27e
\ No newline at end of file
diff --git a/resources/audio_coding/neteq_universal_ref.pcm.sha1 b/resources/audio_coding/neteq_universal_ref.pcm.sha1
deleted file mode 100644
index 95e9de7..0000000
--- a/resources/audio_coding/neteq_universal_ref.pcm.sha1
+++ /dev/null
@@ -1 +0,0 @@
-236da353f05d329f6c83e441b80dbfcc18706cfb
\ No newline at end of file
diff --git a/resources/audio_coding/neteq_universal_ref_win_32.pcm.sha1 b/resources/audio_coding/neteq_universal_ref_win_32.pcm.sha1
deleted file mode 100644
index 47c90cf..0000000
--- a/resources/audio_coding/neteq_universal_ref_win_32.pcm.sha1
+++ /dev/null
@@ -1 +0,0 @@
-1d2d353be4345d30506866ca32fa72825c6d65b8
\ No newline at end of file
diff --git a/setup_links.py b/setup_links.py
index 9aeb1e8..492c38b 100755
--- a/setup_links.py
+++ b/setup_links.py
@@ -34,17 +34,16 @@
DIRECTORIES = [
'build',
'buildtools',
- 'google_apis', # Needed by build/common.gypi.
- 'net',
'testing',
'third_party/binutils',
'third_party/boringssl',
'third_party/colorama',
'third_party/drmemory',
'third_party/expat',
- 'third_party/icu',
+ 'third_party/ffmpeg',
'third_party/instrumented_libraries',
'third_party/jsoncpp',
+ 'third_party/libc++-static',
'third_party/libjpeg',
'third_party/libjpeg_turbo',
'third_party/libsrtp',
@@ -55,6 +54,7 @@
'third_party/lss',
'third_party/nss',
'third_party/ocmock',
+ 'third_party/openh264',
'third_party/openmax_dl',
'third_party/opus',
'third_party/proguard',
@@ -84,10 +84,11 @@
DIRECTORIES += [
'base',
'third_party/android_platform',
- 'third_party/android_testrunner',
'third_party/android_tools',
'third_party/appurify-python',
'third_party/ashmem',
+ 'third_party/catapult',
+ 'third_party/icu',
'third_party/ijar',
'third_party/jsr-305',
'third_party/junit',
@@ -99,13 +100,12 @@
'third_party/robolectric',
'tools/android',
'tools/grit',
- 'tools/relocation_packer'
+ 'tools/telemetry',
]
if 'ios' in target_os:
DIRECTORIES.append('third_party/class-dump')
FILES = {
- 'tools/find_depot_tools.py': None,
'tools/isolate_driver.py': None,
'third_party/BUILD.gn': None,
}
diff --git a/sync_chromium.py b/sync_chromium.py
index 442ddcd..b37e0da 100755
--- a/sync_chromium.py
+++ b/sync_chromium.py
@@ -31,7 +31,7 @@
# Bump this whenever the algorithm changes and you need bots/devs to re-sync,
# ignoring the .last_sync_chromium file
-SCRIPT_VERSION = 5
+SCRIPT_VERSION = 7
ROOT_DIR = os.path.dirname(os.path.abspath(__file__))
CHROMIUM_NO_HISTORY = 'CHROMIUM_NO_HISTORY'
diff --git a/talk/app/webrtc/OWNERS b/talk/app/webrtc/OWNERS
index ffd78e1..20a1fdf 100644
--- a/talk/app/webrtc/OWNERS
+++ b/talk/app/webrtc/OWNERS
@@ -1,5 +1,5 @@
glaznev@webrtc.org
-juberti@google.com
-perkj@google.com
+juberti@webrtc.org
+perkj@webrtc.org
tkchin@webrtc.org
-tommi@google.com
+tommi@webrtc.org
diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java b/talk/app/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
index 1c01ffa..63c05fb 100644
--- a/talk/app/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
+++ b/talk/app/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java
@@ -28,7 +28,6 @@
import android.graphics.SurfaceTexture;
import android.opengl.GLES20;
-import android.opengl.Matrix;
import android.test.ActivityTestCase;
import android.test.suitebuilder.annotation.MediumTest;
import android.test.suitebuilder.annotation.SmallTest;
@@ -36,9 +35,6 @@
import java.nio.ByteBuffer;
import java.util.Random;
-import javax.microedition.khronos.egl.EGL10;
-import javax.microedition.khronos.egl.EGLContext;
-
public final class GlRectDrawerTest extends ActivityTestCase {
// Resolution of the test image.
private static final int WIDTH = 16;
@@ -46,7 +42,7 @@
// Seed for random pixel creation.
private static final int SEED = 42;
// When comparing pixels, allow some slack for float arithmetic and integer rounding.
- private static final float MAX_DIFF = 1.0f;
+ private static final float MAX_DIFF = 1.5f;
private static float normalizedByte(byte b) {
return (b & 0xFF) / 255.0f;
@@ -100,7 +96,7 @@
@SmallTest
public void testRgbRendering() {
// Create EGL base with a pixel buffer as display output.
- final EglBase eglBase = new EglBase(EGL10.EGL_NO_CONTEXT, EglBase.ConfigType.PIXEL_BUFFER);
+ final EglBase eglBase = EglBase.create(null, EglBase.CONFIG_PIXEL_BUFFER);
eglBase.createPbufferSurface(WIDTH, HEIGHT);
eglBase.makeCurrent();
@@ -119,7 +115,7 @@
// Draw the RGB frame onto the pixel buffer.
final GlRectDrawer drawer = new GlRectDrawer();
- drawer.drawRgb(rgbTexture, RendererCommon.identityMatrix());
+ drawer.drawRgb(rgbTexture, RendererCommon.identityMatrix(), 0, 0, WIDTH, HEIGHT);
// Download the pixels in the pixel buffer as RGBA. Not all platforms support RGB, e.g. Nexus 9.
final ByteBuffer rgbaData = ByteBuffer.allocateDirect(WIDTH * HEIGHT * 4);
@@ -137,7 +133,7 @@
@SmallTest
public void testYuvRendering() {
// Create EGL base with a pixel buffer as display output.
- EglBase eglBase = new EglBase(EGL10.EGL_NO_CONTEXT, EglBase.ConfigType.PIXEL_BUFFER);
+ EglBase eglBase = EglBase.create(null, EglBase.CONFIG_PIXEL_BUFFER);
eglBase.createPbufferSurface(WIDTH, HEIGHT);
eglBase.makeCurrent();
@@ -166,7 +162,7 @@
// Draw the YUV frame onto the pixel buffer.
final GlRectDrawer drawer = new GlRectDrawer();
- drawer.drawYuv(yuvTextures, RendererCommon.identityMatrix());
+ drawer.drawYuv(yuvTextures, RendererCommon.identityMatrix(), 0, 0, WIDTH, HEIGHT);
// Download the pixels in the pixel buffer as RGBA. Not all platforms support RGB, e.g. Nexus 9.
final ByteBuffer data = ByteBuffer.allocateDirect(WIDTH * HEIGHT * 4);
@@ -231,8 +227,9 @@
private final int rgbTexture;
public StubOesTextureProducer(
- EGLContext sharedContext, SurfaceTexture surfaceTexture, int width, int height) {
- eglBase = new EglBase(sharedContext, EglBase.ConfigType.PLAIN);
+ EglBase.Context sharedContext, SurfaceTexture surfaceTexture, int width,
+ int height) {
+ eglBase = EglBase.create(sharedContext, EglBase.CONFIG_PLAIN);
surfaceTexture.setDefaultBufferSize(width, height);
eglBase.createSurface(surfaceTexture);
assertEquals(eglBase.surfaceWidth(), width);
@@ -253,7 +250,7 @@
GLES20.glTexImage2D(GLES20.GL_TEXTURE_2D, 0, GLES20.GL_RGB, WIDTH,
HEIGHT, 0, GLES20.GL_RGB, GLES20.GL_UNSIGNED_BYTE, rgbPlane);
// Draw the RGB data onto the SurfaceTexture.
- drawer.drawRgb(rgbTexture, RendererCommon.identityMatrix());
+ drawer.drawRgb(rgbTexture, RendererCommon.identityMatrix(), 0, 0, WIDTH, HEIGHT);
eglBase.swapBuffers();
}
@@ -266,14 +263,14 @@
}
// Create EGL base with a pixel buffer as display output.
- final EglBase eglBase = new EglBase(EGL10.EGL_NO_CONTEXT, EglBase.ConfigType.PIXEL_BUFFER);
+ final EglBase eglBase = EglBase.create(null, EglBase.CONFIG_PIXEL_BUFFER);
eglBase.createPbufferSurface(WIDTH, HEIGHT);
// Create resources for generating OES textures.
final SurfaceTextureHelper surfaceTextureHelper =
- SurfaceTextureHelper.create(eglBase.getContext());
+ SurfaceTextureHelper.create(eglBase.getEglBaseContext());
final StubOesTextureProducer oesProducer = new StubOesTextureProducer(
- eglBase.getContext(), surfaceTextureHelper.getSurfaceTexture(), WIDTH, HEIGHT);
+ eglBase.getEglBaseContext(), surfaceTextureHelper.getSurfaceTexture(), WIDTH, HEIGHT);
final SurfaceTextureHelperTest.MockTextureListener listener =
new SurfaceTextureHelperTest.MockTextureListener();
surfaceTextureHelper.setListener(listener);
@@ -291,7 +288,7 @@
// Draw the OES texture on the pixel buffer.
eglBase.makeCurrent();
final GlRectDrawer drawer = new GlRectDrawer();
- drawer.drawOes(listener.oesTextureId, listener.transformMatrix);
+ drawer.drawOes(listener.oesTextureId, listener.transformMatrix, 0, 0, WIDTH, HEIGHT);
// Download the pixels in the pixel buffer as RGBA. Not all platforms support RGB, e.g. Nexus 9.
final ByteBuffer rgbaData = ByteBuffer.allocateDirect(WIDTH * HEIGHT * 4);
diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java b/talk/app/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
new file mode 100644
index 0000000..b1ec5dd
--- /dev/null
+++ b/talk/app/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
@@ -0,0 +1,180 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+package org.webrtc;
+
+import android.annotation.TargetApi;
+import android.opengl.GLES11Ext;
+import android.opengl.GLES20;
+import android.os.Build;
+import android.test.ActivityTestCase;
+import android.test.suitebuilder.annotation.SmallTest;
+import android.util.Log;
+
+import org.webrtc.MediaCodecVideoEncoder.OutputBufferInfo;
+
+import java.nio.ByteBuffer;
+
+@TargetApi(Build.VERSION_CODES.ICE_CREAM_SANDWICH_MR1)
+public final class MediaCodecVideoEncoderTest extends ActivityTestCase {
+ final static String TAG = "MediaCodecVideoEncoderTest";
+
+ @SmallTest
+ public static void testInitializeUsingByteBuffer() {
+ if (!MediaCodecVideoEncoder.isVp8HwSupported()) {
+ Log.i(TAG,
+ "Hardware does not support VP8 encoding, skipping testInitReleaseUsingByteBuffer");
+ return;
+ }
+ MediaCodecVideoEncoder encoder = new MediaCodecVideoEncoder();
+ assertTrue(encoder.initEncode(
+ MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, 640, 480, 300, 30, null));
+ encoder.release();
+ }
+
+ @SmallTest
+ public static void testInitilizeUsingTextures() {
+ if (!MediaCodecVideoEncoder.isVp8HwSupportedUsingTextures()) {
+ Log.i(TAG, "hardware does not support VP8 encoding, skipping testEncoderUsingTextures");
+ return;
+ }
+ EglBase14 eglBase = new EglBase14(null, EglBase.CONFIG_PLAIN);
+ MediaCodecVideoEncoder encoder = new MediaCodecVideoEncoder();
+ assertTrue(encoder.initEncode(
+ MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, 640, 480, 300, 30,
+ eglBase.getEglBaseContext()));
+ encoder.release();
+ eglBase.release();
+ }
+
+ @SmallTest
+ public static void testInitializeUsingByteBufferReInitilizeUsingTextures() {
+ if (!MediaCodecVideoEncoder.isVp8HwSupportedUsingTextures()) {
+ Log.i(TAG, "hardware does not support VP8 encoding, skipping testEncoderUsingTextures");
+ return;
+ }
+ MediaCodecVideoEncoder encoder = new MediaCodecVideoEncoder();
+ assertTrue(encoder.initEncode(
+ MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, 640, 480, 300, 30,
+ null));
+ encoder.release();
+ EglBase14 eglBase = new EglBase14(null, EglBase.CONFIG_PLAIN);
+ assertTrue(encoder.initEncode(
+ MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, 640, 480, 300, 30,
+ eglBase.getEglBaseContext()));
+ encoder.release();
+ eglBase.release();
+ }
+
+ @SmallTest
+ public static void testEncoderUsingByteBuffer() throws InterruptedException {
+ if (!MediaCodecVideoEncoder.isVp8HwSupported()) {
+ Log.i(TAG, "Hardware does not support VP8 encoding, skipping testEncoderUsingByteBuffer");
+ return;
+ }
+
+ final int width = 640;
+ final int height = 480;
+ final int min_size = width * height * 3 / 2;
+ final long presentationTimestampUs = 2;
+
+ MediaCodecVideoEncoder encoder = new MediaCodecVideoEncoder();
+
+ assertTrue(encoder.initEncode(
+ MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, width, height, 300, 30, null));
+ ByteBuffer[] inputBuffers = encoder.getInputBuffers();
+ assertNotNull(inputBuffers);
+ assertTrue(min_size <= inputBuffers[0].capacity());
+
+ int bufferIndex;
+ do {
+ Thread.sleep(10);
+ bufferIndex = encoder.dequeueInputBuffer();
+ } while (bufferIndex == -1); // |-1| is returned when there is no buffer available yet.
+
+ assertTrue(bufferIndex >= 0);
+ assertTrue(bufferIndex < inputBuffers.length);
+ assertTrue(encoder.encodeBuffer(true, bufferIndex, min_size, presentationTimestampUs));
+
+ OutputBufferInfo info;
+ do {
+ info = encoder.dequeueOutputBuffer();
+ Thread.sleep(10);
+ } while (info == null);
+ assertTrue(info.index >= 0);
+ assertEquals(presentationTimestampUs, info.presentationTimestampUs);
+ assertTrue(info.buffer.capacity() > 0);
+ encoder.releaseOutputBuffer(info.index);
+
+ encoder.release();
+ }
+
+ @SmallTest
+ public static void testEncoderUsingTextures() throws InterruptedException {
+ if (!MediaCodecVideoEncoder.isVp8HwSupportedUsingTextures()) {
+ Log.i(TAG, "Hardware does not support VP8 encoding, skipping testEncoderUsingTextures");
+ return;
+ }
+
+ final int width = 640;
+ final int height = 480;
+ final long presentationTs = 2;
+
+ final EglBase14 eglOesBase = new EglBase14(null, EglBase.CONFIG_PIXEL_BUFFER);
+ eglOesBase.createDummyPbufferSurface();
+ eglOesBase.makeCurrent();
+ int oesTextureId = GlUtil.generateTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES);
+
+ // TODO(perkj): This test is week since we don't fill the texture with valid data with correct
+ // width and height and verify the encoded data. Fill the OES texture and figure out a way to
+ // verify that the output make sense.
+
+ MediaCodecVideoEncoder encoder = new MediaCodecVideoEncoder();
+
+ assertTrue(encoder.initEncode(
+ MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, width, height, 300, 30,
+ eglOesBase.getEglBaseContext()));
+ assertTrue(encoder.encodeTexture(true, oesTextureId, RendererCommon.identityMatrix(),
+ presentationTs));
+ GlUtil.checkNoGLES2Error("encodeTexture");
+
+ // It should be Ok to delete the texture after calling encodeTexture.
+ GLES20.glDeleteTextures(1, new int[] {oesTextureId}, 0);
+
+ OutputBufferInfo info = encoder.dequeueOutputBuffer();
+ while (info == null) {
+ info = encoder.dequeueOutputBuffer();
+ Thread.sleep(20);
+ }
+ assertTrue(info.index != -1);
+ assertTrue(info.buffer.capacity() > 0);
+ assertEquals(presentationTs, info.presentationTimestampUs);
+ encoder.releaseOutputBuffer(info.index);
+
+ encoder.release();
+ eglOesBase.release();
+ }
+}
diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java b/talk/app/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
index 882fde1..9e0164d 100644
--- a/talk/app/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
+++ b/talk/app/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
@@ -37,8 +37,6 @@
import java.nio.ByteBuffer;
-import javax.microedition.khronos.egl.EGL10;
-
public final class SurfaceTextureHelperTest extends ActivityTestCase {
/**
* Mock texture listener with blocking wait functionality.
@@ -99,6 +97,14 @@
}
}
+ /** Assert that two integers are close, with difference at most
+ * {@code threshold}. */
+ public static void assertClose(int threshold, int expected, int actual) {
+ if (Math.abs(expected - actual) <= threshold)
+ return;
+ failNotEquals("Not close enough, threshold " + threshold, expected, actual);
+ }
+
/**
* Test normal use by receiving three uniform texture frames. Texture frames are returned as early
* as possible. The texture pixel values are inspected by drawing the texture frame to a pixel
@@ -109,20 +115,21 @@
final int width = 16;
final int height = 16;
// Create EGL base with a pixel buffer as display output.
- final EglBase eglBase = new EglBase(EGL10.EGL_NO_CONTEXT, EglBase.ConfigType.PIXEL_BUFFER);
+ final EglBase eglBase = EglBase.create(null, EglBase.CONFIG_PIXEL_BUFFER);
eglBase.createPbufferSurface(width, height);
final GlRectDrawer drawer = new GlRectDrawer();
// Create SurfaceTextureHelper and listener.
final SurfaceTextureHelper surfaceTextureHelper =
- SurfaceTextureHelper.create(eglBase.getContext());
+ SurfaceTextureHelper.create(eglBase.getEglBaseContext());
final MockTextureListener listener = new MockTextureListener();
surfaceTextureHelper.setListener(listener);
surfaceTextureHelper.getSurfaceTexture().setDefaultBufferSize(width, height);
// Create resources for stubbing an OES texture producer. |eglOesBase| has the SurfaceTexture in
// |surfaceTextureHelper| as the target EGLSurface.
- final EglBase eglOesBase = new EglBase(eglBase.getContext(), EglBase.ConfigType.PLAIN);
+ final EglBase eglOesBase =
+ EglBase.create(eglBase.getEglBaseContext(), EglBase.CONFIG_PLAIN);
eglOesBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
assertEquals(eglOesBase.surfaceWidth(), width);
assertEquals(eglOesBase.surfaceHeight(), height);
@@ -142,7 +149,7 @@
// Wait for an OES texture to arrive and draw it onto the pixel buffer.
listener.waitForNewFrame();
eglBase.makeCurrent();
- drawer.drawOes(listener.oesTextureId, listener.transformMatrix);
+ drawer.drawOes(listener.oesTextureId, listener.transformMatrix, 0, 0, width, height);
surfaceTextureHelper.returnTextureFrame();
@@ -176,19 +183,20 @@
final int width = 16;
final int height = 16;
// Create EGL base with a pixel buffer as display output.
- final EglBase eglBase = new EglBase(EGL10.EGL_NO_CONTEXT, EglBase.ConfigType.PIXEL_BUFFER);
+ final EglBase eglBase = EglBase.create(null, EglBase.CONFIG_PIXEL_BUFFER);
eglBase.createPbufferSurface(width, height);
// Create SurfaceTextureHelper and listener.
final SurfaceTextureHelper surfaceTextureHelper =
- SurfaceTextureHelper.create(eglBase.getContext());
+ SurfaceTextureHelper.create(eglBase.getEglBaseContext());
final MockTextureListener listener = new MockTextureListener();
surfaceTextureHelper.setListener(listener);
surfaceTextureHelper.getSurfaceTexture().setDefaultBufferSize(width, height);
// Create resources for stubbing an OES texture producer. |eglOesBase| has the SurfaceTexture in
// |surfaceTextureHelper| as the target EGLSurface.
- final EglBase eglOesBase = new EglBase(eglBase.getContext(), EglBase.ConfigType.PLAIN);
+ final EglBase eglOesBase =
+ EglBase.create(eglBase.getEglBaseContext(), EglBase.CONFIG_PLAIN);
eglOesBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
assertEquals(eglOesBase.surfaceWidth(), width);
assertEquals(eglOesBase.surfaceHeight(), height);
@@ -212,7 +220,7 @@
// Draw the pending texture frame onto the pixel buffer.
eglBase.makeCurrent();
final GlRectDrawer drawer = new GlRectDrawer();
- drawer.drawOes(listener.oesTextureId, listener.transformMatrix);
+ drawer.drawOes(listener.oesTextureId, listener.transformMatrix, 0, 0, width, height);
drawer.release();
// Download the pixels in the pixel buffer as RGBA. Not all platforms support RGB, e.g. Nexus 9.
@@ -240,11 +248,11 @@
public static void testDisconnect() throws InterruptedException {
// Create SurfaceTextureHelper and listener.
final SurfaceTextureHelper surfaceTextureHelper =
- SurfaceTextureHelper.create(EGL10.EGL_NO_CONTEXT);
+ SurfaceTextureHelper.create(null);
final MockTextureListener listener = new MockTextureListener();
surfaceTextureHelper.setListener(listener);
// Create EglBase with the SurfaceTexture as target EGLSurface.
- final EglBase eglBase = new EglBase(EGL10.EGL_NO_CONTEXT, EglBase.ConfigType.PLAIN);
+ final EglBase eglBase = EglBase.create(null, EglBase.CONFIG_PLAIN);
eglBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
eglBase.makeCurrent();
// Assert no frame has been received yet.
@@ -276,7 +284,7 @@
@SmallTest
public static void testDisconnectImmediately() {
final SurfaceTextureHelper surfaceTextureHelper =
- SurfaceTextureHelper.create(EGL10.EGL_NO_CONTEXT);
+ SurfaceTextureHelper.create(null);
surfaceTextureHelper.disconnect();
}
@@ -292,14 +300,14 @@
// Create SurfaceTextureHelper and listener.
final SurfaceTextureHelper surfaceTextureHelper =
- SurfaceTextureHelper.create(EGL10.EGL_NO_CONTEXT, handler);
+ SurfaceTextureHelper.create(null, handler);
// Create a mock listener and expect frames to be delivered on |thread|.
final MockTextureListener listener = new MockTextureListener(thread);
surfaceTextureHelper.setListener(listener);
// Create resources for stubbing an OES texture producer. |eglOesBase| has the
// SurfaceTexture in |surfaceTextureHelper| as the target EGLSurface.
- final EglBase eglOesBase = new EglBase(EGL10.EGL_NO_CONTEXT, EglBase.ConfigType.PLAIN);
+ final EglBase eglOesBase = EglBase.create(null, EglBase.CONFIG_PLAIN);
eglOesBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
eglOesBase.makeCurrent();
// Draw a frame onto the SurfaceTexture.
@@ -313,7 +321,119 @@
// Return the frame from this thread.
surfaceTextureHelper.returnTextureFrame();
+ surfaceTextureHelper.disconnect(handler);
+ }
+
+ /**
+ * Test use SurfaceTextureHelper on a separate thread. A uniform texture frame is created and
+ * received on a thread separate from the test thread and returned after disconnect.
+ */
+ @MediumTest
+ public static void testLateReturnFrameOnSeparateThread() throws InterruptedException {
+ final HandlerThread thread = new HandlerThread("SurfaceTextureHelperTestThread");
+ thread.start();
+ final Handler handler = new Handler(thread.getLooper());
+
+ // Create SurfaceTextureHelper and listener.
+ final SurfaceTextureHelper surfaceTextureHelper =
+ SurfaceTextureHelper.create(null, handler);
+ // Create a mock listener and expect frames to be delivered on |thread|.
+ final MockTextureListener listener = new MockTextureListener(thread);
+ surfaceTextureHelper.setListener(listener);
+
+ // Create resources for stubbing an OES texture producer. |eglOesBase| has the
+ // SurfaceTexture in |surfaceTextureHelper| as the target EGLSurface.
+ final EglBase eglOesBase = EglBase.create(null, EglBase.CONFIG_PLAIN);
+ eglOesBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
+ eglOesBase.makeCurrent();
+ // Draw a frame onto the SurfaceTexture.
+ GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
+ // swapBuffers() will ultimately trigger onTextureFrameAvailable().
+ eglOesBase.swapBuffers();
+ eglOesBase.release();
+
+ // Wait for an OES texture to arrive.
+ listener.waitForNewFrame();
+
+ surfaceTextureHelper.disconnect(handler);
+
+ surfaceTextureHelper.returnTextureFrame();
+ }
+
+ @MediumTest
+ public static void testTexturetoYUV() throws InterruptedException {
+ final int width = 16;
+ final int height = 16;
+
+ final EglBase eglBase = EglBase.create(null, EglBase.CONFIG_PLAIN);
+
+ // Create SurfaceTextureHelper and listener.
+ final SurfaceTextureHelper surfaceTextureHelper =
+ SurfaceTextureHelper.create(eglBase.getEglBaseContext());
+ final MockTextureListener listener = new MockTextureListener();
+ surfaceTextureHelper.setListener(listener);
+ surfaceTextureHelper.getSurfaceTexture().setDefaultBufferSize(width, height);
+
+ // Create resources for stubbing an OES texture producer. |eglBase| has the SurfaceTexture in
+ // |surfaceTextureHelper| as the target EGLSurface.
+
+ eglBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
+ assertEquals(eglBase.surfaceWidth(), width);
+ assertEquals(eglBase.surfaceHeight(), height);
+
+ final int red[] = new int[] {79, 144, 185};
+ final int green[] = new int[] {66, 210, 162};
+ final int blue[] = new int[] {161, 117, 158};
+
+ final int ref_y[] = new int[] {81, 180, 168};
+ final int ref_u[] = new int[] {173, 93, 122};
+ final int ref_v[] = new int[] {127, 103, 140};
+
+ // Draw three frames.
+ for (int i = 0; i < 3; ++i) {
+ // Draw a constant color frame onto the SurfaceTexture.
+ eglBase.makeCurrent();
+ GLES20.glClearColor(red[i] / 255.0f, green[i] / 255.0f, blue[i] / 255.0f, 1.0f);
+ GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
+ // swapBuffers() will ultimately trigger onTextureFrameAvailable().
+ eglBase.swapBuffers();
+
+ // Wait for an OES texture to arrive.
+ listener.waitForNewFrame();
+
+ // Memory layout: Lines are 16 bytes. First 16 lines are
+ // the Y data. These are followed by 8 lines with 8 bytes of U
+ // data on the left and 8 bytes of V data on the right.
+ //
+ // Offset
+ // 0 YYYYYYYY YYYYYYYY
+ // 16 YYYYYYYY YYYYYYYY
+ // ...
+ // 240 YYYYYYYY YYYYYYYY
+ // 256 UUUUUUUU VVVVVVVV
+ // 272 UUUUUUUU VVVVVVVV
+ // ...
+ // 368 UUUUUUUU VVVVVVVV
+ // 384 buffer end
+ ByteBuffer buffer = ByteBuffer.allocateDirect(width * height * 3 / 2);
+ surfaceTextureHelper.textureToYUV(buffer, width, height, width,
+ listener.oesTextureId, listener.transformMatrix);
+
+ surfaceTextureHelper.returnTextureFrame();
+
+ // Allow off-by-one differences due to different rounding.
+ while (buffer.position() < width*height) {
+ assertClose(1, buffer.get() & 0xff, ref_y[i]);
+ }
+ while (buffer.hasRemaining()) {
+ if (buffer.position() % width < width/2)
+ assertClose(1, buffer.get() & 0xff, ref_u[i]);
+ else
+ assertClose(1, buffer.get() & 0xff, ref_v[i]);
+ }
+ }
+
surfaceTextureHelper.disconnect();
- thread.quitSafely();
+ eglBase.release();
}
}
diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java b/talk/app/webrtc/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java
index 47fe780..341c632 100644
--- a/talk/app/webrtc/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java
+++ b/talk/app/webrtc/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java
@@ -36,8 +36,6 @@
import java.util.Arrays;
import java.util.List;
-import javax.microedition.khronos.egl.EGL10;
-
public final class SurfaceViewRendererOnMeasureTest extends ActivityTestCase {
/**
* List with all possible scaling types.
@@ -111,7 +109,7 @@
}
// Test behaviour after SurfaceViewRenderer.init() is called, but still no frame.
- surfaceViewRenderer.init(EGL10.EGL_NO_CONTEXT, null);
+ surfaceViewRenderer.init((EglBase.Context) null, null);
for (RendererCommon.ScalingType scalingType : scalingTypes) {
for (int measureSpecMode : measureSpecModes) {
final int zeroMeasureSize = MeasureSpec.makeMeasureSpec(0, measureSpecMode);
@@ -134,7 +132,7 @@
public void testFrame1280x720() {
final SurfaceViewRenderer surfaceViewRenderer =
new SurfaceViewRenderer(getInstrumentation().getContext());
- surfaceViewRenderer.init(EGL10.EGL_NO_CONTEXT, null);
+ surfaceViewRenderer.init((EglBase.Context) null, null);
// Test different rotation degress, but same rotated size.
for (int rotationDegree : new int[] {0, 90, 180, 270}) {
diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java b/talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
index dbbe596..1b97201 100644
--- a/talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
+++ b/talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
@@ -29,7 +29,6 @@
import android.test.ActivityTestCase;
import android.test.suitebuilder.annotation.MediumTest;
import android.test.suitebuilder.annotation.SmallTest;
-import android.util.Log;
import android.util.Size;
import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
@@ -37,8 +36,6 @@
import java.util.HashSet;
import java.util.Set;
-import javax.microedition.khronos.egl.EGL10;
-
@SuppressWarnings("deprecation")
public class VideoCapturerAndroidTest extends ActivityTestCase {
static final String TAG = "VideoCapturerAndroidTest";
@@ -87,8 +84,10 @@
@SmallTest
public void testCreateAndReleaseUsingTextures() {
+ EglBase eglBase = EglBase.create();
VideoCapturerAndroidTestFixtures.release(
- VideoCapturerAndroid.create("", null, EGL10.EGL_NO_CONTEXT));
+ VideoCapturerAndroid.create("", null, eglBase.getEglBaseContext()));
+ eglBase.release();
}
@SmallTest
@@ -108,12 +107,13 @@
VideoCapturerAndroidTestFixtures.startCapturerAndRender(capturer);
}
- // TODO(perkj): Enable once VideoCapture to texture support has landed in C++.
@SmallTest
- public void DISABLED_testStartVideoCapturerUsingTextures() throws InterruptedException {
+ public void testStartVideoCapturerUsingTextures() throws InterruptedException {
+ EglBase eglBase = EglBase.create();
VideoCapturerAndroid capturer =
- VideoCapturerAndroid.create("", null, EGL10.EGL_NO_CONTEXT);
+ VideoCapturerAndroid.create("", null, eglBase.getEglBaseContext());
VideoCapturerAndroidTestFixtures.startCapturerAndRender(capturer);
+ eglBase.release();
}
@SmallTest
@@ -151,11 +151,13 @@
VideoCapturerAndroidTestFixtures.switchCamera(capturer);
}
- // TODO(perkj): Enable once VideoCapture to texture support has landed in C++.
@SmallTest
- public void DISABLED_testSwitchVideoCapturerUsingTextures() throws InterruptedException {
- VideoCapturerAndroid capturer = VideoCapturerAndroid.create("", null, EGL10.EGL_NO_CONTEXT);
+ public void testSwitchVideoCapturerUsingTextures() throws InterruptedException {
+ EglBase eglBase = EglBase.create();
+ VideoCapturerAndroid capturer =
+ VideoCapturerAndroid.create("", null, eglBase.getEglBaseContext());
VideoCapturerAndroidTestFixtures.switchCamera(capturer);
+ eglBase.release();
}
@MediumTest
@@ -179,12 +181,14 @@
@MediumTest
public void testCameraCallsAfterStopUsingTextures() throws InterruptedException {
+ EglBase eglBase = EglBase.create();
final String deviceName = CameraEnumerationAndroid.getDeviceName(0);
final VideoCapturerAndroid capturer = VideoCapturerAndroid.create(deviceName, null,
- EGL10.EGL_NO_CONTEXT);
+ eglBase.getEglBaseContext());
VideoCapturerAndroidTestFixtures.cameraCallsAfterStop(capturer,
getInstrumentation().getContext());
+ eglBase.release();
}
@SmallTest
@@ -195,11 +199,13 @@
VideoCapturerAndroidTestFixtures.stopRestartVideoSource(capturer);
}
- // TODO(perkj): Enable once VideoCapture to texture support has landed in C++.
@SmallTest
- public void DISABLED_testStopRestartVideoSourceUsingTextures() throws InterruptedException {
- VideoCapturerAndroid capturer = VideoCapturerAndroid.create("", null, EGL10.EGL_NO_CONTEXT);
+ public void testStopRestartVideoSourceUsingTextures() throws InterruptedException {
+ EglBase eglBase = EglBase.create();
+ VideoCapturerAndroid capturer =
+ VideoCapturerAndroid.create("", null, eglBase.getEglBaseContext());
VideoCapturerAndroidTestFixtures.stopRestartVideoSource(capturer);
+ eglBase.release();
}
@SmallTest
@@ -215,14 +221,51 @@
@SmallTest
public void testStartStopWithDifferentResolutionsUsingTextures() throws InterruptedException {
+ EglBase eglBase = EglBase.create();
String deviceName = CameraEnumerationAndroid.getDeviceName(0);
VideoCapturerAndroid capturer =
- VideoCapturerAndroid.create(deviceName, null, EGL10.EGL_NO_CONTEXT);
+ VideoCapturerAndroid.create(deviceName, null, eglBase.getEglBaseContext());
VideoCapturerAndroidTestFixtures.startStopWithDifferentResolutions(capturer,
getInstrumentation().getContext());
+ eglBase.release();
}
@SmallTest
+ // This test that an error is reported if the camera is already opened
+ // when VideoCapturerAndroid is started.
+ public void testStartWhileCameraAlreadyOpened() throws InterruptedException {
+ String deviceName = CameraEnumerationAndroid.getDeviceName(0);
+ VideoCapturerAndroid capturer =
+ VideoCapturerAndroid.create(deviceName, null);
+ VideoCapturerAndroidTestFixtures.startWhileCameraIsAlreadyOpen(
+ capturer, getInstrumentation().getContext());
+ }
+
+ @SmallTest
+ // This test that VideoCapturerAndroid can be started, even if the camera is already opened
+ // if the camera is closed while VideoCapturerAndroid is re-trying to start.
+ public void testStartWhileCameraIsAlreadyOpenAndCloseCamera() throws InterruptedException {
+ String deviceName = CameraEnumerationAndroid.getDeviceName(0);
+ VideoCapturerAndroid capturer =
+ VideoCapturerAndroid.create(deviceName, null);
+ VideoCapturerAndroidTestFixtures.startWhileCameraIsAlreadyOpenAndCloseCamera(
+ capturer, getInstrumentation().getContext());
+ }
+
+ @SmallTest
+ // This test that VideoCapturerAndroid.stop can be called while VideoCapturerAndroid is
+ // re-trying to start.
+ public void startWhileCameraIsAlreadyOpenAndStop() throws InterruptedException {
+ String deviceName = CameraEnumerationAndroid.getDeviceName(0);
+ VideoCapturerAndroid capturer =
+ VideoCapturerAndroid.create(deviceName, null);
+ VideoCapturerAndroidTestFixtures.startWhileCameraIsAlreadyOpenAndStop(
+ capturer, getInstrumentation().getContext());
+ }
+
+
+
+ @SmallTest
// This test what happens if buffers are returned after the capturer have
// been stopped and restarted. It does not test or use the C++ layer.
public void testReturnBufferLate() throws InterruptedException {
@@ -235,11 +278,13 @@
@SmallTest
public void testReturnBufferLateUsingTextures() throws InterruptedException {
+ EglBase eglBase = EglBase.create();
String deviceName = CameraEnumerationAndroid.getDeviceName(0);
VideoCapturerAndroid capturer =
- VideoCapturerAndroid.create(deviceName, null, EGL10.EGL_NO_CONTEXT);
+ VideoCapturerAndroid.create(deviceName, null, eglBase.getEglBaseContext());
VideoCapturerAndroidTestFixtures.returnBufferLate(capturer,
getInstrumentation().getContext());
+ eglBase.release();
}
@MediumTest
@@ -251,11 +296,45 @@
VideoCapturerAndroidTestFixtures.returnBufferLateEndToEnd(capturer);
}
- // TODO(perkj): Enable once VideoCapture to texture support has landed in C++.
@MediumTest
- public void DISABLED_testReturnBufferLateEndToEndUsingTextures() throws InterruptedException {
+ public void testReturnBufferLateEndToEndUsingTextures() throws InterruptedException {
+ EglBase eglBase = EglBase.create();
final VideoCapturerAndroid capturer =
- VideoCapturerAndroid.create("", null, EGL10.EGL_NO_CONTEXT);
+ VideoCapturerAndroid.create("", null, eglBase.getEglBaseContext());
VideoCapturerAndroidTestFixtures.returnBufferLateEndToEnd(capturer);
+ eglBase.release();
+ }
+
+ @MediumTest
+ // This test that CameraEventsHandler.onError is triggered if video buffers are not returned to
+ // the capturer.
+ public void testCameraFreezedEventOnBufferStarvationUsingTextures() throws InterruptedException {
+ EglBase eglBase = EglBase.create();
+ VideoCapturerAndroidTestFixtures.CameraEvents cameraEvents =
+ VideoCapturerAndroidTestFixtures.createCameraEvents();
+ VideoCapturerAndroid capturer = VideoCapturerAndroid.create("", cameraEvents,
+ eglBase.getEglBaseContext());
+ VideoCapturerAndroidTestFixtures.cameraFreezedEventOnBufferStarvationUsingTextures(capturer,
+ cameraEvents, getInstrumentation().getContext());
+ eglBase.release();
+ }
+
+ @MediumTest
+ // This test that frames forwarded to a renderer is scaled if onOutputFormatRequest is
+ // called. This test both Java and C++ parts of of the stack.
+ public void testScaleCameraOutput() throws InterruptedException {
+ VideoCapturerAndroid capturer = VideoCapturerAndroid.create("", null);
+ VideoCapturerAndroidTestFixtures.scaleCameraOutput(capturer);
+ }
+
+ @MediumTest
+ // This test that frames forwarded to a renderer is scaled if onOutputFormatRequest is
+ // called. This test both Java and C++ parts of of the stack.
+ public void testScaleCameraOutputUsingTextures() throws InterruptedException {
+ EglBase eglBase = EglBase.create();
+ VideoCapturerAndroid capturer =
+ VideoCapturerAndroid.create("", null, eglBase.getEglBaseContext());
+ VideoCapturerAndroidTestFixtures.scaleCameraOutput(capturer);
+ eglBase.release();
}
}
diff --git a/talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java b/talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
index 11b3ce9..0b42e33 100644
--- a/talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
+++ b/talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
@@ -29,6 +29,7 @@
import android.content.Context;
import android.hardware.Camera;
+import org.webrtc.VideoCapturerAndroidTestFixtures;
import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
import org.webrtc.VideoRenderer.I420Frame;
@@ -42,16 +43,32 @@
static class RendererCallbacks implements VideoRenderer.Callbacks {
private int framesRendered = 0;
private Object frameLock = 0;
+ private int width = 0;
+ private int height = 0;
@Override
public void renderFrame(I420Frame frame) {
synchronized (frameLock) {
++framesRendered;
+ width = frame.rotatedWidth();
+ height = frame.rotatedHeight();
frameLock.notify();
}
VideoRenderer.renderFrameDone(frame);
}
+ public int frameWidth() {
+ synchronized (frameLock) {
+ return width;
+ }
+ }
+
+ public int frameHeight() {
+ synchronized (frameLock) {
+ return height;
+ }
+ }
+
public int WaitForNextFrameToRender() throws InterruptedException {
synchronized (frameLock) {
frameLock.wait();
@@ -102,11 +119,11 @@
}
@Override
- public void onByteBufferFrameCaptured(byte[] frame, int length, int width, int height,
- int rotation, long timeStamp) {
+ public void onByteBufferFrameCaptured(byte[] frame, int width, int height, int rotation,
+ long timeStamp) {
synchronized (frameLock) {
++framesCaptured;
- frameSize = length;
+ frameSize = frame.length;
frameWidth = width;
frameHeight = height;
timestamps.add(timeStamp);
@@ -115,7 +132,8 @@
}
@Override
public void onTextureFrameCaptured(
- int width, int height, int oesTextureId, float[] transformMatrix, long timeStamp) {
+ int width, int height, int oesTextureId, float[] transformMatrix, int rotation,
+ long timeStamp) {
synchronized (frameLock) {
++framesCaptured;
frameWidth = width;
@@ -174,9 +192,20 @@
VideoCapturerAndroid.CameraEventsHandler {
public boolean onCameraOpeningCalled;
public boolean onFirstFrameAvailableCalled;
+ public final Object onCameraFreezedLock = new Object();
+ private String onCameraFreezedDescription;
@Override
- public void onCameraError(String errorDescription) { }
+ public void onCameraError(String errorDescription) {
+ }
+
+ @Override
+ public void onCameraFreezed(String errorDescription) {
+ synchronized (onCameraFreezedLock) {
+ onCameraFreezedDescription = errorDescription;
+ onCameraFreezedLock.notifyAll();
+ }
+ }
@Override
public void onCameraOpening(int cameraId) {
@@ -190,6 +219,13 @@
@Override
public void onCameraClosed() { }
+
+ public String WaitForCameraFreezed() throws InterruptedException {
+ synchronized (onCameraFreezedLock) {
+ onCameraFreezedLock.wait();
+ return onCameraFreezedDescription;
+ }
+ }
}
static public CameraEvents createCameraEvents() {
@@ -275,8 +311,8 @@
assertTrue(observer.WaitForCapturerToStart());
observer.WaitForNextCapturedFrame();
capturer.stopCapture();
- for (long timeStamp : observer.getCopyAndResetListOftimeStamps()) {
- capturer.returnBuffer(timeStamp);
+ if (capturer.isCapturingToTexture()) {
+ capturer.surfaceHelper.returnTextureFrame();
}
capturer.dispose();
@@ -296,9 +332,10 @@
// Make sure camera is started and then stop it.
assertTrue(observer.WaitForCapturerToStart());
capturer.stopCapture();
- for (long timeStamp : observer.getCopyAndResetListOftimeStamps()) {
- capturer.returnBuffer(timeStamp);
+ if (capturer.isCapturingToTexture()) {
+ capturer.surfaceHelper.returnTextureFrame();
}
+
// We can't change |capturer| at this point, but we should not crash.
capturer.switchCamera(null);
capturer.onOutputFormatRequest(640, 480, 15);
@@ -357,17 +394,90 @@
if (capturer.isCapturingToTexture()) {
assertEquals(0, observer.frameSize());
} else {
- assertEquals(format.frameSize(), observer.frameSize());
+ assertTrue(format.frameSize() <= observer.frameSize());
}
capturer.stopCapture();
- for (long timestamp : observer.getCopyAndResetListOftimeStamps()) {
- capturer.returnBuffer(timestamp);
+ if (capturer.isCapturingToTexture()) {
+ capturer.surfaceHelper.returnTextureFrame();
}
}
capturer.dispose();
assertTrue(capturer.isReleased());
}
+ static void waitUntilIdle(VideoCapturerAndroid capturer) throws InterruptedException {
+ final CountDownLatch barrier = new CountDownLatch(1);
+ capturer.getCameraThreadHandler().post(new Runnable() {
+ @Override public void run() {
+ barrier.countDown();
+ }
+ });
+ barrier.await();
+ }
+
+ static public void startWhileCameraIsAlreadyOpen(
+ VideoCapturerAndroid capturer, Context appContext) throws InterruptedException {
+ Camera camera = Camera.open(capturer.getCurrentCameraId());
+
+ final List<CaptureFormat> formats = capturer.getSupportedFormats();
+ final CameraEnumerationAndroid.CaptureFormat format = formats.get(0);
+
+ final FakeCapturerObserver observer = new FakeCapturerObserver();
+ capturer.startCapture(format.width, format.height, format.maxFramerate,
+ appContext, observer);
+
+ if (android.os.Build.VERSION.SDK_INT > android.os.Build.VERSION_CODES.LOLLIPOP_MR1) {
+ // The first opened camera client will be evicted.
+ assertTrue(observer.WaitForCapturerToStart());
+ capturer.stopCapture();
+ } else {
+ assertFalse(observer.WaitForCapturerToStart());
+ }
+
+ capturer.dispose();
+ camera.release();
+ }
+
+ static public void startWhileCameraIsAlreadyOpenAndCloseCamera(
+ VideoCapturerAndroid capturer, Context appContext) throws InterruptedException {
+ Camera camera = Camera.open(capturer.getCurrentCameraId());
+
+ final List<CaptureFormat> formats = capturer.getSupportedFormats();
+ final CameraEnumerationAndroid.CaptureFormat format = formats.get(0);
+
+ final FakeCapturerObserver observer = new FakeCapturerObserver();
+ capturer.startCapture(format.width, format.height, format.maxFramerate,
+ appContext, observer);
+ waitUntilIdle(capturer);
+
+ camera.release();
+
+ // Make sure camera is started and first frame is received and then stop it.
+ assertTrue(observer.WaitForCapturerToStart());
+ observer.WaitForNextCapturedFrame();
+ capturer.stopCapture();
+ if (capturer.isCapturingToTexture()) {
+ capturer.surfaceHelper.returnTextureFrame();
+ }
+ capturer.dispose();
+ assertTrue(capturer.isReleased());
+ }
+
+ static public void startWhileCameraIsAlreadyOpenAndStop(
+ VideoCapturerAndroid capturer, Context appContext) throws InterruptedException {
+ Camera camera = Camera.open(capturer.getCurrentCameraId());
+ final List<CaptureFormat> formats = capturer.getSupportedFormats();
+ final CameraEnumerationAndroid.CaptureFormat format = formats.get(0);
+
+ final FakeCapturerObserver observer = new FakeCapturerObserver();
+ capturer.startCapture(format.width, format.height, format.maxFramerate,
+ appContext, observer);
+ capturer.stopCapture();
+ capturer.dispose();
+ assertTrue(capturer.isReleased());
+ camera.release();
+ }
+
static public void returnBufferLate(VideoCapturerAndroid capturer,
Context appContext) throws InterruptedException {
FakeCapturerObserver observer = new FakeCapturerObserver();
@@ -387,9 +497,8 @@
capturer.startCapture(format.width, format.height, format.maxFramerate,
appContext, observer);
observer.WaitForCapturerToStart();
-
- for (Long timeStamp : listOftimestamps) {
- capturer.returnBuffer(timeStamp);
+ if (capturer.isCapturingToTexture()) {
+ capturer.surfaceHelper.returnTextureFrame();
}
observer.WaitForNextCapturedFrame();
@@ -397,9 +506,10 @@
listOftimestamps = observer.getCopyAndResetListOftimeStamps();
assertTrue(listOftimestamps.size() >= 1);
- for (Long timeStamp : listOftimestamps) {
- capturer.returnBuffer(timeStamp);
+ if (capturer.isCapturingToTexture()) {
+ capturer.surfaceHelper.returnTextureFrame();
}
+
capturer.dispose();
assertTrue(capturer.isReleased());
}
@@ -410,6 +520,7 @@
final VideoSource source = factory.createVideoSource(capturer, new MediaConstraints());
final VideoTrack track = factory.createVideoTrack("dummy", source);
final FakeAsyncRenderer renderer = new FakeAsyncRenderer();
+
track.addRenderer(new VideoRenderer(renderer));
// Wait for at least one frame that has not been returned.
assertFalse(renderer.waitForPendingFrames().isEmpty());
@@ -420,9 +531,7 @@
track.dispose();
source.dispose();
factory.dispose();
-
- // The pending frames should keep the JNI parts and |capturer| alive.
- assertFalse(capturer.isReleased());
+ assertTrue(capturer.isReleased());
// Return the frame(s), on a different thread out of spite.
final List<I420Frame> pendingFrames = renderer.waitForPendingFrames();
@@ -436,8 +545,71 @@
});
returnThread.start();
returnThread.join();
+ }
- // Check that frames have successfully returned. This will cause |capturer| to be released.
+ static public void cameraFreezedEventOnBufferStarvationUsingTextures(
+ VideoCapturerAndroid capturer,
+ CameraEvents events, Context appContext) throws InterruptedException {
+ assertTrue("Not capturing to textures.", capturer.isCapturingToTexture());
+
+ final List<CaptureFormat> formats = capturer.getSupportedFormats();
+ final CameraEnumerationAndroid.CaptureFormat format = formats.get(0);
+
+ final FakeCapturerObserver observer = new FakeCapturerObserver();
+ capturer.startCapture(format.width, format.height, format.maxFramerate,
+ appContext, observer);
+ // Make sure camera is started.
+ assertTrue(observer.WaitForCapturerToStart());
+ // Since we don't return the buffer, we should get a starvation message if we are
+ // capturing to a texture.
+ assertEquals("Camera failure. Client must return video buffers.",
+ events.WaitForCameraFreezed());
+
+ capturer.stopCapture();
+ if (capturer.isCapturingToTexture()) {
+ capturer.surfaceHelper.returnTextureFrame();
+ }
+
+ capturer.dispose();
assertTrue(capturer.isReleased());
}
+
+ static public void scaleCameraOutput(VideoCapturerAndroid capturer) throws InterruptedException {
+ PeerConnectionFactory factory = new PeerConnectionFactory();
+ VideoSource source =
+ factory.createVideoSource(capturer, new MediaConstraints());
+ VideoTrack track = factory.createVideoTrack("dummy", source);
+ RendererCallbacks renderer = new RendererCallbacks();
+ track.addRenderer(new VideoRenderer(renderer));
+ assertTrue(renderer.WaitForNextFrameToRender() > 0);
+
+ final int startWidth = renderer.frameWidth();
+ final int startHeight = renderer.frameHeight();
+ final int frameRate = 30;
+ final int scaledWidth = startWidth / 2;
+ final int scaledHeight = startHeight / 2;
+
+ // Request the captured frames to be scaled.
+ capturer.onOutputFormatRequest(scaledWidth, scaledHeight, frameRate);
+
+ boolean gotExpectedResolution = false;
+ int numberOfInspectedFrames = 0;
+
+ do {
+ renderer.WaitForNextFrameToRender();
+ ++numberOfInspectedFrames;
+
+ gotExpectedResolution = (renderer.frameWidth() == scaledWidth
+ && renderer.frameHeight() == scaledHeight);
+ } while (!gotExpectedResolution && numberOfInspectedFrames < 30);
+
+ source.stop();
+ track.dispose();
+ source.dispose();
+ factory.dispose();
+ assertTrue(capturer.isReleased());
+
+ assertTrue(gotExpectedResolution);
+ }
+
}
diff --git a/talk/app/webrtc/androidvideocapturer.cc b/talk/app/webrtc/androidvideocapturer.cc
index afcfb5b..d8f1217 100644
--- a/talk/app/webrtc/androidvideocapturer.cc
+++ b/talk/app/webrtc/androidvideocapturer.cc
@@ -26,6 +26,7 @@
*/
#include "talk/app/webrtc/androidvideocapturer.h"
+#include "talk/app/webrtc/java/jni/native_handle_impl.h"
#include "talk/media/webrtc/webrtcvideoframe.h"
#include "webrtc/base/common.h"
#include "webrtc/base/json.h"
@@ -57,11 +58,13 @@
const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
int rotation,
int64_t time_stamp_in_ns) {
+ RTC_DCHECK(rotation == 0 || rotation == 90 || rotation == 180 ||
+ rotation == 270);
buffer_ = buffer;
captured_frame_.width = buffer->width();
captured_frame_.height = buffer->height();
captured_frame_.time_stamp = time_stamp_in_ns;
- captured_frame_.rotation = rotation;
+ captured_frame_.rotation = static_cast<webrtc::VideoRotation>(rotation);
}
void ClearCapturedFrame() {
@@ -85,7 +88,7 @@
rtc::scoped_ptr<cricket::VideoFrame> frame(new cricket::WebRtcVideoFrame(
ShallowCenterCrop(buffer_, dst_width, dst_height),
- captured_frame->time_stamp, captured_frame->GetRotation()));
+ captured_frame->time_stamp, captured_frame->rotation));
// Caller takes ownership.
// TODO(magjed): Change CreateAliasedFrame() to return a rtc::scoped_ptr.
return apply_rotation_ ? frame->GetCopyWithRotationApplied()->Copy()
@@ -99,10 +102,17 @@
int output_width,
int output_height) const override {
if (buffer_->native_handle() != nullptr) {
- // TODO(perkj): Implement CreateAliasedFrame properly for textures.
- rtc::scoped_ptr<cricket::VideoFrame> frame(new cricket::WebRtcVideoFrame(
- buffer_, input_frame->time_stamp, input_frame->GetRotation()));
- return frame.release();
+ // TODO(perkj) Implement cropping.
+ RTC_CHECK_EQ(cropped_input_width, buffer_->width());
+ RTC_CHECK_EQ(cropped_input_height, buffer_->height());
+ rtc::scoped_refptr<webrtc::VideoFrameBuffer> scaled_buffer(
+ static_cast<webrtc_jni::AndroidTextureBuffer*>(buffer_.get())
+ ->ScaleAndRotate(output_width, output_height,
+ apply_rotation_ ? input_frame->rotation :
+ webrtc::kVideoRotation_0));
+ return new cricket::WebRtcVideoFrame(
+ scaled_buffer, input_frame->time_stamp,
+ apply_rotation_ ? webrtc::kVideoRotation_0 : input_frame->rotation);
}
return VideoFrameFactory::CreateAliasedFrame(input_frame,
cropped_input_width,
diff --git a/talk/app/webrtc/androidvideocapturer.h b/talk/app/webrtc/androidvideocapturer.h
index df783bd..c665eab 100644
--- a/talk/app/webrtc/androidvideocapturer.h
+++ b/talk/app/webrtc/androidvideocapturer.h
@@ -32,7 +32,7 @@
#include "talk/media/base/videocapturer.h"
#include "webrtc/base/thread_checker.h"
-#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
namespace webrtc {
diff --git a/talk/app/webrtc/audiotrack.cc b/talk/app/webrtc/audiotrack.cc
index b0c9129..b3223cd 100644
--- a/talk/app/webrtc/audiotrack.cc
+++ b/talk/app/webrtc/audiotrack.cc
@@ -27,27 +27,82 @@
#include "talk/app/webrtc/audiotrack.h"
-#include <string>
+#include "webrtc/base/checks.h"
+
+using rtc::scoped_refptr;
namespace webrtc {
-static const char kAudioTrackKind[] = "audio";
+const char MediaStreamTrackInterface::kAudioKind[] = "audio";
+
+// static
+scoped_refptr<AudioTrack> AudioTrack::Create(
+ const std::string& id,
+ const scoped_refptr<AudioSourceInterface>& source) {
+ return new rtc::RefCountedObject<AudioTrack>(id, source);
+}
AudioTrack::AudioTrack(const std::string& label,
- AudioSourceInterface* audio_source)
- : MediaStreamTrack<AudioTrackInterface>(label),
- audio_source_(audio_source) {
+ const scoped_refptr<AudioSourceInterface>& source)
+ : MediaStreamTrack<AudioTrackInterface>(label), audio_source_(source) {
+ if (audio_source_) {
+ audio_source_->RegisterObserver(this);
+ OnChanged();
+ }
+}
+
+AudioTrack::~AudioTrack() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ set_state(MediaStreamTrackInterface::kEnded);
+ if (audio_source_)
+ audio_source_->UnregisterObserver(this);
}
std::string AudioTrack::kind() const {
- return kAudioTrackKind;
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ return kAudioKind;
}
-rtc::scoped_refptr<AudioTrack> AudioTrack::Create(
- const std::string& id, AudioSourceInterface* source) {
- rtc::RefCountedObject<AudioTrack>* track =
- new rtc::RefCountedObject<AudioTrack>(id, source);
- return track;
+AudioSourceInterface* AudioTrack::GetSource() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ return audio_source_.get();
+}
+
+void AudioTrack::AddSink(AudioTrackSinkInterface* sink) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (audio_source_)
+ audio_source_->AddSink(sink);
+}
+
+void AudioTrack::RemoveSink(AudioTrackSinkInterface* sink) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (audio_source_)
+ audio_source_->RemoveSink(sink);
+}
+
+void AudioTrack::OnChanged() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (state() == kFailed)
+ return; // We can't recover from this state (do we ever set it?).
+
+ TrackState new_state = kInitializing;
+
+ // |audio_source_| must be non-null if we ever get here.
+ switch (audio_source_->state()) {
+ case MediaSourceInterface::kLive:
+ case MediaSourceInterface::kMuted:
+ new_state = kLive;
+ break;
+ case MediaSourceInterface::kEnded:
+ new_state = kEnded;
+ break;
+ case MediaSourceInterface::kInitializing:
+ default:
+ // use kInitializing.
+ break;
+ }
+
+ set_state(new_state);
}
} // namespace webrtc
diff --git a/talk/app/webrtc/audiotrack.h b/talk/app/webrtc/audiotrack.h
index 750f272..55f4837 100644
--- a/talk/app/webrtc/audiotrack.h
+++ b/talk/app/webrtc/audiotrack.h
@@ -28,40 +28,47 @@
#ifndef TALK_APP_WEBRTC_AUDIOTRACK_H_
#define TALK_APP_WEBRTC_AUDIOTRACK_H_
+#include <string>
+
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/mediastreamtrack.h"
#include "talk/app/webrtc/notifier.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/thread_checker.h"
namespace webrtc {
-class AudioTrack : public MediaStreamTrack<AudioTrackInterface> {
+class AudioTrack : public MediaStreamTrack<AudioTrackInterface>,
+ public ObserverInterface {
+ protected:
+ // Protected ctor to force use of factory method.
+ AudioTrack(const std::string& label,
+ const rtc::scoped_refptr<AudioSourceInterface>& source);
+ ~AudioTrack() override;
+
public:
static rtc::scoped_refptr<AudioTrack> Create(
- const std::string& id, AudioSourceInterface* source);
+ const std::string& id,
+ const rtc::scoped_refptr<AudioSourceInterface>& source);
- // AudioTrackInterface implementation.
- AudioSourceInterface* GetSource() const override {
- return audio_source_.get();
- }
- // TODO(xians): Implement these methods.
- void AddSink(AudioTrackSinkInterface* sink) override {}
- void RemoveSink(AudioTrackSinkInterface* sink) override {}
- bool GetSignalLevel(int* level) override { return false; }
- rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
- return NULL;
- }
- cricket::AudioRenderer* GetRenderer() override { return NULL; }
-
+ private:
// MediaStreamTrack implementation.
std::string kind() const override;
- protected:
- AudioTrack(const std::string& label, AudioSourceInterface* audio_source);
+ // AudioTrackInterface implementation.
+ AudioSourceInterface* GetSource() const override;
+
+ void AddSink(AudioTrackSinkInterface* sink) override;
+ void RemoveSink(AudioTrackSinkInterface* sink) override;
+
+ // ObserverInterface implementation.
+ void OnChanged() override;
private:
- rtc::scoped_refptr<AudioSourceInterface> audio_source_;
+ const rtc::scoped_refptr<AudioSourceInterface> audio_source_;
+ rtc::ThreadChecker thread_checker_;
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioTrack);
};
} // namespace webrtc
diff --git a/talk/app/webrtc/dtlsidentitystore.cc b/talk/app/webrtc/dtlsidentitystore.cc
index 2758779..390ec0d 100644
--- a/talk/app/webrtc/dtlsidentitystore.cc
+++ b/talk/app/webrtc/dtlsidentitystore.cc
@@ -27,6 +27,8 @@
#include "talk/app/webrtc/dtlsidentitystore.h"
+#include <utility>
+
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
#include "webrtc/base/logging.h"
@@ -72,7 +74,7 @@
// Posting to |this| avoids touching |store_| on threads other than
// |signaling_thread_| and thus avoids having to use locks.
IdentityResultMessageData* msg = new IdentityResultMessageData(
- new IdentityResult(key_type_, identity.Pass()));
+ new IdentityResult(key_type_, std::move(identity)));
signaling_thread_->Post(this, MSG_GENERATE_IDENTITY_RESULT, msg);
}
@@ -93,7 +95,7 @@
static_cast<IdentityResultMessageData*>(msg->pdata));
if (store_) {
store_->OnIdentityGenerated(pdata->data()->key_type_,
- pdata->data()->identity_.Pass());
+ std::move(pdata->data()->identity_));
}
}
break;
@@ -152,7 +154,7 @@
rtc::scoped_ptr<IdentityResultMessageData> pdata(
static_cast<IdentityResultMessageData*>(msg->pdata));
OnIdentityGenerated(pdata->data()->key_type_,
- pdata->data()->identity_.Pass());
+ std::move(pdata->data()->identity_));
break;
}
}
@@ -178,9 +180,9 @@
// Return identity async - post even though we are on |signaling_thread_|.
LOG(LS_VERBOSE) << "Using a free DTLS identity.";
++request_info_[key_type].gen_in_progress_counts_;
- IdentityResultMessageData* msg = new IdentityResultMessageData(
- new IdentityResult(key_type,
- request_info_[key_type].free_identity_.Pass()));
+ IdentityResultMessageData* msg =
+ new IdentityResultMessageData(new IdentityResult(
+ key_type, std::move(request_info_[key_type].free_identity_)));
signaling_thread_->Post(this, MSG_GENERATE_IDENTITY_RESULT, msg);
return;
}
@@ -228,7 +230,7 @@
// Return the result to the observer.
if (identity.get()) {
LOG(LS_VERBOSE) << "A DTLS identity is returned to an observer.";
- observer->OnSuccess(identity.Pass());
+ observer->OnSuccess(std::move(identity));
} else {
LOG(LS_WARNING) << "Failed to generate DTLS identity.";
observer->OnFailure(0);
diff --git a/talk/app/webrtc/dtlsidentitystore.h b/talk/app/webrtc/dtlsidentitystore.h
index a0eef98..2a5309d 100644
--- a/talk/app/webrtc/dtlsidentitystore.h
+++ b/talk/app/webrtc/dtlsidentitystore.h
@@ -30,6 +30,7 @@
#include <queue>
#include <string>
+#include <utility>
#include "webrtc/base/messagehandler.h"
#include "webrtc/base/messagequeue.h"
@@ -129,7 +130,7 @@
struct IdentityResult {
IdentityResult(rtc::KeyType key_type,
rtc::scoped_ptr<rtc::SSLIdentity> identity)
- : key_type_(key_type), identity_(identity.Pass()) {}
+ : key_type_(key_type), identity_(std::move(identity)) {}
rtc::KeyType key_type_;
rtc::scoped_ptr<rtc::SSLIdentity> identity_;
diff --git a/talk/app/webrtc/fakeportallocatorfactory.h b/talk/app/webrtc/fakeportallocatorfactory.h
deleted file mode 100644
index f326b62..0000000
--- a/talk/app/webrtc/fakeportallocatorfactory.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * libjingle
- * Copyright 2011 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-// This file defines a fake port allocator factory used for testing.
-// This implementation creates instances of cricket::FakePortAllocator.
-
-#ifndef TALK_APP_WEBRTC_FAKEPORTALLOCATORFACTORY_H_
-#define TALK_APP_WEBRTC_FAKEPORTALLOCATORFACTORY_H_
-
-#include "talk/app/webrtc/peerconnectioninterface.h"
-#include "webrtc/p2p/client/fakeportallocator.h"
-
-namespace webrtc {
-
-class FakePortAllocatorFactory : public PortAllocatorFactoryInterface {
- public:
- static FakePortAllocatorFactory* Create() {
- rtc::RefCountedObject<FakePortAllocatorFactory>* allocator =
- new rtc::RefCountedObject<FakePortAllocatorFactory>();
- return allocator;
- }
-
- virtual cricket::PortAllocator* CreatePortAllocator(
- const std::vector<StunConfiguration>& stun_configurations,
- const std::vector<TurnConfiguration>& turn_configurations) {
- stun_configs_ = stun_configurations;
- turn_configs_ = turn_configurations;
- return new cricket::FakePortAllocator(rtc::Thread::Current(), NULL);
- }
-
- const std::vector<StunConfiguration>& stun_configs() const {
- return stun_configs_;
- }
-
- const std::vector<TurnConfiguration>& turn_configs() const {
- return turn_configs_;
- }
-
- void SetNetworkIgnoreMask(int network_ignore_mask) {}
-
- protected:
- FakePortAllocatorFactory() {}
- ~FakePortAllocatorFactory() {}
-
- private:
- std::vector<PortAllocatorFactoryInterface::StunConfiguration> stun_configs_;
- std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turn_configs_;
-};
-
-} // namespace webrtc
-
-#endif // TALK_APP_WEBRTC_FAKEPORTALLOCATORFACTORY_H_
diff --git a/talk/app/webrtc/java/android/org/webrtc/Camera2Enumerator.java b/talk/app/webrtc/java/android/org/webrtc/Camera2Enumerator.java
index 097d1cd..3444529 100644
--- a/talk/app/webrtc/java/android/org/webrtc/Camera2Enumerator.java
+++ b/talk/app/webrtc/java/android/org/webrtc/Camera2Enumerator.java
@@ -27,7 +27,9 @@
package org.webrtc;
+import android.annotation.TargetApi;
import android.content.Context;
+
import android.graphics.ImageFormat;
import android.hardware.camera2.CameraCharacteristics;
import android.hardware.camera2.CameraManager;
@@ -45,6 +47,7 @@
import java.util.List;
import java.util.Map;
+@TargetApi(21)
public class Camera2Enumerator implements CameraEnumerationAndroid.Enumerator {
private final static String TAG = "Camera2Enumerator";
private final static double NANO_SECONDS_PER_SECOND = 1.0e9;
diff --git a/talk/app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java b/talk/app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
index 3e37f6a..5f68c37 100644
--- a/talk/app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
+++ b/talk/app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java
@@ -29,7 +29,6 @@
import static java.lang.Math.abs;
import static java.lang.Math.ceil;
-import android.hardware.Camera;
import android.graphics.ImageFormat;
import org.json.JSONArray;
@@ -72,7 +71,7 @@
// other image formats then this needs to be updated and
// VideoCapturerAndroid.getSupportedFormats need to return CaptureFormats of
// all imageFormats.
- public final int imageFormat = ImageFormat.YV12;
+ public final int imageFormat = ImageFormat.NV21;
public CaptureFormat(int width, int height, int minFramerate,
int maxFramerate) {
@@ -88,25 +87,15 @@
}
// Calculates the frame size of the specified image format. Currently only
- // supporting ImageFormat.YV12. The YV12's stride is the closest rounded up
- // multiple of 16 of the width and width and height are always even.
- // Android guarantees this:
- // http://developer.android.com/reference/android/hardware/Camera.Parameters.html#setPreviewFormat%28int%29
+ // supporting ImageFormat.NV21.
+ // The size is width * height * number of bytes per pixel.
+ // http://developer.android.com/reference/android/hardware/Camera.html#addCallbackBuffer(byte[])
public static int frameSize(int width, int height, int imageFormat) {
- if (imageFormat != ImageFormat.YV12) {
+ if (imageFormat != ImageFormat.NV21) {
throw new UnsupportedOperationException("Don't know how to calculate "
- + "the frame size of non-YV12 image formats.");
+ + "the frame size of non-NV21 image formats.");
}
- int yStride = roundUp(width, 16);
- int uvStride = roundUp(yStride / 2, 16);
- int ySize = yStride * height;
- int uvSize = uvStride * height / 2;
- return ySize + uvSize * 2;
- }
-
- // Rounds up |x| to the closest value that is a multiple of |alignment|.
- private static int roundUp(int x, int alignment) {
- return (int)ceil(x / (double)alignment) * alignment;
+ return (width * height * ImageFormat.getBitsPerPixel(imageFormat)) / 8;
}
@Override
@@ -114,21 +103,19 @@
return width + "x" + height + "@[" + minFramerate + ":" + maxFramerate + "]";
}
- @Override
- public boolean equals(Object that) {
- if (!(that instanceof CaptureFormat)) {
+ public boolean isSameFormat(final CaptureFormat that) {
+ if (that == null) {
return false;
}
- final CaptureFormat c = (CaptureFormat) that;
- return width == c.width && height == c.height && maxFramerate == c.maxFramerate
- && minFramerate == c.minFramerate;
+ return width == that.width && height == that.height && maxFramerate == that.maxFramerate
+ && minFramerate == that.minFramerate;
}
}
// Returns device names that can be used to create a new VideoCapturerAndroid.
public static String[] getDeviceNames() {
- String[] names = new String[Camera.getNumberOfCameras()];
- for (int i = 0; i < Camera.getNumberOfCameras(); ++i) {
+ String[] names = new String[android.hardware.Camera.getNumberOfCameras()];
+ for (int i = 0; i < android.hardware.Camera.getNumberOfCameras(); ++i) {
names[i] = getDeviceName(i);
}
return names;
@@ -136,22 +123,22 @@
// Returns number of cameras on device.
public static int getDeviceCount() {
- return Camera.getNumberOfCameras();
+ return android.hardware.Camera.getNumberOfCameras();
}
// Returns the name of the camera with camera index. Returns null if the
// camera can not be used.
public static String getDeviceName(int index) {
- Camera.CameraInfo info = new Camera.CameraInfo();
+ android.hardware.Camera.CameraInfo info = new android.hardware.Camera.CameraInfo();
try {
- Camera.getCameraInfo(index, info);
+ android.hardware.Camera.getCameraInfo(index, info);
} catch (Exception e) {
Logging.e(TAG, "getCameraInfo failed on index " + index,e);
return null;
}
String facing =
- (info.facing == Camera.CameraInfo.CAMERA_FACING_FRONT) ? "front" : "back";
+ (info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_FRONT) ? "front" : "back";
return "Camera " + index + ", Facing " + facing
+ ", Orientation " + info.orientation;
}
@@ -159,13 +146,13 @@
// Returns the name of the front facing camera. Returns null if the
// camera can not be used or does not exist.
public static String getNameOfFrontFacingDevice() {
- return getNameOfDevice(Camera.CameraInfo.CAMERA_FACING_FRONT);
+ return getNameOfDevice(android.hardware.Camera.CameraInfo.CAMERA_FACING_FRONT);
}
// Returns the name of the back facing camera. Returns null if the
// camera can not be used or does not exist.
public static String getNameOfBackFacingDevice() {
- return getNameOfDevice(Camera.CameraInfo.CAMERA_FACING_BACK);
+ return getNameOfDevice(android.hardware.Camera.CameraInfo.CAMERA_FACING_BACK);
}
public static String getSupportedFormatsAsJson(int id) throws JSONException {
@@ -194,7 +181,8 @@
}
}
- public static int[] getFramerateRange(Camera.Parameters parameters, final int framerate) {
+ public static int[] getFramerateRange(android.hardware.Camera.Parameters parameters,
+ final int framerate) {
List<int[]> listFpsRange = parameters.getSupportedPreviewFpsRange();
if (listFpsRange.isEmpty()) {
Logging.w(TAG, "No supported preview fps range");
@@ -203,27 +191,30 @@
return Collections.min(listFpsRange,
new ClosestComparator<int[]>() {
@Override int diff(int[] range) {
- return abs(framerate - range[Camera.Parameters.PREVIEW_FPS_MIN_INDEX])
- + abs(framerate - range[Camera.Parameters.PREVIEW_FPS_MAX_INDEX]);
+ final int maxFpsWeight = 10;
+ return range[android.hardware.Camera.Parameters.PREVIEW_FPS_MIN_INDEX]
+ + maxFpsWeight * abs(framerate
+ - range[android.hardware.Camera.Parameters.PREVIEW_FPS_MAX_INDEX]);
}
});
}
- public static Camera.Size getClosestSupportedSize(
- List<Camera.Size> supportedSizes, final int requestedWidth, final int requestedHeight) {
+ public static android.hardware.Camera.Size getClosestSupportedSize(
+ List<android.hardware.Camera.Size> supportedSizes, final int requestedWidth,
+ final int requestedHeight) {
return Collections.min(supportedSizes,
- new ClosestComparator<Camera.Size>() {
- @Override int diff(Camera.Size size) {
+ new ClosestComparator<android.hardware.Camera.Size>() {
+ @Override int diff(android.hardware.Camera.Size size) {
return abs(requestedWidth - size.width) + abs(requestedHeight - size.height);
}
});
}
private static String getNameOfDevice(int facing) {
- final Camera.CameraInfo info = new Camera.CameraInfo();
- for (int i = 0; i < Camera.getNumberOfCameras(); ++i) {
+ final android.hardware.Camera.CameraInfo info = new android.hardware.Camera.CameraInfo();
+ for (int i = 0; i < android.hardware.Camera.getNumberOfCameras(); ++i) {
try {
- Camera.getCameraInfo(i, info);
+ android.hardware.Camera.getCameraInfo(i, info);
if (info.facing == facing) {
return getDeviceName(i);
}
diff --git a/talk/app/webrtc/java/android/org/webrtc/CameraEnumerator.java b/talk/app/webrtc/java/android/org/webrtc/CameraEnumerator.java
index 2f35dc3..54469cc 100644
--- a/talk/app/webrtc/java/android/org/webrtc/CameraEnumerator.java
+++ b/talk/app/webrtc/java/android/org/webrtc/CameraEnumerator.java
@@ -27,7 +27,6 @@
package org.webrtc;
-import android.hardware.Camera;
import android.os.SystemClock;
import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
@@ -60,11 +59,11 @@
private List<CaptureFormat> enumerateFormats(int cameraId) {
Logging.d(TAG, "Get supported formats for camera index " + cameraId + ".");
final long startTimeMs = SystemClock.elapsedRealtime();
- final Camera.Parameters parameters;
- Camera camera = null;
+ final android.hardware.Camera.Parameters parameters;
+ android.hardware.Camera camera = null;
try {
Logging.d(TAG, "Opening camera with index " + cameraId);
- camera = Camera.open(cameraId);
+ camera = android.hardware.Camera.open(cameraId);
parameters = camera.getParameters();
} catch (RuntimeException e) {
Logging.e(TAG, "Open camera failed on camera index " + cameraId, e);
@@ -84,10 +83,10 @@
// getSupportedPreviewFpsRange() returns a sorted list. Take the fps range
// corresponding to the highest fps.
final int[] range = listFpsRange.get(listFpsRange.size() - 1);
- minFps = range[Camera.Parameters.PREVIEW_FPS_MIN_INDEX];
- maxFps = range[Camera.Parameters.PREVIEW_FPS_MAX_INDEX];
+ minFps = range[android.hardware.Camera.Parameters.PREVIEW_FPS_MIN_INDEX];
+ maxFps = range[android.hardware.Camera.Parameters.PREVIEW_FPS_MAX_INDEX];
}
- for (Camera.Size size : parameters.getSupportedPreviewSizes()) {
+ for (android.hardware.Camera.Size size : parameters.getSupportedPreviewSizes()) {
formatList.add(new CaptureFormat(size.width, size.height, minFps, maxFps));
}
} catch (Exception e) {
diff --git a/talk/app/webrtc/java/android/org/webrtc/EglBase.java b/talk/app/webrtc/java/android/org/webrtc/EglBase.java
index 2ee3688..035645b 100644
--- a/talk/app/webrtc/java/android/org/webrtc/EglBase.java
+++ b/talk/app/webrtc/java/android/org/webrtc/EglBase.java
@@ -28,244 +28,108 @@
package org.webrtc;
import android.graphics.SurfaceTexture;
-import android.view.SurfaceHolder;
-
-import org.webrtc.Logging;
+import android.view.Surface;
import javax.microedition.khronos.egl.EGL10;
-import javax.microedition.khronos.egl.EGLConfig;
-import javax.microedition.khronos.egl.EGLContext;
-import javax.microedition.khronos.egl.EGLDisplay;
-import javax.microedition.khronos.egl.EGLSurface;
+
/**
- * Holds EGL state and utility methods for handling an EGLContext, an EGLDisplay, and an EGLSurface.
+ * Holds EGL state and utility methods for handling an egl 1.0 EGLContext, an EGLDisplay,
+ * and an EGLSurface.
*/
-public final class EglBase {
- private static final String TAG = "EglBase";
+public abstract class EglBase {
+ // EGL wrapper for an actual EGLContext.
+ public static class Context {
+ }
+
// These constants are taken from EGL14.EGL_OPENGL_ES2_BIT and EGL14.EGL_CONTEXT_CLIENT_VERSION.
// https://android.googlesource.com/platform/frameworks/base/+/master/opengl/java/android/opengl/EGL14.java
// This is similar to how GlSurfaceView does:
// http://grepcode.com/file/repository.grepcode.com/java/ext/com.google.android/android/5.1.1_r1/android/opengl/GLSurfaceView.java#760
private static final int EGL_OPENGL_ES2_BIT = 4;
- private static final int EGL_CONTEXT_CLIENT_VERSION = 0x3098;
// Android-specific extension.
private static final int EGL_RECORDABLE_ANDROID = 0x3142;
- private final EGL10 egl;
- private EGLContext eglContext;
- private ConfigType configType;
- private EGLConfig eglConfig;
- private EGLDisplay eglDisplay;
- private EGLSurface eglSurface = EGL10.EGL_NO_SURFACE;
+ public static final int[] CONFIG_PLAIN = {
+ EGL10.EGL_RED_SIZE, 8,
+ EGL10.EGL_GREEN_SIZE, 8,
+ EGL10.EGL_BLUE_SIZE, 8,
+ EGL10.EGL_RENDERABLE_TYPE, EGL_OPENGL_ES2_BIT,
+ EGL10.EGL_NONE
+ };
+ public static final int[] CONFIG_RGBA = {
+ EGL10.EGL_RED_SIZE, 8,
+ EGL10.EGL_GREEN_SIZE, 8,
+ EGL10.EGL_BLUE_SIZE, 8,
+ EGL10.EGL_ALPHA_SIZE, 8,
+ EGL10.EGL_RENDERABLE_TYPE, EGL_OPENGL_ES2_BIT,
+ EGL10.EGL_NONE
+ };
+ public static final int[] CONFIG_PIXEL_BUFFER = {
+ EGL10.EGL_RED_SIZE, 8,
+ EGL10.EGL_GREEN_SIZE, 8,
+ EGL10.EGL_BLUE_SIZE, 8,
+ EGL10.EGL_RENDERABLE_TYPE, EGL_OPENGL_ES2_BIT,
+ EGL10.EGL_SURFACE_TYPE, EGL10.EGL_PBUFFER_BIT,
+ EGL10.EGL_NONE
+ };
+ public static final int[] CONFIG_PIXEL_RGBA_BUFFER = {
+ EGL10.EGL_RED_SIZE, 8,
+ EGL10.EGL_GREEN_SIZE, 8,
+ EGL10.EGL_BLUE_SIZE, 8,
+ EGL10.EGL_ALPHA_SIZE, 8,
+ EGL10.EGL_RENDERABLE_TYPE, EGL_OPENGL_ES2_BIT,
+ EGL10.EGL_SURFACE_TYPE, EGL10.EGL_PBUFFER_BIT,
+ EGL10.EGL_NONE
+ };
+ public static final int[] CONFIG_RECORDABLE = {
+ EGL10.EGL_RED_SIZE, 8,
+ EGL10.EGL_GREEN_SIZE, 8,
+ EGL10.EGL_BLUE_SIZE, 8,
+ EGL10.EGL_RENDERABLE_TYPE, EGL_OPENGL_ES2_BIT,
+ EGL_RECORDABLE_ANDROID, 1,
+ EGL10.EGL_NONE
+ };
- // EGLConfig constructor type. Influences eglChooseConfig arguments.
- public static enum ConfigType {
- // No special parameters.
- PLAIN,
- // Configures with EGL_SURFACE_TYPE = EGL_PBUFFER_BIT.
- PIXEL_BUFFER,
- // Configures with EGL_RECORDABLE_ANDROID = 1.
- // Discourages EGL from using pixel formats that cannot efficiently be
- // converted to something usable by the video encoder.
- RECORDABLE
+ // Create a new context with the specified config attributes, sharing data with sharedContext.
+ // |sharedContext| can be null.
+ public static EglBase create(Context sharedContext, int[] configAttributes) {
+ return (EglBase14.isEGL14Supported()
+ && (sharedContext == null || sharedContext instanceof EglBase14.Context))
+ ? new EglBase14((EglBase14.Context) sharedContext, configAttributes)
+ : new EglBase10((EglBase10.Context) sharedContext, configAttributes);
}
- // Create root context without any EGLSurface or parent EGLContext. This can be used for branching
- // new contexts that share data.
- public EglBase() {
- this(EGL10.EGL_NO_CONTEXT, ConfigType.PLAIN);
+ public static EglBase create() {
+ return create(null, CONFIG_PLAIN);
}
- // Create a new context with the specified config type, sharing data with sharedContext.
- public EglBase(EGLContext sharedContext, ConfigType configType) {
- this.egl = (EGL10) EGLContext.getEGL();
- this.configType = configType;
- eglDisplay = getEglDisplay();
- eglConfig = getEglConfig(eglDisplay, configType);
- eglContext = createEglContext(sharedContext, eglDisplay, eglConfig);
- }
-
- // Create EGLSurface from the Android SurfaceHolder.
- public void createSurface(SurfaceHolder surfaceHolder) {
- createSurfaceInternal(surfaceHolder);
- }
+ public abstract void createSurface(Surface surface);
// Create EGLSurface from the Android SurfaceTexture.
- public void createSurface(SurfaceTexture surfaceTexture) {
- createSurfaceInternal(surfaceTexture);
- }
-
- // Create EGLSurface from either a SurfaceHolder or a SurfaceTexture.
- private void createSurfaceInternal(Object nativeWindow) {
- if (!(nativeWindow instanceof SurfaceHolder) && !(nativeWindow instanceof SurfaceTexture)) {
- throw new IllegalStateException("Input must be either a SurfaceHolder or SurfaceTexture");
- }
- checkIsNotReleased();
- if (configType == ConfigType.PIXEL_BUFFER) {
- Logging.w(TAG, "This EGL context is configured for PIXEL_BUFFER, but uses regular Surface");
- }
- if (eglSurface != EGL10.EGL_NO_SURFACE) {
- throw new RuntimeException("Already has an EGLSurface");
- }
- int[] surfaceAttribs = {EGL10.EGL_NONE};
- eglSurface = egl.eglCreateWindowSurface(eglDisplay, eglConfig, nativeWindow, surfaceAttribs);
- if (eglSurface == EGL10.EGL_NO_SURFACE) {
- throw new RuntimeException("Failed to create window surface");
- }
- }
+ public abstract void createSurface(SurfaceTexture surfaceTexture);
// Create dummy 1x1 pixel buffer surface so the context can be made current.
- public void createDummyPbufferSurface() {
- createPbufferSurface(1, 1);
- }
+ public abstract void createDummyPbufferSurface();
- public void createPbufferSurface(int width, int height) {
- checkIsNotReleased();
- if (configType != ConfigType.PIXEL_BUFFER) {
- throw new RuntimeException(
- "This EGL context is not configured to use a pixel buffer: " + configType);
- }
- if (eglSurface != EGL10.EGL_NO_SURFACE) {
- throw new RuntimeException("Already has an EGLSurface");
- }
- int[] surfaceAttribs = {EGL10.EGL_WIDTH, width, EGL10.EGL_HEIGHT, height, EGL10.EGL_NONE};
- eglSurface = egl.eglCreatePbufferSurface(eglDisplay, eglConfig, surfaceAttribs);
- if (eglSurface == EGL10.EGL_NO_SURFACE) {
- throw new RuntimeException("Failed to create pixel buffer surface");
- }
- }
+ public abstract void createPbufferSurface(int width, int height);
- public EGLContext getContext() {
- return eglContext;
- }
+ public abstract Context getEglBaseContext();
- public boolean hasSurface() {
- return eglSurface != EGL10.EGL_NO_SURFACE;
- }
+ public abstract boolean hasSurface();
- public int surfaceWidth() {
- final int widthArray[] = new int[1];
- egl.eglQuerySurface(eglDisplay, eglSurface, EGL10.EGL_WIDTH, widthArray);
- return widthArray[0];
- }
+ public abstract int surfaceWidth();
- public int surfaceHeight() {
- final int heightArray[] = new int[1];
- egl.eglQuerySurface(eglDisplay, eglSurface, EGL10.EGL_HEIGHT, heightArray);
- return heightArray[0];
- }
+ public abstract int surfaceHeight();
- public void releaseSurface() {
- if (eglSurface != EGL10.EGL_NO_SURFACE) {
- egl.eglDestroySurface(eglDisplay, eglSurface);
- eglSurface = EGL10.EGL_NO_SURFACE;
- }
- }
+ public abstract void releaseSurface();
- private void checkIsNotReleased() {
- if (eglDisplay == EGL10.EGL_NO_DISPLAY || eglContext == EGL10.EGL_NO_CONTEXT
- || eglConfig == null) {
- throw new RuntimeException("This object has been released");
- }
- }
+ public abstract void release();
- public void release() {
- checkIsNotReleased();
- releaseSurface();
- detachCurrent();
- egl.eglDestroyContext(eglDisplay, eglContext);
- egl.eglTerminate(eglDisplay);
- eglContext = EGL10.EGL_NO_CONTEXT;
- eglDisplay = EGL10.EGL_NO_DISPLAY;
- eglConfig = null;
- }
-
- public void makeCurrent() {
- checkIsNotReleased();
- if (eglSurface == EGL10.EGL_NO_SURFACE) {
- throw new RuntimeException("No EGLSurface - can't make current");
- }
- if (!egl.eglMakeCurrent(eglDisplay, eglSurface, eglSurface, eglContext)) {
- throw new RuntimeException("eglMakeCurrent failed");
- }
- }
+ public abstract void makeCurrent();
// Detach the current EGL context, so that it can be made current on another thread.
- public void detachCurrent() {
- if (!egl.eglMakeCurrent(
- eglDisplay, EGL10.EGL_NO_SURFACE, EGL10.EGL_NO_SURFACE, EGL10.EGL_NO_CONTEXT)) {
- throw new RuntimeException("eglMakeCurrent failed");
- }
- }
+ public abstract void detachCurrent();
- public void swapBuffers() {
- checkIsNotReleased();
- if (eglSurface == EGL10.EGL_NO_SURFACE) {
- throw new RuntimeException("No EGLSurface - can't swap buffers");
- }
- egl.eglSwapBuffers(eglDisplay, eglSurface);
- }
-
- // Return an EGLDisplay, or die trying.
- private EGLDisplay getEglDisplay() {
- EGLDisplay eglDisplay = egl.eglGetDisplay(EGL10.EGL_DEFAULT_DISPLAY);
- if (eglDisplay == EGL10.EGL_NO_DISPLAY) {
- throw new RuntimeException("Unable to get EGL10 display");
- }
- int[] version = new int[2];
- if (!egl.eglInitialize(eglDisplay, version)) {
- throw new RuntimeException("Unable to initialize EGL10");
- }
- return eglDisplay;
- }
-
- // Return an EGLConfig, or die trying.
- private EGLConfig getEglConfig(EGLDisplay eglDisplay, ConfigType configType) {
- // Always RGB888, GLES2.
- int[] configAttributes = {
- EGL10.EGL_RED_SIZE, 8,
- EGL10.EGL_GREEN_SIZE, 8,
- EGL10.EGL_BLUE_SIZE, 8,
- EGL10.EGL_RENDERABLE_TYPE, EGL_OPENGL_ES2_BIT,
- EGL10.EGL_NONE, 0, // Allocate dummy fields for specific options.
- EGL10.EGL_NONE
- };
-
- // Fill in dummy fields based on configType.
- switch (configType) {
- case PLAIN:
- break;
- case PIXEL_BUFFER:
- configAttributes[configAttributes.length - 3] = EGL10.EGL_SURFACE_TYPE;
- configAttributes[configAttributes.length - 2] = EGL10.EGL_PBUFFER_BIT;
- break;
- case RECORDABLE:
- configAttributes[configAttributes.length - 3] = EGL_RECORDABLE_ANDROID;
- configAttributes[configAttributes.length - 2] = 1;
- break;
- default:
- throw new IllegalArgumentException();
- }
-
- EGLConfig[] configs = new EGLConfig[1];
- int[] numConfigs = new int[1];
- if (!egl.eglChooseConfig(
- eglDisplay, configAttributes, configs, configs.length, numConfigs)) {
- throw new RuntimeException("Unable to find RGB888 " + configType + " EGL config");
- }
- return configs[0];
- }
-
- // Return an EGLConfig, or die trying.
- private EGLContext createEglContext(
- EGLContext sharedContext, EGLDisplay eglDisplay, EGLConfig eglConfig) {
- int[] contextAttributes = {EGL_CONTEXT_CLIENT_VERSION, 2, EGL10.EGL_NONE};
- EGLContext eglContext =
- egl.eglCreateContext(eglDisplay, eglConfig, sharedContext, contextAttributes);
- if (eglContext == EGL10.EGL_NO_CONTEXT) {
- throw new RuntimeException("Failed to create EGL context");
- }
- return eglContext;
- }
+ public abstract void swapBuffers();
}
diff --git a/talk/app/webrtc/java/android/org/webrtc/EglBase10.java b/talk/app/webrtc/java/android/org/webrtc/EglBase10.java
new file mode 100644
index 0000000..f2aa985
--- /dev/null
+++ b/talk/app/webrtc/java/android/org/webrtc/EglBase10.java
@@ -0,0 +1,299 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+package org.webrtc;
+
+import android.graphics.Canvas;
+import android.graphics.SurfaceTexture;
+import android.graphics.Rect;
+import android.view.Surface;
+import android.view.SurfaceHolder;
+
+import javax.microedition.khronos.egl.EGL10;
+import javax.microedition.khronos.egl.EGLConfig;
+import javax.microedition.khronos.egl.EGLContext;
+import javax.microedition.khronos.egl.EGLDisplay;
+import javax.microedition.khronos.egl.EGLSurface;
+
+/**
+ * Holds EGL state and utility methods for handling an egl 1.0 EGLContext, an EGLDisplay,
+ * and an EGLSurface.
+ */
+final class EglBase10 extends EglBase {
+ // This constant is taken from EGL14.EGL_CONTEXT_CLIENT_VERSION.
+ private static final int EGL_CONTEXT_CLIENT_VERSION = 0x3098;
+
+ private final EGL10 egl;
+ private EGLContext eglContext;
+ private EGLConfig eglConfig;
+ private EGLDisplay eglDisplay;
+ private EGLSurface eglSurface = EGL10.EGL_NO_SURFACE;
+
+ // EGL wrapper for an actual EGLContext.
+ public static class Context extends EglBase.Context {
+ private final EGLContext eglContext;
+
+ public Context(EGLContext eglContext) {
+ this.eglContext = eglContext;
+ }
+ }
+
+ // Create a new context with the specified config type, sharing data with sharedContext.
+ EglBase10(Context sharedContext, int[] configAttributes) {
+ this.egl = (EGL10) EGLContext.getEGL();
+ eglDisplay = getEglDisplay();
+ eglConfig = getEglConfig(eglDisplay, configAttributes);
+ eglContext = createEglContext(sharedContext, eglDisplay, eglConfig);
+ }
+
+ @Override
+ public void createSurface(Surface surface) {
+ /**
+ * We have to wrap Surface in a SurfaceHolder because for some reason eglCreateWindowSurface
+ * couldn't actually take a Surface object until API 17. Older versions fortunately just call
+ * SurfaceHolder.getSurface(), so we'll do that. No other methods are relevant.
+ */
+ class FakeSurfaceHolder implements SurfaceHolder {
+ private final Surface surface;
+
+ FakeSurfaceHolder(Surface surface) {
+ this.surface = surface;
+ }
+
+ @Override
+ public void addCallback(Callback callback) {}
+
+ @Override
+ public void removeCallback(Callback callback) {}
+
+ @Override
+ public boolean isCreating() {
+ return false;
+ }
+
+ @Deprecated
+ @Override
+ public void setType(int i) {}
+
+ @Override
+ public void setFixedSize(int i, int i2) {}
+
+ @Override
+ public void setSizeFromLayout() {}
+
+ @Override
+ public void setFormat(int i) {}
+
+ @Override
+ public void setKeepScreenOn(boolean b) {}
+
+ @Override
+ public Canvas lockCanvas() {
+ return null;
+ }
+
+ @Override
+ public Canvas lockCanvas(Rect rect) {
+ return null;
+ }
+
+ @Override
+ public void unlockCanvasAndPost(Canvas canvas) {}
+
+ @Override
+ public Rect getSurfaceFrame() {
+ return null;
+ }
+
+ @Override
+ public Surface getSurface() {
+ return surface;
+ }
+ }
+
+ createSurfaceInternal(new FakeSurfaceHolder(surface));
+ }
+
+ // Create EGLSurface from the Android SurfaceTexture.
+ @Override
+ public void createSurface(SurfaceTexture surfaceTexture) {
+ createSurfaceInternal(surfaceTexture);
+ }
+
+ // Create EGLSurface from either a SurfaceHolder or a SurfaceTexture.
+ private void createSurfaceInternal(Object nativeWindow) {
+ if (!(nativeWindow instanceof SurfaceHolder) && !(nativeWindow instanceof SurfaceTexture)) {
+ throw new IllegalStateException("Input must be either a SurfaceHolder or SurfaceTexture");
+ }
+ checkIsNotReleased();
+ if (eglSurface != EGL10.EGL_NO_SURFACE) {
+ throw new RuntimeException("Already has an EGLSurface");
+ }
+ int[] surfaceAttribs = {EGL10.EGL_NONE};
+ eglSurface = egl.eglCreateWindowSurface(eglDisplay, eglConfig, nativeWindow, surfaceAttribs);
+ if (eglSurface == EGL10.EGL_NO_SURFACE) {
+ throw new RuntimeException("Failed to create window surface");
+ }
+ }
+
+ // Create dummy 1x1 pixel buffer surface so the context can be made current.
+ @Override
+ public void createDummyPbufferSurface() {
+ createPbufferSurface(1, 1);
+ }
+
+ @Override
+ public void createPbufferSurface(int width, int height) {
+ checkIsNotReleased();
+ if (eglSurface != EGL10.EGL_NO_SURFACE) {
+ throw new RuntimeException("Already has an EGLSurface");
+ }
+ int[] surfaceAttribs = {EGL10.EGL_WIDTH, width, EGL10.EGL_HEIGHT, height, EGL10.EGL_NONE};
+ eglSurface = egl.eglCreatePbufferSurface(eglDisplay, eglConfig, surfaceAttribs);
+ if (eglSurface == EGL10.EGL_NO_SURFACE) {
+ throw new RuntimeException("Failed to create pixel buffer surface");
+ }
+ }
+
+ @Override
+ public org.webrtc.EglBase.Context getEglBaseContext() {
+ return new EglBase10.Context(eglContext);
+ }
+
+ @Override
+ public boolean hasSurface() {
+ return eglSurface != EGL10.EGL_NO_SURFACE;
+ }
+
+ @Override
+ public int surfaceWidth() {
+ final int widthArray[] = new int[1];
+ egl.eglQuerySurface(eglDisplay, eglSurface, EGL10.EGL_WIDTH, widthArray);
+ return widthArray[0];
+ }
+
+ @Override
+ public int surfaceHeight() {
+ final int heightArray[] = new int[1];
+ egl.eglQuerySurface(eglDisplay, eglSurface, EGL10.EGL_HEIGHT, heightArray);
+ return heightArray[0];
+ }
+
+ @Override
+ public void releaseSurface() {
+ if (eglSurface != EGL10.EGL_NO_SURFACE) {
+ egl.eglDestroySurface(eglDisplay, eglSurface);
+ eglSurface = EGL10.EGL_NO_SURFACE;
+ }
+ }
+
+ private void checkIsNotReleased() {
+ if (eglDisplay == EGL10.EGL_NO_DISPLAY || eglContext == EGL10.EGL_NO_CONTEXT
+ || eglConfig == null) {
+ throw new RuntimeException("This object has been released");
+ }
+ }
+
+ @Override
+ public void release() {
+ checkIsNotReleased();
+ releaseSurface();
+ detachCurrent();
+ egl.eglDestroyContext(eglDisplay, eglContext);
+ egl.eglTerminate(eglDisplay);
+ eglContext = EGL10.EGL_NO_CONTEXT;
+ eglDisplay = EGL10.EGL_NO_DISPLAY;
+ eglConfig = null;
+ }
+
+ @Override
+ public void makeCurrent() {
+ checkIsNotReleased();
+ if (eglSurface == EGL10.EGL_NO_SURFACE) {
+ throw new RuntimeException("No EGLSurface - can't make current");
+ }
+ if (!egl.eglMakeCurrent(eglDisplay, eglSurface, eglSurface, eglContext)) {
+ throw new RuntimeException("eglMakeCurrent failed");
+ }
+ }
+
+ // Detach the current EGL context, so that it can be made current on another thread.
+ @Override
+ public void detachCurrent() {
+ if (!egl.eglMakeCurrent(
+ eglDisplay, EGL10.EGL_NO_SURFACE, EGL10.EGL_NO_SURFACE, EGL10.EGL_NO_CONTEXT)) {
+ throw new RuntimeException("eglMakeCurrent failed");
+ }
+ }
+
+ @Override
+ public void swapBuffers() {
+ checkIsNotReleased();
+ if (eglSurface == EGL10.EGL_NO_SURFACE) {
+ throw new RuntimeException("No EGLSurface - can't swap buffers");
+ }
+ egl.eglSwapBuffers(eglDisplay, eglSurface);
+ }
+
+ // Return an EGLDisplay, or die trying.
+ private EGLDisplay getEglDisplay() {
+ EGLDisplay eglDisplay = egl.eglGetDisplay(EGL10.EGL_DEFAULT_DISPLAY);
+ if (eglDisplay == EGL10.EGL_NO_DISPLAY) {
+ throw new RuntimeException("Unable to get EGL10 display");
+ }
+ int[] version = new int[2];
+ if (!egl.eglInitialize(eglDisplay, version)) {
+ throw new RuntimeException("Unable to initialize EGL10");
+ }
+ return eglDisplay;
+ }
+
+ // Return an EGLConfig, or die trying.
+ private EGLConfig getEglConfig(EGLDisplay eglDisplay, int[] configAttributes) {
+ EGLConfig[] configs = new EGLConfig[1];
+ int[] numConfigs = new int[1];
+ if (!egl.eglChooseConfig(
+ eglDisplay, configAttributes, configs, configs.length, numConfigs)) {
+ throw new RuntimeException("Unable to find any matching EGL config");
+ }
+ return configs[0];
+ }
+
+ // Return an EGLConfig, or die trying.
+ private EGLContext createEglContext(
+ Context sharedContext, EGLDisplay eglDisplay, EGLConfig eglConfig) {
+ int[] contextAttributes = {EGL_CONTEXT_CLIENT_VERSION, 2, EGL10.EGL_NONE};
+ EGLContext rootContext =
+ sharedContext == null ? EGL10.EGL_NO_CONTEXT : sharedContext.eglContext;
+ EGLContext eglContext =
+ egl.eglCreateContext(eglDisplay, eglConfig, rootContext, contextAttributes);
+ if (eglContext == EGL10.EGL_NO_CONTEXT) {
+ throw new RuntimeException("Failed to create EGL context");
+ }
+ return eglContext;
+ }
+}
diff --git a/talk/app/webrtc/java/android/org/webrtc/EglBase14.java b/talk/app/webrtc/java/android/org/webrtc/EglBase14.java
new file mode 100644
index 0000000..c6f98c3
--- /dev/null
+++ b/talk/app/webrtc/java/android/org/webrtc/EglBase14.java
@@ -0,0 +1,254 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+package org.webrtc;
+
+import android.annotation.TargetApi;
+import android.graphics.SurfaceTexture;
+import android.opengl.EGL14;
+import android.opengl.EGLConfig;
+import android.opengl.EGLContext;
+import android.opengl.EGLDisplay;
+import android.opengl.EGLExt;
+import android.opengl.EGLSurface;
+import android.view.Surface;
+
+import org.webrtc.Logging;
+
+/**
+ * Holds EGL state and utility methods for handling an EGL14 EGLContext, an EGLDisplay,
+ * and an EGLSurface.
+ */
+@TargetApi(18)
+final class EglBase14 extends EglBase {
+ private static final String TAG = "EglBase14";
+ private static final int EGLExt_SDK_VERSION = android.os.Build.VERSION_CODES.JELLY_BEAN_MR2;
+ private static final int CURRENT_SDK_VERSION = android.os.Build.VERSION.SDK_INT;
+ private EGLContext eglContext;
+ private EGLConfig eglConfig;
+ private EGLDisplay eglDisplay;
+ private EGLSurface eglSurface = EGL14.EGL_NO_SURFACE;
+
+ // EGL 1.4 is supported from API 17. But EGLExt that is used for setting presentation
+ // time stamp on a surface is supported from 18 so we require 18.
+ public static boolean isEGL14Supported() {
+ Logging.d(TAG, "SDK version: " + CURRENT_SDK_VERSION
+ + ". isEGL14Supported: " + (CURRENT_SDK_VERSION >= EGLExt_SDK_VERSION));
+ return (CURRENT_SDK_VERSION >= EGLExt_SDK_VERSION);
+ }
+
+ public static class Context extends EglBase.Context {
+ private final android.opengl.EGLContext egl14Context;
+
+ Context(android.opengl.EGLContext eglContext) {
+ this.egl14Context = eglContext;
+ }
+ }
+
+ // Create a new context with the specified config type, sharing data with sharedContext.
+ // |sharedContext| may be null.
+ EglBase14(EglBase14.Context sharedContext, int[] configAttributes) {
+ eglDisplay = getEglDisplay();
+ eglConfig = getEglConfig(eglDisplay, configAttributes);
+ eglContext = createEglContext(sharedContext, eglDisplay, eglConfig);
+ }
+
+ // Create EGLSurface from the Android Surface.
+ @Override
+ public void createSurface(Surface surface) {
+ createSurfaceInternal(surface);
+ }
+
+ // Create EGLSurface from the Android SurfaceTexture.
+ @Override
+ public void createSurface(SurfaceTexture surfaceTexture) {
+ createSurfaceInternal(surfaceTexture);
+ }
+
+ // Create EGLSurface from either Surface or SurfaceTexture.
+ private void createSurfaceInternal(Object surface) {
+ if (!(surface instanceof Surface) && !(surface instanceof SurfaceTexture)) {
+ throw new IllegalStateException("Input must be either a Surface or SurfaceTexture");
+ }
+ checkIsNotReleased();
+ if (eglSurface != EGL14.EGL_NO_SURFACE) {
+ throw new RuntimeException("Already has an EGLSurface");
+ }
+ int[] surfaceAttribs = {EGL14.EGL_NONE};
+ eglSurface = EGL14.eglCreateWindowSurface(eglDisplay, eglConfig, surface, surfaceAttribs, 0);
+ if (eglSurface == EGL14.EGL_NO_SURFACE) {
+ throw new RuntimeException("Failed to create window surface");
+ }
+ }
+
+ @Override
+ public void createDummyPbufferSurface() {
+ createPbufferSurface(1, 1);
+ }
+
+ @Override
+ public void createPbufferSurface(int width, int height) {
+ checkIsNotReleased();
+ if (eglSurface != EGL14.EGL_NO_SURFACE) {
+ throw new RuntimeException("Already has an EGLSurface");
+ }
+ int[] surfaceAttribs = {EGL14.EGL_WIDTH, width, EGL14.EGL_HEIGHT, height, EGL14.EGL_NONE};
+ eglSurface = EGL14.eglCreatePbufferSurface(eglDisplay, eglConfig, surfaceAttribs, 0);
+ if (eglSurface == EGL14.EGL_NO_SURFACE) {
+ throw new RuntimeException("Failed to create pixel buffer surface");
+ }
+ }
+
+ @Override
+ public Context getEglBaseContext() {
+ return new EglBase14.Context(eglContext);
+ }
+
+ @Override
+ public boolean hasSurface() {
+ return eglSurface != EGL14.EGL_NO_SURFACE;
+ }
+
+ @Override
+ public int surfaceWidth() {
+ final int widthArray[] = new int[1];
+ EGL14.eglQuerySurface(eglDisplay, eglSurface, EGL14.EGL_WIDTH, widthArray, 0);
+ return widthArray[0];
+ }
+
+ @Override
+ public int surfaceHeight() {
+ final int heightArray[] = new int[1];
+ EGL14.eglQuerySurface(eglDisplay, eglSurface, EGL14.EGL_HEIGHT, heightArray, 0);
+ return heightArray[0];
+ }
+
+ @Override
+ public void releaseSurface() {
+ if (eglSurface != EGL14.EGL_NO_SURFACE) {
+ EGL14.eglDestroySurface(eglDisplay, eglSurface);
+ eglSurface = EGL14.EGL_NO_SURFACE;
+ }
+ }
+
+ private void checkIsNotReleased() {
+ if (eglDisplay == EGL14.EGL_NO_DISPLAY || eglContext == EGL14.EGL_NO_CONTEXT
+ || eglConfig == null) {
+ throw new RuntimeException("This object has been released");
+ }
+ }
+
+ @Override
+ public void release() {
+ checkIsNotReleased();
+ releaseSurface();
+ detachCurrent();
+ EGL14.eglDestroyContext(eglDisplay, eglContext);
+ EGL14.eglReleaseThread();
+ EGL14.eglTerminate(eglDisplay);
+ eglContext = EGL14.EGL_NO_CONTEXT;
+ eglDisplay = EGL14.EGL_NO_DISPLAY;
+ eglConfig = null;
+ }
+
+ @Override
+ public void makeCurrent() {
+ checkIsNotReleased();
+ if (eglSurface == EGL14.EGL_NO_SURFACE) {
+ throw new RuntimeException("No EGLSurface - can't make current");
+ }
+ if (!EGL14.eglMakeCurrent(eglDisplay, eglSurface, eglSurface, eglContext)) {
+ throw new RuntimeException("eglMakeCurrent failed");
+ }
+ }
+
+ // Detach the current EGL context, so that it can be made current on another thread.
+ @Override
+ public void detachCurrent() {
+ if (!EGL14.eglMakeCurrent(
+ eglDisplay, EGL14.EGL_NO_SURFACE, EGL14.EGL_NO_SURFACE, EGL14.EGL_NO_CONTEXT)) {
+ throw new RuntimeException("eglMakeCurrent failed");
+ }
+ }
+
+ @Override
+ public void swapBuffers() {
+ checkIsNotReleased();
+ if (eglSurface == EGL14.EGL_NO_SURFACE) {
+ throw new RuntimeException("No EGLSurface - can't swap buffers");
+ }
+ EGL14.eglSwapBuffers(eglDisplay, eglSurface);
+ }
+
+ public void swapBuffers(long timeStampNs) {
+ checkIsNotReleased();
+ if (eglSurface == EGL14.EGL_NO_SURFACE) {
+ throw new RuntimeException("No EGLSurface - can't swap buffers");
+ }
+ // See https://android.googlesource.com/platform/frameworks/native/+/tools_r22.2/opengl/specs/EGL_ANDROID_presentation_time.txt
+ EGLExt.eglPresentationTimeANDROID(eglDisplay, eglSurface, timeStampNs);
+ EGL14.eglSwapBuffers(eglDisplay, eglSurface);
+ }
+
+ // Return an EGLDisplay, or die trying.
+ private static EGLDisplay getEglDisplay() {
+ EGLDisplay eglDisplay = EGL14.eglGetDisplay(EGL14.EGL_DEFAULT_DISPLAY);
+ if (eglDisplay == EGL14.EGL_NO_DISPLAY) {
+ throw new RuntimeException("Unable to get EGL14 display");
+ }
+ int[] version = new int[2];
+ if (!EGL14.eglInitialize(eglDisplay, version, 0, version, 1)) {
+ throw new RuntimeException("Unable to initialize EGL14");
+ }
+ return eglDisplay;
+ }
+
+ // Return an EGLConfig, or die trying.
+ private static EGLConfig getEglConfig(EGLDisplay eglDisplay, int[] configAttributes) {
+ EGLConfig[] configs = new EGLConfig[1];
+ int[] numConfigs = new int[1];
+ if (!EGL14.eglChooseConfig(
+ eglDisplay, configAttributes, 0, configs, 0, configs.length, numConfigs, 0)) {
+ throw new RuntimeException("Unable to find any matching EGL config");
+ }
+ return configs[0];
+ }
+
+ // Return an EGLConfig, or die trying.
+ private static EGLContext createEglContext(
+ EglBase14.Context sharedContext, EGLDisplay eglDisplay, EGLConfig eglConfig) {
+ int[] contextAttributes = {EGL14.EGL_CONTEXT_CLIENT_VERSION, 2, EGL14.EGL_NONE};
+ EGLContext rootContext =
+ sharedContext == null ? EGL14.EGL_NO_CONTEXT : sharedContext.egl14Context;
+ EGLContext eglContext =
+ EGL14.eglCreateContext(eglDisplay, eglConfig, rootContext, contextAttributes, 0);
+ if (eglContext == EGL14.EGL_NO_CONTEXT) {
+ throw new RuntimeException("Failed to create EGL context");
+ }
+ return eglContext;
+ }
+}
diff --git a/talk/app/webrtc/java/android/org/webrtc/GlRectDrawer.java b/talk/app/webrtc/java/android/org/webrtc/GlRectDrawer.java
index 2cb8af7..6d3d5d2 100644
--- a/talk/app/webrtc/java/android/org/webrtc/GlRectDrawer.java
+++ b/talk/app/webrtc/java/android/org/webrtc/GlRectDrawer.java
@@ -40,13 +40,13 @@
import java.util.Map;
/**
- * Helper class to draw a quad that covers the entire viewport. Rotation, mirror, and cropping is
- * specified using a 4x4 texture coordinate transform matrix. The frame input can either be an OES
- * texture or YUV textures in I420 format. The GL state must be preserved between draw calls, this
- * is intentional to maximize performance. The function release() must be called manually to free
- * the resources held by this object.
+ * Helper class to draw an opaque quad on the target viewport location. Rotation, mirror, and
+ * cropping is specified using a 4x4 texture coordinate transform matrix. The frame input can either
+ * be an OES texture or YUV textures in I420 format. The GL state must be preserved between draw
+ * calls, this is intentional to maximize performance. The function release() must be called
+ * manually to free the resources held by this object.
*/
-public class GlRectDrawer {
+public class GlRectDrawer implements RendererCommon.GlDrawer {
// Simple vertex shader, used for both YUV and OES.
private static final String VERTEX_SHADER_STRING =
"varying vec2 interp_tc;\n"
@@ -118,67 +118,31 @@
1.0f, 1.0f // Top right.
});
- // The keys are one of the fragments shaders above.
- private final Map<String, GlShader> shaders = new IdentityHashMap<String, GlShader>();
- private GlShader currentShader;
- private float[] currentTexMatrix;
- private int texMatrixLocation;
- // Intermediate copy buffer for uploading yuv frames that are not packed, i.e. stride > width.
- // TODO(magjed): Investigate when GL_UNPACK_ROW_LENGTH is available, or make a custom shader that
- // handles stride and compare performance with intermediate copy.
- private ByteBuffer copyBuffer;
+ private static class Shader {
+ public final GlShader glShader;
+ public final int texMatrixLocation;
- /**
- * Upload |planes| into |outputYuvTextures|, taking stride into consideration. |outputYuvTextures|
- * must have been generated in advance.
- */
- public void uploadYuvData(
- int[] outputYuvTextures, int width, int height, int[] strides, ByteBuffer[] planes) {
- // Make a first pass to see if we need a temporary copy buffer.
- int copyCapacityNeeded = 0;
- for (int i = 0; i < 3; ++i) {
- final int planeWidth = (i == 0) ? width : width / 2;
- final int planeHeight = (i == 0) ? height : height / 2;
- if (strides[i] > planeWidth) {
- copyCapacityNeeded = Math.max(copyCapacityNeeded, planeWidth * planeHeight);
- }
- }
- // Allocate copy buffer if necessary.
- if (copyCapacityNeeded > 0
- && (copyBuffer == null || copyBuffer.capacity() < copyCapacityNeeded)) {
- copyBuffer = ByteBuffer.allocateDirect(copyCapacityNeeded);
- }
- // Upload each plane.
- for (int i = 0; i < 3; ++i) {
- GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
- GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, outputYuvTextures[i]);
- final int planeWidth = (i == 0) ? width : width / 2;
- final int planeHeight = (i == 0) ? height : height / 2;
- // GLES only accepts packed data, i.e. stride == planeWidth.
- final ByteBuffer packedByteBuffer;
- if (strides[i] == planeWidth) {
- // Input is packed already.
- packedByteBuffer = planes[i];
- } else {
- VideoRenderer.nativeCopyPlane(
- planes[i], planeWidth, planeHeight, strides[i], copyBuffer, planeWidth);
- packedByteBuffer = copyBuffer;
- }
- GLES20.glTexImage2D(GLES20.GL_TEXTURE_2D, 0, GLES20.GL_LUMINANCE, planeWidth, planeHeight, 0,
- GLES20.GL_LUMINANCE, GLES20.GL_UNSIGNED_BYTE, packedByteBuffer);
+ public Shader(String fragmentShader) {
+ this.glShader = new GlShader(VERTEX_SHADER_STRING, fragmentShader);
+ this.texMatrixLocation = glShader.getUniformLocation("texMatrix");
}
}
+ // The keys are one of the fragments shaders above.
+ private final Map<String, Shader> shaders = new IdentityHashMap<String, Shader>();
+
/**
* Draw an OES texture frame with specified texture transformation matrix. Required resources are
* allocated at the first call to this function.
*/
- public void drawOes(int oesTextureId, float[] texMatrix) {
- prepareShader(OES_FRAGMENT_SHADER_STRING);
+ @Override
+ public void drawOes(int oesTextureId, float[] texMatrix, int x, int y, int width, int height) {
+ prepareShader(OES_FRAGMENT_SHADER_STRING, texMatrix);
+ GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
// updateTexImage() may be called from another thread in another EGL context, so we need to
// bind/unbind the texture in each draw call so that GLES understads it's a new texture.
GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, oesTextureId);
- drawRectangle(texMatrix);
+ drawRectangle(x, y, width, height);
GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, 0);
}
@@ -186,10 +150,12 @@
* Draw a RGB(A) texture frame with specified texture transformation matrix. Required resources
* are allocated at the first call to this function.
*/
- public void drawRgb(int textureId, float[] texMatrix) {
- prepareShader(RGB_FRAGMENT_SHADER_STRING);
+ @Override
+ public void drawRgb(int textureId, float[] texMatrix, int x, int y, int width, int height) {
+ prepareShader(RGB_FRAGMENT_SHADER_STRING, texMatrix);
+ GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, textureId);
- drawRectangle(texMatrix);
+ drawRectangle(x, y, width, height);
// Unbind the texture as a precaution.
GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, 0);
}
@@ -198,14 +164,15 @@
* Draw a YUV frame with specified texture transformation matrix. Required resources are
* allocated at the first call to this function.
*/
- public void drawYuv(int[] yuvTextures, float[] texMatrix) {
- prepareShader(YUV_FRAGMENT_SHADER_STRING);
+ @Override
+ public void drawYuv(int[] yuvTextures, float[] texMatrix, int x, int y, int width, int height) {
+ prepareShader(YUV_FRAGMENT_SHADER_STRING, texMatrix);
// Bind the textures.
for (int i = 0; i < 3; ++i) {
GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, yuvTextures[i]);
}
- drawRectangle(texMatrix);
+ drawRectangle(x, y, width, height);
// Unbind the textures as a precaution..
for (int i = 0; i < 3; ++i) {
GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
@@ -213,60 +180,51 @@
}
}
- private void drawRectangle(float[] texMatrix) {
- // Try avoid uploading the texture if possible.
- if (!Arrays.equals(currentTexMatrix, texMatrix)) {
- currentTexMatrix = texMatrix.clone();
- // Copy the texture transformation matrix over.
- GLES20.glUniformMatrix4fv(texMatrixLocation, 1, false, texMatrix, 0);
- }
+ private void drawRectangle(int x, int y, int width, int height) {
// Draw quad.
+ GLES20.glViewport(x, y, width, height);
GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
}
- private void prepareShader(String fragmentShader) {
- // Lazy allocation.
- if (!shaders.containsKey(fragmentShader)) {
- final GlShader shader = new GlShader(VERTEX_SHADER_STRING, fragmentShader);
+ private void prepareShader(String fragmentShader, float[] texMatrix) {
+ final Shader shader;
+ if (shaders.containsKey(fragmentShader)) {
+ shader = shaders.get(fragmentShader);
+ } else {
+ // Lazy allocation.
+ shader = new Shader(fragmentShader);
shaders.put(fragmentShader, shader);
- shader.useProgram();
+ shader.glShader.useProgram();
// Initialize fragment shader uniform values.
if (fragmentShader == YUV_FRAGMENT_SHADER_STRING) {
- GLES20.glUniform1i(shader.getUniformLocation("y_tex"), 0);
- GLES20.glUniform1i(shader.getUniformLocation("u_tex"), 1);
- GLES20.glUniform1i(shader.getUniformLocation("v_tex"), 2);
+ GLES20.glUniform1i(shader.glShader.getUniformLocation("y_tex"), 0);
+ GLES20.glUniform1i(shader.glShader.getUniformLocation("u_tex"), 1);
+ GLES20.glUniform1i(shader.glShader.getUniformLocation("v_tex"), 2);
} else if (fragmentShader == RGB_FRAGMENT_SHADER_STRING) {
- GLES20.glUniform1i(shader.getUniformLocation("rgb_tex"), 0);
+ GLES20.glUniform1i(shader.glShader.getUniformLocation("rgb_tex"), 0);
} else if (fragmentShader == OES_FRAGMENT_SHADER_STRING) {
- GLES20.glUniform1i(shader.getUniformLocation("oes_tex"), 0);
+ GLES20.glUniform1i(shader.glShader.getUniformLocation("oes_tex"), 0);
} else {
throw new IllegalStateException("Unknown fragment shader: " + fragmentShader);
}
GlUtil.checkNoGLES2Error("Initialize fragment shader uniform values.");
// Initialize vertex shader attributes.
- shader.setVertexAttribArray("in_pos", 2, FULL_RECTANGLE_BUF);
- shader.setVertexAttribArray("in_tc", 2, FULL_RECTANGLE_TEX_BUF);
+ shader.glShader.setVertexAttribArray("in_pos", 2, FULL_RECTANGLE_BUF);
+ shader.glShader.setVertexAttribArray("in_tc", 2, FULL_RECTANGLE_TEX_BUF);
}
-
- // Update GLES state if shader is not already current.
- final GlShader shader = shaders.get(fragmentShader);
- if (currentShader != shader) {
- currentShader = shader;
- shader.useProgram();
- GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
- currentTexMatrix = null;
- texMatrixLocation = shader.getUniformLocation("texMatrix");
- }
+ shader.glShader.useProgram();
+ // Copy the texture transformation matrix over.
+ GLES20.glUniformMatrix4fv(shader.texMatrixLocation, 1, false, texMatrix, 0);
}
/**
* Release all GLES resources. This needs to be done manually, otherwise the resources are leaked.
*/
+ @Override
public void release() {
- for (GlShader shader : shaders.values()) {
- shader.release();
+ for (Shader shader : shaders.values()) {
+ shader.glShader.release();
}
shaders.clear();
- copyBuffer = null;
}
}
diff --git a/talk/app/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.java b/talk/app/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.java
index e3a7850..950dcdf 100644
--- a/talk/app/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.java
+++ b/talk/app/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.java
@@ -55,7 +55,7 @@
* ACCESS_NETWORK_STATE permission.
*/
public class NetworkMonitorAutoDetect extends BroadcastReceiver {
- static enum ConnectionType {
+ public static enum ConnectionType {
CONNECTION_UNKNOWN,
CONNECTION_ETHERNET,
CONNECTION_WIFI,
@@ -96,6 +96,10 @@
/** Queries the ConnectivityManager for information about the current connection. */
static class ConnectivityManagerDelegate {
+ /**
+ * Note: In some rare Android systems connectivityManager is null. We handle that
+ * gracefully below.
+ */
private final ConnectivityManager connectivityManager;
ConnectivityManagerDelegate(Context context) {
@@ -114,6 +118,9 @@
* default network.
*/
NetworkState getNetworkState() {
+ if (connectivityManager == null) {
+ return new NetworkState(false, -1, -1);
+ }
return getNetworkState(connectivityManager.getActiveNetworkInfo());
}
@@ -123,6 +130,9 @@
*/
@SuppressLint("NewApi")
NetworkState getNetworkState(Network network) {
+ if (connectivityManager == null) {
+ return new NetworkState(false, -1, -1);
+ }
return getNetworkState(connectivityManager.getNetworkInfo(network));
}
@@ -142,6 +152,9 @@
*/
@SuppressLint("NewApi")
Network[] getAllNetworks() {
+ if (connectivityManager == null) {
+ return new Network[0];
+ }
return connectivityManager.getAllNetworks();
}
@@ -152,6 +165,9 @@
*/
@SuppressLint("NewApi")
int getDefaultNetId() {
+ if (connectivityManager == null) {
+ return INVALID_NET_ID;
+ }
// Android Lollipop had no API to get the default network; only an
// API to return the NetworkInfo for the default network. To
// determine the default network one can find the network with
@@ -188,6 +204,9 @@
*/
@SuppressLint("NewApi")
boolean hasInternetCapability(Network network) {
+ if (connectivityManager == null) {
+ return false;
+ }
final NetworkCapabilities capabilities =
connectivityManager.getNetworkCapabilities(network);
return capabilities != null && capabilities.hasCapability(NET_CAPABILITY_INTERNET);
@@ -240,7 +259,6 @@
static final int INVALID_NET_ID = -1;
private static final String TAG = "NetworkMonitorAutoDetect";
- private static final int UNKNOWN_LINK_SPEED = -1;
private final IntentFilter intentFilter;
// Observer for the connection type change.
diff --git a/talk/app/webrtc/java/android/org/webrtc/RendererCommon.java b/talk/app/webrtc/java/android/org/webrtc/RendererCommon.java
index 94d180d..5ada4cc 100644
--- a/talk/app/webrtc/java/android/org/webrtc/RendererCommon.java
+++ b/talk/app/webrtc/java/android/org/webrtc/RendererCommon.java
@@ -28,8 +28,11 @@
package org.webrtc;
import android.graphics.Point;
+import android.opengl.GLES20;
import android.opengl.Matrix;
+import java.nio.ByteBuffer;
+
/**
* Static helper functions for renderer implementations.
*/
@@ -47,6 +50,73 @@
public void onFrameResolutionChanged(int videoWidth, int videoHeight, int rotation);
}
+ /** Interface for rendering frames on an EGLSurface. */
+ public static interface GlDrawer {
+ /**
+ * Functions for drawing frames with different sources. The rendering surface target is
+ * implied by the current EGL context of the calling thread and requires no explicit argument.
+ * The coordinates specify the viewport location on the surface target.
+ */
+ void drawOes(int oesTextureId, float[] texMatrix, int x, int y, int width, int height);
+ void drawRgb(int textureId, float[] texMatrix, int x, int y, int width, int height);
+ void drawYuv(int[] yuvTextures, float[] texMatrix, int x, int y, int width, int height);
+
+ /**
+ * Release all GL resources. This needs to be done manually, otherwise resources may leak.
+ */
+ void release();
+ }
+
+ /**
+ * Helper class for uploading YUV bytebuffer frames to textures that handles stride > width. This
+ * class keeps an internal ByteBuffer to avoid unnecessary allocations for intermediate copies.
+ */
+ public static class YuvUploader {
+ // Intermediate copy buffer for uploading yuv frames that are not packed, i.e. stride > width.
+ // TODO(magjed): Investigate when GL_UNPACK_ROW_LENGTH is available, or make a custom shader
+ // that handles stride and compare performance with intermediate copy.
+ private ByteBuffer copyBuffer;
+
+ /**
+ * Upload |planes| into |outputYuvTextures|, taking stride into consideration.
+ * |outputYuvTextures| must have been generated in advance.
+ */
+ public void uploadYuvData(
+ int[] outputYuvTextures, int width, int height, int[] strides, ByteBuffer[] planes) {
+ final int[] planeWidths = new int[] {width, width / 2, width / 2};
+ final int[] planeHeights = new int[] {height, height / 2, height / 2};
+ // Make a first pass to see if we need a temporary copy buffer.
+ int copyCapacityNeeded = 0;
+ for (int i = 0; i < 3; ++i) {
+ if (strides[i] > planeWidths[i]) {
+ copyCapacityNeeded = Math.max(copyCapacityNeeded, planeWidths[i] * planeHeights[i]);
+ }
+ }
+ // Allocate copy buffer if necessary.
+ if (copyCapacityNeeded > 0
+ && (copyBuffer == null || copyBuffer.capacity() < copyCapacityNeeded)) {
+ copyBuffer = ByteBuffer.allocateDirect(copyCapacityNeeded);
+ }
+ // Upload each plane.
+ for (int i = 0; i < 3; ++i) {
+ GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
+ GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, outputYuvTextures[i]);
+ // GLES only accepts packed data, i.e. stride == planeWidth.
+ final ByteBuffer packedByteBuffer;
+ if (strides[i] == planeWidths[i]) {
+ // Input is packed already.
+ packedByteBuffer = planes[i];
+ } else {
+ VideoRenderer.nativeCopyPlane(
+ planes[i], planeWidths[i], planeHeights[i], strides[i], copyBuffer, planeWidths[i]);
+ packedByteBuffer = copyBuffer;
+ }
+ GLES20.glTexImage2D(GLES20.GL_TEXTURE_2D, 0, GLES20.GL_LUMINANCE, planeWidths[i],
+ planeHeights[i], 0, GLES20.GL_LUMINANCE, GLES20.GL_UNSIGNED_BYTE, packedByteBuffer);
+ }
+ }
+ }
+
// Types of video scaling:
// SCALE_ASPECT_FIT - video frame is scaled to fit the size of the view by
// maintaining the aspect ratio (black borders may be displayed).
@@ -182,9 +252,9 @@
}
// Each dimension is constrained on max display size and how much we are allowed to crop.
final int width = Math.min(maxDisplayWidth,
- (int) (maxDisplayHeight / minVisibleFraction * videoAspectRatio));
+ Math.round(maxDisplayHeight / minVisibleFraction * videoAspectRatio));
final int height = Math.min(maxDisplayHeight,
- (int) (maxDisplayWidth / minVisibleFraction / videoAspectRatio));
+ Math.round(maxDisplayWidth / minVisibleFraction / videoAspectRatio));
return new Point(width, height);
}
}
diff --git a/talk/app/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java b/talk/app/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
index b9c158f..b001d2a 100644
--- a/talk/app/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
+++ b/talk/app/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java
@@ -35,12 +35,12 @@
import android.os.HandlerThread;
import android.os.SystemClock;
+import java.nio.ByteBuffer;
+import java.nio.FloatBuffer;
import java.util.concurrent.Callable;
import java.util.concurrent.CountDownLatch;
import java.util.concurrent.TimeUnit;
-import javax.microedition.khronos.egl.EGLContext;
-
/**
* Helper class to create and synchronize access to a SurfaceTexture. The caller will get notified
* of new frames in onTextureFrameAvailable(), and should call returnTextureFrame() when done with
@@ -51,7 +51,7 @@
* wrapping texture frames into webrtc::VideoFrames and also handles calling returnTextureFrame()
* when the webrtc::VideoFrame is no longer used.
*/
-final class SurfaceTextureHelper {
+class SurfaceTextureHelper {
private static final String TAG = "SurfaceTextureHelper";
/**
* Callback interface for being notified that a new texture frame is available. The calls will be
@@ -65,7 +65,7 @@
int oesTextureId, float[] transformMatrix, long timestampNs);
}
- public static SurfaceTextureHelper create(EGLContext sharedContext) {
+ public static SurfaceTextureHelper create(EglBase.Context sharedContext) {
return create(sharedContext, null);
}
@@ -74,7 +74,8 @@
* |handler| is non-null, the callback will be executed on that handler's thread. If |handler| is
* null, a dedicated private thread is created for the callbacks.
*/
- public static SurfaceTextureHelper create(final EGLContext sharedContext, final Handler handler) {
+ public static SurfaceTextureHelper create(final EglBase.Context sharedContext,
+ final Handler handler) {
final Handler finalHandler;
if (handler != null) {
finalHandler = handler;
@@ -94,25 +95,240 @@
});
}
+ // State for YUV conversion, instantiated on demand.
+ static private class YuvConverter {
+ private final EglBase eglBase;
+ private final GlShader shader;
+ private boolean released = false;
+
+ // Vertex coordinates in Normalized Device Coordinates, i.e.
+ // (-1, -1) is bottom-left and (1, 1) is top-right.
+ private static final FloatBuffer DEVICE_RECTANGLE =
+ GlUtil.createFloatBuffer(new float[] {
+ -1.0f, -1.0f, // Bottom left.
+ 1.0f, -1.0f, // Bottom right.
+ -1.0f, 1.0f, // Top left.
+ 1.0f, 1.0f, // Top right.
+ });
+
+ // Texture coordinates - (0, 0) is bottom-left and (1, 1) is top-right.
+ private static final FloatBuffer TEXTURE_RECTANGLE =
+ GlUtil.createFloatBuffer(new float[] {
+ 0.0f, 0.0f, // Bottom left.
+ 1.0f, 0.0f, // Bottom right.
+ 0.0f, 1.0f, // Top left.
+ 1.0f, 1.0f // Top right.
+ });
+
+ private static final String VERTEX_SHADER =
+ "varying vec2 interp_tc;\n"
+ + "attribute vec4 in_pos;\n"
+ + "attribute vec4 in_tc;\n"
+ + "\n"
+ + "uniform mat4 texMatrix;\n"
+ + "\n"
+ + "void main() {\n"
+ + " gl_Position = in_pos;\n"
+ + " interp_tc = (texMatrix * in_tc).xy;\n"
+ + "}\n";
+
+ private static final String FRAGMENT_SHADER =
+ "#extension GL_OES_EGL_image_external : require\n"
+ + "precision mediump float;\n"
+ + "varying vec2 interp_tc;\n"
+ + "\n"
+ + "uniform samplerExternalOES oesTex;\n"
+ // Difference in texture coordinate corresponding to one
+ // sub-pixel in the x direction.
+ + "uniform vec2 xUnit;\n"
+ // Color conversion coefficients, including constant term
+ + "uniform vec4 coeffs;\n"
+ + "\n"
+ + "void main() {\n"
+ // Since the alpha read from the texture is always 1, this could
+ // be written as a mat4 x vec4 multiply. However, that seems to
+ // give a worse framerate, possibly because the additional
+ // multiplies by 1.0 consume resources. TODO(nisse): Could also
+ // try to do it as a vec3 x mat3x4, followed by an add in of a
+ // constant vector.
+ + " gl_FragColor.r = coeffs.a + dot(coeffs.rgb,\n"
+ + " texture2D(oesTex, interp_tc - 1.5 * xUnit).rgb);\n"
+ + " gl_FragColor.g = coeffs.a + dot(coeffs.rgb,\n"
+ + " texture2D(oesTex, interp_tc - 0.5 * xUnit).rgb);\n"
+ + " gl_FragColor.b = coeffs.a + dot(coeffs.rgb,\n"
+ + " texture2D(oesTex, interp_tc + 0.5 * xUnit).rgb);\n"
+ + " gl_FragColor.a = coeffs.a + dot(coeffs.rgb,\n"
+ + " texture2D(oesTex, interp_tc + 1.5 * xUnit).rgb);\n"
+ + "}\n";
+
+ private int texMatrixLoc;
+ private int xUnitLoc;
+ private int coeffsLoc;;
+
+ YuvConverter (EglBase.Context sharedContext) {
+ eglBase = EglBase.create(sharedContext, EglBase.CONFIG_PIXEL_RGBA_BUFFER);
+ eglBase.createDummyPbufferSurface();
+ eglBase.makeCurrent();
+
+ shader = new GlShader(VERTEX_SHADER, FRAGMENT_SHADER);
+ shader.useProgram();
+ texMatrixLoc = shader.getUniformLocation("texMatrix");
+ xUnitLoc = shader.getUniformLocation("xUnit");
+ coeffsLoc = shader.getUniformLocation("coeffs");
+ GLES20.glUniform1i(shader.getUniformLocation("oesTex"), 0);
+ GlUtil.checkNoGLES2Error("Initialize fragment shader uniform values.");
+ // Initialize vertex shader attributes.
+ shader.setVertexAttribArray("in_pos", 2, DEVICE_RECTANGLE);
+ // If the width is not a multiple of 4 pixels, the texture
+ // will be scaled up slightly and clipped at the right border.
+ shader.setVertexAttribArray("in_tc", 2, TEXTURE_RECTANGLE);
+ eglBase.detachCurrent();
+ }
+
+ synchronized void convert(ByteBuffer buf,
+ int width, int height, int stride, int textureId, float [] transformMatrix) {
+ if (released) {
+ throw new IllegalStateException(
+ "YuvConverter.convert called on released object");
+ }
+
+ // We draw into a buffer laid out like
+ //
+ // +---------+
+ // | |
+ // | Y |
+ // | |
+ // | |
+ // +----+----+
+ // | U | V |
+ // | | |
+ // +----+----+
+ //
+ // In memory, we use the same stride for all of Y, U and V. The
+ // U data starts at offset |height| * |stride| from the Y data,
+ // and the V data starts at at offset |stride/2| from the U
+ // data, with rows of U and V data alternating.
+ //
+ // Now, it would have made sense to allocate a pixel buffer with
+ // a single byte per pixel (EGL10.EGL_COLOR_BUFFER_TYPE,
+ // EGL10.EGL_LUMINANCE_BUFFER,), but that seems to be
+ // unsupported by devices. So do the following hack: Allocate an
+ // RGBA buffer, of width |stride|/4. To render each of these
+ // large pixels, sample the texture at 4 different x coordinates
+ // and store the results in the four components.
+ //
+ // Since the V data needs to start on a boundary of such a
+ // larger pixel, it is not sufficient that |stride| is even, it
+ // has to be a multiple of 8 pixels.
+
+ if (stride % 8 != 0) {
+ throw new IllegalArgumentException(
+ "Invalid stride, must be a multiple of 8");
+ }
+ if (stride < width){
+ throw new IllegalArgumentException(
+ "Invalid stride, must >= width");
+ }
+
+ int y_width = (width+3) / 4;
+ int uv_width = (width+7) / 8;
+ int uv_height = (height+1)/2;
+ int total_height = height + uv_height;
+ int size = stride * total_height;
+
+ if (buf.capacity() < size) {
+ throw new IllegalArgumentException("YuvConverter.convert called with too small buffer");
+ }
+ // Produce a frame buffer starting at top-left corner, not
+ // bottom-left.
+ transformMatrix =
+ RendererCommon.multiplyMatrices(transformMatrix,
+ RendererCommon.verticalFlipMatrix());
+
+ // Create new pBuffferSurface with the correct size if needed.
+ if (eglBase.hasSurface()) {
+ if (eglBase.surfaceWidth() != stride/4 ||
+ eglBase.surfaceHeight() != total_height){
+ eglBase.releaseSurface();
+ eglBase.createPbufferSurface(stride/4, total_height);
+ }
+ } else {
+ eglBase.createPbufferSurface(stride/4, total_height);
+ }
+
+ eglBase.makeCurrent();
+
+ GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
+ GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, textureId);
+ GLES20.glUniformMatrix4fv(texMatrixLoc, 1, false, transformMatrix, 0);
+
+ // Draw Y
+ GLES20.glViewport(0, 0, y_width, height);
+ // Matrix * (1;0;0;0) / width. Note that opengl uses column major order.
+ GLES20.glUniform2f(xUnitLoc,
+ transformMatrix[0] / width,
+ transformMatrix[1] / width);
+ // Y'UV444 to RGB888, see
+ // https://en.wikipedia.org/wiki/YUV#Y.27UV444_to_RGB888_conversion.
+ // We use the ITU-R coefficients for U and V */
+ GLES20.glUniform4f(coeffsLoc, 0.299f, 0.587f, 0.114f, 0.0f);
+ GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
+
+ // Draw U
+ GLES20.glViewport(0, height, uv_width, uv_height);
+ // Matrix * (1;0;0;0) / (2*width). Note that opengl uses column major order.
+ GLES20.glUniform2f(xUnitLoc,
+ transformMatrix[0] / (2.0f*width),
+ transformMatrix[1] / (2.0f*width));
+ GLES20.glUniform4f(coeffsLoc, -0.169f, -0.331f, 0.499f, 0.5f);
+ GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
+
+ // Draw V
+ GLES20.glViewport(stride/8, height, uv_width, uv_height);
+ GLES20.glUniform4f(coeffsLoc, 0.499f, -0.418f, -0.0813f, 0.5f);
+ GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
+
+ GLES20.glReadPixels(0, 0, stride/4, total_height, GLES20.GL_RGBA,
+ GLES20.GL_UNSIGNED_BYTE, buf);
+
+ GlUtil.checkNoGLES2Error("YuvConverter.convert");
+
+ // Unbind texture. Reportedly needed on some devices to get
+ // the texture updated from the camera.
+ GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, 0);
+ eglBase.detachCurrent();
+ }
+
+ synchronized void release() {
+ released = true;
+ eglBase.makeCurrent();
+ shader.release();
+ eglBase.release();
+ }
+ }
+
private final Handler handler;
- private final boolean isOwningThread;
+ private boolean isOwningThread;
private final EglBase eglBase;
private final SurfaceTexture surfaceTexture;
private final int oesTextureId;
+ private YuvConverter yuvConverter;
+
private OnTextureFrameAvailableListener listener;
// The possible states of this class.
private boolean hasPendingTexture = false;
- private boolean isTextureInUse = false;
+ private volatile boolean isTextureInUse = false;
private boolean isQuitting = false;
- private SurfaceTextureHelper(EGLContext sharedContext, Handler handler, boolean isOwningThread) {
+ private SurfaceTextureHelper(EglBase.Context sharedContext,
+ Handler handler, boolean isOwningThread) {
if (handler.getLooper().getThread() != Thread.currentThread()) {
throw new IllegalStateException("SurfaceTextureHelper must be created on the handler thread");
}
this.handler = handler;
this.isOwningThread = isOwningThread;
- eglBase = new EglBase(sharedContext, EglBase.ConfigType.PIXEL_BUFFER);
+ eglBase = EglBase.create(sharedContext, EglBase.CONFIG_PIXEL_BUFFER);
eglBase.createDummyPbufferSurface();
eglBase.makeCurrent();
@@ -120,6 +336,18 @@
surfaceTexture = new SurfaceTexture(oesTextureId);
}
+ private YuvConverter getYuvConverter() {
+ // yuvConverter is assigned once
+ if (yuvConverter != null)
+ return yuvConverter;
+
+ synchronized(this) {
+ if (yuvConverter == null)
+ yuvConverter = new YuvConverter(eglBase.getEglBaseContext());
+ return yuvConverter;
+ }
+ }
+
/**
* Start to stream textures to the given |listener|.
* A Listener can only be set once.
@@ -164,12 +392,19 @@
});
}
+ public boolean isTextureInUse() {
+ return isTextureInUse;
+ }
+
/**
* Call disconnect() to stop receiving frames. Resources are released when the texture frame has
* been returned by a call to returnTextureFrame(). You are guaranteed to not receive any more
* onTextureFrameAvailable() after this function returns.
*/
public void disconnect() {
+ if (!isOwningThread) {
+ throw new IllegalStateException("Must call disconnect(handler).");
+ }
if (handler.getLooper().getThread() == Thread.currentThread()) {
isQuitting = true;
if (!isTextureInUse) {
@@ -190,6 +425,28 @@
ThreadUtils.awaitUninterruptibly(barrier);
}
+ /**
+ * Call disconnect() to stop receiving frames and quit the looper used by |handler|.
+ * Resources are released when the texture frame has been returned by a call to
+ * returnTextureFrame(). You are guaranteed to not receive any more
+ * onTextureFrameAvailable() after this function returns.
+ */
+ public void disconnect(Handler handler) {
+ if (this.handler != handler) {
+ throw new IllegalStateException("Wrong handler.");
+ }
+ isOwningThread = true;
+ disconnect();
+ }
+
+ public void textureToYUV(ByteBuffer buf,
+ int width, int height, int stride, int textureId, float [] transformMatrix) {
+ if (textureId != oesTextureId)
+ throw new IllegalStateException("textureToByteBuffer called with unexpected textureId");
+
+ getYuvConverter().convert(buf, width, height, stride, textureId, transformMatrix);
+ }
+
private void tryDeliverTextureFrame() {
if (handler.getLooper().getThread() != Thread.currentThread()) {
throw new IllegalStateException("Wrong thread.");
@@ -218,12 +475,14 @@
if (isTextureInUse || !isQuitting) {
throw new IllegalStateException("Unexpected release.");
}
+ synchronized (this) {
+ if (yuvConverter != null)
+ yuvConverter.release();
+ }
eglBase.makeCurrent();
GLES20.glDeleteTextures(1, new int[] {oesTextureId}, 0);
surfaceTexture.release();
eglBase.release();
- if (isOwningThread) {
- handler.getLooper().quit();
- }
+ handler.getLooper().quit();
}
}
diff --git a/talk/app/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java b/talk/app/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
index d7c9e2a..fa199b3 100644
--- a/talk/app/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
+++ b/talk/app/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java
@@ -28,10 +28,9 @@
package org.webrtc;
import android.content.Context;
+import android.content.res.Resources.NotFoundException;
import android.graphics.Point;
-import android.graphics.SurfaceTexture;
import android.opengl.GLES20;
-import android.opengl.Matrix;
import android.os.Handler;
import android.os.HandlerThread;
import android.util.AttributeSet;
@@ -67,7 +66,8 @@
// EGL and GL resources for drawing YUV/OES textures. After initilization, these are only accessed
// from the render thread.
private EglBase eglBase;
- private GlRectDrawer drawer;
+ private final RendererCommon.YuvUploader yuvUploader = new RendererCommon.YuvUploader();
+ private RendererCommon.GlDrawer drawer;
// Texture ids for YUV frames. Allocated on first arrival of a YUV frame.
private int[] yuvTextures = null;
@@ -77,23 +77,22 @@
// These variables are synchronized on |layoutLock|.
private final Object layoutLock = new Object();
- // These three different dimension values are used to keep track of the state in these functions:
- // requestLayout() -> onMeasure() -> onLayout() -> surfaceChanged().
- // requestLayout() is triggered internally by frame size changes, but can also be triggered
- // externally by layout update requests.
- // Most recent measurement specification from onMeasure().
- private int widthSpec;
- private int heightSpec;
- // Current size on screen in pixels. Updated in onLayout(), and should be consistent with
- // |widthSpec|/|heightSpec| after that.
- private int layoutWidth;
- private int layoutHeight;
- // Current surface size of the underlying Surface. Updated in surfaceChanged(), and should be
- // consistent with |layoutWidth|/|layoutHeight| after that.
+ // These dimension values are used to keep track of the state in these functions: onMeasure(),
+ // onLayout(), and surfaceChanged(). A new layout is triggered with requestLayout(). This happens
+ // internally when the incoming frame size changes. requestLayout() can also be triggered
+ // externally. The layout change is a two pass process: first onMeasure() is called in a top-down
+ // traversal of the View tree, followed by an onLayout() pass that is also top-down. During the
+ // onLayout() pass, each parent is responsible for positioning its children using the sizes
+ // computed in the measure pass.
+ // |desiredLayoutsize| is the layout size we have requested in onMeasure() and are waiting for to
+ // take effect.
+ private Point desiredLayoutSize = new Point();
+ // |layoutSize|/|surfaceSize| is the actual current layout/surface size. They are updated in
+ // onLayout() and surfaceChanged() respectively.
+ private final Point layoutSize = new Point();
// TODO(magjed): Enable hardware scaler with SurfaceHolder.setFixedSize(). This will decouple
// layout and surface size.
- private int surfaceWidth;
- private int surfaceHeight;
+ private final Point surfaceSize = new Point();
// |isSurfaceCreated| keeps track of the current status in surfaceCreated()/surfaceDestroyed().
private boolean isSurfaceCreated;
// Last rendered frame dimensions, or 0 if no frame has been rendered yet.
@@ -121,12 +120,18 @@
// Time in ns spent in renderFrameOnRenderThread() function.
private long renderTimeNs;
- // Runnable for posting frames to render thread..
+ // Runnable for posting frames to render thread.
private final Runnable renderFrameRunnable = new Runnable() {
@Override public void run() {
renderFrameOnRenderThread();
}
};
+ // Runnable for clearing Surface to black.
+ private final Runnable makeBlackRunnable = new Runnable() {
+ @Override public void run() {
+ makeBlack();
+ }
+ };
/**
* Standard View constructor. In order to render something, you must first call init().
@@ -149,17 +154,28 @@
* reinitialize the renderer after a previous init()/release() cycle.
*/
public void init(
- EGLContext sharedContext, RendererCommon.RendererEvents rendererEvents) {
+ EglBase.Context sharedContext, RendererCommon.RendererEvents rendererEvents) {
+ init(sharedContext, rendererEvents, EglBase.CONFIG_PLAIN, new GlRectDrawer());
+ }
+
+ /**
+ * Initialize this class, sharing resources with |sharedContext|. The custom |drawer| will be used
+ * for drawing frames on the EGLSurface. This class is responsible for calling release() on
+ * |drawer|. It is allowed to call init() to reinitialize the renderer after a previous
+ * init()/release() cycle.
+ */
+ public void init(EglBase.Context sharedContext, RendererCommon.RendererEvents rendererEvents,
+ int[] configAttributes, RendererCommon.GlDrawer drawer) {
synchronized (handlerLock) {
if (renderThreadHandler != null) {
- throw new IllegalStateException("Already initialized");
+ throw new IllegalStateException(getResourceName() + "Already initialized");
}
- Logging.d(TAG, "Initializing");
+ Logging.d(TAG, getResourceName() + "Initializing.");
this.rendererEvents = rendererEvents;
+ this.drawer = drawer;
renderThread = new HandlerThread(TAG);
renderThread.start();
- drawer = new GlRectDrawer();
- eglBase = new EglBase(sharedContext, EglBase.ConfigType.PLAIN);
+ eglBase = EglBase.create(sharedContext, configAttributes);
renderThreadHandler = new Handler(renderThread.getLooper());
}
tryCreateEglSurface();
@@ -174,8 +190,8 @@
runOnRenderThread(new Runnable() {
@Override public void run() {
synchronized (layoutLock) {
- if (isSurfaceCreated) {
- eglBase.createSurface(getHolder());
+ if (isSurfaceCreated && !eglBase.hasSurface()) {
+ eglBase.createSurface(getHolder().getSurface());
eglBase.makeCurrent();
// Necessary for YUV frames with odd width.
GLES20.glPixelStorei(GLES20.GL_UNPACK_ALIGNMENT, 1);
@@ -195,7 +211,7 @@
final CountDownLatch eglCleanupBarrier = new CountDownLatch(1);
synchronized (handlerLock) {
if (renderThreadHandler == null) {
- Logging.d(TAG, "Already released");
+ Logging.d(TAG, getResourceName() + "Already released");
return;
}
// Release EGL and GL resources on render thread.
@@ -210,11 +226,8 @@
GLES20.glDeleteTextures(3, yuvTextures, 0);
yuvTextures = null;
}
- if (eglBase.hasSurface()) {
- // Clear last rendered image to black.
- GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
- eglBase.swapBuffers();
- }
+ // Clear last rendered image to black.
+ makeBlack();
eglBase.release();
eglBase = null;
eglCleanupBarrier.countDown();
@@ -242,6 +255,14 @@
frameRotation = 0;
rendererEvents = null;
}
+ resetStatistics();
+ }
+
+ /**
+ * Reset statistics. This will reset the logged statistics in logStatistics(), and
+ * RendererEvents.onFirstFrameRendered() will be called for the next frame.
+ */
+ public void resetStatistics() {
synchronized (statisticsLock) {
framesReceived = 0;
framesDropped = 0;
@@ -277,27 +298,28 @@
}
synchronized (handlerLock) {
if (renderThreadHandler == null) {
- Logging.d(TAG, "Dropping frame - SurfaceViewRenderer not initialized or already released.");
- } else {
- synchronized (frameLock) {
- if (pendingFrame == null) {
- updateFrameDimensionsAndReportEvents(frame);
- pendingFrame = frame;
- renderThreadHandler.post(renderFrameRunnable);
- return;
+ Logging.d(TAG, getResourceName()
+ + "Dropping frame - Not initialized or already released.");
+ VideoRenderer.renderFrameDone(frame);
+ return;
+ }
+ synchronized (frameLock) {
+ if (pendingFrame != null) {
+ // Drop old frame.
+ synchronized (statisticsLock) {
+ ++framesDropped;
}
+ VideoRenderer.renderFrameDone(pendingFrame);
}
+ pendingFrame = frame;
+ updateFrameDimensionsAndReportEvents(frame);
+ renderThreadHandler.post(renderFrameRunnable);
}
}
- // Drop frame.
- synchronized (statisticsLock) {
- ++framesDropped;
- }
- VideoRenderer.renderFrameDone(frame);
}
// Returns desired layout size given current measure specification and video aspect ratio.
- private Point getDesiredLayoutSize() {
+ private Point getDesiredLayoutSize(int widthSpec, int heightSpec) {
synchronized (layoutLock) {
final int maxWidth = getDefaultSize(Integer.MAX_VALUE, widthSpec);
final int maxHeight = getDefaultSize(Integer.MAX_VALUE, heightSpec);
@@ -317,18 +339,30 @@
@Override
protected void onMeasure(int widthSpec, int heightSpec) {
synchronized (layoutLock) {
- this.widthSpec = widthSpec;
- this.heightSpec = heightSpec;
- final Point size = getDesiredLayoutSize();
- setMeasuredDimension(size.x, size.y);
+ if (frameWidth == 0 || frameHeight == 0) {
+ super.onMeasure(widthSpec, heightSpec);
+ return;
+ }
+ desiredLayoutSize = getDesiredLayoutSize(widthSpec, heightSpec);
+ if (desiredLayoutSize.x != getMeasuredWidth() || desiredLayoutSize.y != getMeasuredHeight()) {
+ // Clear the surface asap before the layout change to avoid stretched video and other
+ // render artifacs. Don't wait for it to finish because the IO thread should never be
+ // blocked, so it's a best-effort attempt.
+ synchronized (handlerLock) {
+ if (renderThreadHandler != null) {
+ renderThreadHandler.postAtFrontOfQueue(makeBlackRunnable);
+ }
+ }
+ }
+ setMeasuredDimension(desiredLayoutSize.x, desiredLayoutSize.y);
}
}
@Override
protected void onLayout(boolean changed, int left, int top, int right, int bottom) {
synchronized (layoutLock) {
- layoutWidth = right - left;
- layoutHeight = bottom - top;
+ layoutSize.x = right - left;
+ layoutSize.y = bottom - top;
}
// Might have a pending frame waiting for a layout of correct size.
runOnRenderThread(renderFrameRunnable);
@@ -337,7 +371,7 @@
// SurfaceHolder.Callback interface.
@Override
public void surfaceCreated(final SurfaceHolder holder) {
- Logging.d(TAG, "Surface created");
+ Logging.d(TAG, getResourceName() + "Surface created.");
synchronized (layoutLock) {
isSurfaceCreated = true;
}
@@ -346,11 +380,11 @@
@Override
public void surfaceDestroyed(SurfaceHolder holder) {
- Logging.d(TAG, "Surface destroyed");
+ Logging.d(TAG, getResourceName() + "Surface destroyed.");
synchronized (layoutLock) {
isSurfaceCreated = false;
- surfaceWidth = 0;
- surfaceHeight = 0;
+ surfaceSize.x = 0;
+ surfaceSize.y = 0;
}
runOnRenderThread(new Runnable() {
@Override public void run() {
@@ -361,10 +395,10 @@
@Override
public void surfaceChanged(SurfaceHolder holder, int format, int width, int height) {
- Logging.d(TAG, "Surface changed: " + width + "x" + height);
+ Logging.d(TAG, getResourceName() + "Surface changed: " + width + "x" + height);
synchronized (layoutLock) {
- surfaceWidth = width;
- surfaceHeight = height;
+ surfaceSize.x = width;
+ surfaceSize.y = height;
}
// Might have a pending frame waiting for a surface of correct size.
runOnRenderThread(renderFrameRunnable);
@@ -381,26 +415,35 @@
}
}
+ private String getResourceName() {
+ try {
+ return getResources().getResourceEntryName(getId()) + ": ";
+ } catch (NotFoundException e) {
+ return "";
+ }
+ }
+
+ private void makeBlack() {
+ if (Thread.currentThread() != renderThread) {
+ throw new IllegalStateException(getResourceName() + "Wrong thread.");
+ }
+ if (eglBase != null && eglBase.hasSurface()) {
+ GLES20.glClearColor(0, 0, 0, 0);
+ GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
+ eglBase.swapBuffers();
+ }
+ }
+
/**
* Requests new layout if necessary. Returns true if layout and surface size are consistent.
*/
private boolean checkConsistentLayout() {
+ if (Thread.currentThread() != renderThread) {
+ throw new IllegalStateException(getResourceName() + "Wrong thread.");
+ }
synchronized (layoutLock) {
- final Point desiredLayoutSize = getDesiredLayoutSize();
- if (desiredLayoutSize.x != layoutWidth || desiredLayoutSize.y != layoutHeight) {
- Logging.d(TAG, "Requesting new layout with size: "
- + desiredLayoutSize.x + "x" + desiredLayoutSize.y);
- // Request layout update on UI thread.
- post(new Runnable() {
- @Override public void run() {
- requestLayout();
- }
- });
- return false;
- }
- // Wait for requestLayout() to propagate through this sequence before returning true:
- // requestLayout() -> onMeasure() -> onLayout() -> surfaceChanged().
- return surfaceWidth == layoutWidth && surfaceHeight == layoutHeight;
+ // Return false while we are in the middle of a layout change.
+ return layoutSize.equals(desiredLayoutSize) && surfaceSize.equals(layoutSize);
}
}
@@ -408,24 +451,8 @@
* Renders and releases |pendingFrame|.
*/
private void renderFrameOnRenderThread() {
- if (eglBase == null || !eglBase.hasSurface()) {
- Logging.d(TAG, "No surface to draw on");
- return;
- }
- if (!checkConsistentLayout()) {
- // Output intermediate black frames while the layout is updated.
- GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
- eglBase.swapBuffers();
- return;
- }
- // After a surface size change, the EGLSurface might still have a buffer of the old size in the
- // pipeline. Querying the EGLSurface will show if the underlying buffer dimensions haven't yet
- // changed. Such a buffer will be rendered incorrectly, so flush it with a black frame.
- synchronized (layoutLock) {
- if (eglBase.surfaceWidth() != surfaceWidth || eglBase.surfaceHeight() != surfaceHeight) {
- GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
- eglBase.swapBuffers();
- }
+ if (Thread.currentThread() != renderThread) {
+ throw new IllegalStateException(getResourceName() + "Wrong thread.");
}
// Fetch and render |pendingFrame|.
final VideoRenderer.I420Frame frame;
@@ -436,33 +463,39 @@
frame = pendingFrame;
pendingFrame = null;
}
-
- final long startTimeNs = System.nanoTime();
- final float[] samplingMatrix;
- if (frame.yuvFrame) {
- // The convention in WebRTC is that the first element in a ByteBuffer corresponds to the
- // top-left corner of the image, but in glTexImage2D() the first element corresponds to the
- // bottom-left corner. We correct this discrepancy by setting a vertical flip as sampling
- // matrix.
- samplingMatrix = RendererCommon.verticalFlipMatrix();
- } else {
- // TODO(magjed): Move updateTexImage() to the video source instead.
- SurfaceTexture surfaceTexture = (SurfaceTexture) frame.textureObject;
- surfaceTexture.updateTexImage();
- samplingMatrix = new float[16];
- surfaceTexture.getTransformMatrix(samplingMatrix);
+ if (eglBase == null || !eglBase.hasSurface()) {
+ Logging.d(TAG, getResourceName() + "No surface to draw on");
+ VideoRenderer.renderFrameDone(frame);
+ return;
+ }
+ if (!checkConsistentLayout()) {
+ // Output intermediate black frames while the layout is updated.
+ makeBlack();
+ VideoRenderer.renderFrameDone(frame);
+ return;
+ }
+ // After a surface size change, the EGLSurface might still have a buffer of the old size in the
+ // pipeline. Querying the EGLSurface will show if the underlying buffer dimensions haven't yet
+ // changed. Such a buffer will be rendered incorrectly, so flush it with a black frame.
+ synchronized (layoutLock) {
+ if (eglBase.surfaceWidth() != surfaceSize.x || eglBase.surfaceHeight() != surfaceSize.y) {
+ makeBlack();
+ }
}
+ final long startTimeNs = System.nanoTime();
final float[] texMatrix;
synchronized (layoutLock) {
final float[] rotatedSamplingMatrix =
- RendererCommon.rotateTextureMatrix(samplingMatrix, frame.rotationDegree);
+ RendererCommon.rotateTextureMatrix(frame.samplingMatrix, frame.rotationDegree);
final float[] layoutMatrix = RendererCommon.getLayoutMatrix(
- mirror, frameAspectRatio(), (float) layoutWidth / layoutHeight);
+ mirror, frameAspectRatio(), (float) layoutSize.x / layoutSize.y);
texMatrix = RendererCommon.multiplyMatrices(rotatedSamplingMatrix, layoutMatrix);
}
- GLES20.glViewport(0, 0, surfaceWidth, surfaceHeight);
+ // TODO(magjed): glClear() shouldn't be necessary since every pixel is covered anyway, but it's
+ // a workaround for bug 5147. Performance will be slightly worse.
+ GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
if (frame.yuvFrame) {
// Make sure YUV textures are allocated.
if (yuvTextures == null) {
@@ -471,11 +504,11 @@
yuvTextures[i] = GlUtil.generateTexture(GLES20.GL_TEXTURE_2D);
}
}
- drawer.uploadYuvData(
+ yuvUploader.uploadYuvData(
yuvTextures, frame.width, frame.height, frame.yuvStrides, frame.yuvPlanes);
- drawer.drawYuv(yuvTextures, texMatrix);
+ drawer.drawYuv(yuvTextures, texMatrix, 0, 0, surfaceSize.x, surfaceSize.y);
} else {
- drawer.drawOes(frame.textureId, texMatrix);
+ drawer.drawOes(frame.textureId, texMatrix, 0, 0, surfaceSize.x, surfaceSize.y);
}
eglBase.swapBuffers();
@@ -483,6 +516,12 @@
synchronized (statisticsLock) {
if (framesRendered == 0) {
firstFrameTimeNs = startTimeNs;
+ synchronized (layoutLock) {
+ Logging.d(TAG, getResourceName() + "Reporting first rendered frame.");
+ if (rendererEvents != null) {
+ rendererEvents.onFirstFrameRendered();
+ }
+ }
}
++framesRendered;
renderTimeNs += (System.nanoTime() - startTimeNs);
@@ -508,32 +547,32 @@
synchronized (layoutLock) {
if (frameWidth != frame.width || frameHeight != frame.height
|| frameRotation != frame.rotationDegree) {
+ Logging.d(TAG, getResourceName() + "Reporting frame resolution changed to "
+ + frame.width + "x" + frame.height + " with rotation " + frame.rotationDegree);
if (rendererEvents != null) {
- final String id = getResources().getResourceEntryName(getId());
- if (frameWidth == 0 || frameHeight == 0) {
- Logging.d(TAG, "ID: " + id + ". Reporting first rendered frame.");
- rendererEvents.onFirstFrameRendered();
- }
- Logging.d(TAG, "ID: " + id + ". Reporting frame resolution changed to "
- + frame.width + "x" + frame.height + " with rotation " + frame.rotationDegree);
rendererEvents.onFrameResolutionChanged(frame.width, frame.height, frame.rotationDegree);
}
frameWidth = frame.width;
frameHeight = frame.height;
frameRotation = frame.rotationDegree;
+ post(new Runnable() {
+ @Override public void run() {
+ requestLayout();
+ }
+ });
}
}
}
private void logStatistics() {
synchronized (statisticsLock) {
- Logging.d(TAG, "ID: " + getResources().getResourceEntryName(getId()) + ". Frames received: "
+ Logging.d(TAG, getResourceName() + "Frames received: "
+ framesReceived + ". Dropped: " + framesDropped + ". Rendered: " + framesRendered);
if (framesReceived > 0 && framesRendered > 0) {
final long timeSinceFirstFrameNs = System.nanoTime() - firstFrameTimeNs;
- Logging.d(TAG, "Duration: " + (int) (timeSinceFirstFrameNs / 1e6) +
- " ms. FPS: " + (float) framesRendered * 1e9 / timeSinceFirstFrameNs);
- Logging.d(TAG, "Average render time: "
+ Logging.d(TAG, getResourceName() + "Duration: " + (int) (timeSinceFirstFrameNs / 1e6) +
+ " ms. FPS: " + framesRendered * 1e9 / timeSinceFirstFrameNs);
+ Logging.d(TAG, getResourceName() + "Average render time: "
+ (int) (renderTimeNs / (1000 * framesRendered)) + " us.");
}
}
diff --git a/talk/app/webrtc/java/android/org/webrtc/ThreadUtils.java b/talk/app/webrtc/java/android/org/webrtc/ThreadUtils.java
index 0d8968a..e60ead9 100644
--- a/talk/app/webrtc/java/android/org/webrtc/ThreadUtils.java
+++ b/talk/app/webrtc/java/android/org/webrtc/ThreadUtils.java
@@ -28,11 +28,13 @@
package org.webrtc;
import android.os.Handler;
+import android.os.SystemClock;
import java.util.concurrent.Callable;
import java.util.concurrent.CountDownLatch;
+import java.util.concurrent.TimeUnit;
-final class ThreadUtils {
+public class ThreadUtils {
/**
* Utility class to be used for checking that a method is called on the correct thread.
*/
@@ -86,6 +88,29 @@
}
}
+ public static boolean joinUninterruptibly(final Thread thread, long timeoutMs) {
+ final long startTimeMs = SystemClock.elapsedRealtime();
+ long timeRemainingMs = timeoutMs;
+ boolean wasInterrupted = false;
+ while (timeRemainingMs > 0) {
+ try {
+ thread.join(timeRemainingMs);
+ break;
+ } catch (InterruptedException e) {
+ // Someone is asking us to return early at our convenience. We can't cancel this operation,
+ // but we should preserve the information and pass it along.
+ wasInterrupted = true;
+ final long elapsedTimeMs = SystemClock.elapsedRealtime() - startTimeMs;
+ timeRemainingMs = timeoutMs - elapsedTimeMs;
+ }
+ }
+ // Pass interruption information along.
+ if (wasInterrupted) {
+ Thread.currentThread().interrupt();
+ }
+ return !thread.isAlive();
+ }
+
public static void joinUninterruptibly(final Thread thread) {
executeUninterruptibly(new BlockingOperation() {
@Override
@@ -104,6 +129,30 @@
});
}
+ public static boolean awaitUninterruptibly(CountDownLatch barrier, long timeoutMs) {
+ final long startTimeMs = SystemClock.elapsedRealtime();
+ long timeRemainingMs = timeoutMs;
+ boolean wasInterrupted = false;
+ boolean result = false;
+ do {
+ try {
+ result = barrier.await(timeRemainingMs, TimeUnit.MILLISECONDS);
+ break;
+ } catch (InterruptedException e) {
+ // Someone is asking us to return early at our convenience. We can't cancel this operation,
+ // but we should preserve the information and pass it along.
+ wasInterrupted = true;
+ final long elapsedTimeMs = SystemClock.elapsedRealtime() - startTimeMs;
+ timeRemainingMs = timeoutMs - elapsedTimeMs;
+ }
+ } while (timeRemainingMs > 0);
+ // Pass interruption information along.
+ if (wasInterrupted) {
+ Thread.currentThread().interrupt();
+ }
+ return result;
+ }
+
/**
* Post |callable| to |handler| and wait for the result.
*/
diff --git a/talk/app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java b/talk/app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
index 4caefc5..36f60ed 100644
--- a/talk/app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
+++ b/talk/app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java
@@ -28,9 +28,6 @@
package org.webrtc;
import android.content.Context;
-import android.graphics.SurfaceTexture;
-import android.hardware.Camera;
-import android.hardware.Camera.PreviewCallback;
import android.os.Handler;
import android.os.HandlerThread;
import android.os.SystemClock;
@@ -53,9 +50,6 @@
import java.util.concurrent.CountDownLatch;
import java.util.concurrent.TimeUnit;
-import javax.microedition.khronos.egl.EGLContext;
-import javax.microedition.khronos.egl.EGL10;
-
// Android specific implementation of VideoCapturer.
// An instance of this class can be created by an application using
// VideoCapturerAndroid.create();
@@ -68,21 +62,22 @@
// camera thread. The internal *OnCameraThread() methods must check |camera| for null to check if
// the camera has been stopped.
@SuppressWarnings("deprecation")
-public class VideoCapturerAndroid extends VideoCapturer implements PreviewCallback,
+public class VideoCapturerAndroid extends VideoCapturer implements
+ android.hardware.Camera.PreviewCallback,
SurfaceTextureHelper.OnTextureFrameAvailableListener {
private final static String TAG = "VideoCapturerAndroid";
private final static int CAMERA_OBSERVER_PERIOD_MS = 2000;
+ private final static int CAMERA_FREEZE_REPORT_TIMOUT_MS = 6000;
- private Camera camera; // Only non-null while capturing.
+ private android.hardware.Camera camera; // Only non-null while capturing.
private HandlerThread cameraThread;
private final Handler cameraThreadHandler;
private Context applicationContext;
// Synchronization lock for |id|.
private final Object cameraIdLock = new Object();
private int id;
- private Camera.CameraInfo info;
- private final FramePool videoBuffers;
- private final CameraStatistics cameraStatistics = new CameraStatistics();
+ private android.hardware.Camera.CameraInfo info;
+ private final CameraStatistics cameraStatistics;
// Remember the requested format in case we want to switch cameras.
private int requestedWidth;
private int requestedHeight;
@@ -94,17 +89,28 @@
private CapturerObserver frameObserver = null;
private final CameraEventsHandler eventsHandler;
private boolean firstFrameReported;
+ // Arbitrary queue depth. Higher number means more memory allocated & held,
+ // lower number means more sensitivity to processing time in the client (and
+ // potentially stalling the capturer if it runs out of buffers to write to).
+ private static final int NUMBER_OF_CAPTURE_BUFFERS = 3;
+ private final Set<byte[]> queuedBuffers = new HashSet<byte[]>();
private final boolean isCapturingToTexture;
- private final SurfaceTextureHelper surfaceHelper;
+ final SurfaceTextureHelper surfaceHelper; // Package visible for testing purposes.
// The camera API can output one old frame after the camera has been switched or the resolution
// has been changed. This flag is used for dropping the first frame after camera restart.
private boolean dropNextFrame = false;
+ // |openCameraOnCodecThreadRunner| is used for retrying to open the camera if it is in use by
+ // another application when startCaptureOnCameraThread is called.
+ private Runnable openCameraOnCodecThreadRunner;
+ private final static int MAX_OPEN_CAMERA_ATTEMPTS = 3;
+ private final static int OPEN_CAMERA_DELAY_MS = 500;
+ private int openCameraAttempts;
// Camera error callback.
- private final Camera.ErrorCallback cameraErrorCallback =
- new Camera.ErrorCallback() {
+ private final android.hardware.Camera.ErrorCallback cameraErrorCallback =
+ new android.hardware.Camera.ErrorCallback() {
@Override
- public void onError(int error, Camera camera) {
+ public void onError(int error, android.hardware.Camera camera) {
String errorMessage;
if (error == android.hardware.Camera.CAMERA_ERROR_SERVER_DIED) {
errorMessage = "Camera server died!";
@@ -120,47 +126,45 @@
// Camera observer - monitors camera framerate. Observer is executed on camera thread.
private final Runnable cameraObserver = new Runnable() {
+ private int freezePeriodCount;
@Override
public void run() {
int cameraFramesCount = cameraStatistics.getAndResetFrameCount();
int cameraFps = (cameraFramesCount * 1000 + CAMERA_OBSERVER_PERIOD_MS / 2)
/ CAMERA_OBSERVER_PERIOD_MS;
- Logging.d(TAG, "Camera fps: " + cameraFps +
- ". Pending buffers: " + cameraStatistics.pendingFramesTimeStamps());
+ Logging.d(TAG, "Camera fps: " + cameraFps +".");
if (cameraFramesCount == 0) {
- Logging.e(TAG, "Camera freezed.");
- if (eventsHandler != null) {
- eventsHandler.onCameraError("Camera failure.");
+ ++freezePeriodCount;
+ if (CAMERA_OBSERVER_PERIOD_MS * freezePeriodCount > CAMERA_FREEZE_REPORT_TIMOUT_MS
+ && eventsHandler != null) {
+ Logging.e(TAG, "Camera freezed.");
+ if (surfaceHelper.isTextureInUse()) {
+ // This can only happen if we are capturing to textures.
+ eventsHandler.onCameraFreezed("Camera failure. Client must return video buffers.");
+ } else {
+ eventsHandler.onCameraFreezed("Camera failure.");
+ }
+ return;
}
} else {
- cameraThreadHandler.postDelayed(this, CAMERA_OBSERVER_PERIOD_MS);
+ freezePeriodCount = 0;
}
+ cameraThreadHandler.postDelayed(this, CAMERA_OBSERVER_PERIOD_MS);
}
};
private static class CameraStatistics {
private int frameCount = 0;
private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker();
- private final Set<Long> timeStampsNs = new HashSet<Long>();
CameraStatistics() {
threadChecker.detachThread();
}
- public void addPendingFrame(long timestamp) {
+ public void addFrame() {
threadChecker.checkIsOnValidThread();
++frameCount;
- timeStampsNs.add(timestamp);
- }
-
- public void frameReturned(long timestamp) {
- threadChecker.checkIsOnValidThread();
- if (!timeStampsNs.contains(timestamp)) {
- throw new IllegalStateException(
- "CameraStatistics.frameReturned called with unknown timestamp " + timestamp);
- }
- timeStampsNs.remove(timestamp);
}
public int getAndResetFrameCount() {
@@ -169,28 +173,16 @@
frameCount = 0;
return count;
}
-
- // Return number of pending frames that have not been returned.
- public int pendingFramesCount() {
- threadChecker.checkIsOnValidThread();
- return timeStampsNs.size();
- }
-
- public String pendingFramesTimeStamps() {
- threadChecker.checkIsOnValidThread();
- List<Long> timeStampsMs = new ArrayList<Long>();
- for (long ts : timeStampsNs) {
- timeStampsMs.add(TimeUnit.NANOSECONDS.toMillis(ts));
- }
- return timeStampsMs.toString();
- }
}
public static interface CameraEventsHandler {
- // Camera error handler - invoked when camera stops receiving frames
+ // Camera error handler - invoked when camera can not be opened
// or any camera exception happens on camera thread.
void onCameraError(String errorDescription);
+ // Invoked when camera stops receiving frames
+ void onCameraFreezed(String errorDescription);
+
// Callback invoked when camera is opening.
void onCameraOpening(int cameraId);
@@ -216,7 +208,7 @@
}
public static VideoCapturerAndroid create(String name,
- CameraEventsHandler eventsHandler, EGLContext sharedEglContext) {
+ CameraEventsHandler eventsHandler, EglBase.Context sharedEglContext) {
final int cameraId = lookupDeviceName(name);
if (cameraId == -1) {
return null;
@@ -224,7 +216,8 @@
final VideoCapturerAndroid capturer = new VideoCapturerAndroid(cameraId, eventsHandler,
sharedEglContext);
- capturer.setNativeCapturer(nativeCreateVideoCapturer(capturer));
+ capturer.setNativeCapturer(
+ nativeCreateVideoCapturer(capturer, capturer.surfaceHelper));
return capturer;
}
@@ -243,7 +236,7 @@
// Switch camera to the next valid camera id. This can only be called while
// the camera is running.
public void switchCamera(final CameraSwitchHandler handler) {
- if (Camera.getNumberOfCameras() < 2) {
+ if (android.hardware.Camera.getNumberOfCameras() < 2) {
if (handler != null) {
handler.onCameraSwitchError("No camera to switch to.");
}
@@ -274,7 +267,8 @@
pendingCameraSwitch = false;
}
if (handler != null) {
- handler.onCameraSwitchDone(info.facing == Camera.CameraInfo.CAMERA_FACING_FRONT);
+ handler.onCameraSwitchDone(
+ info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_FRONT);
}
}
});
@@ -282,6 +276,8 @@
// Requests a new output format from the video capturer. Captured frames
// by the camera will be scaled/or dropped by the video capturer.
+ // It does not matter if width and height are flipped. I.E, |width| = 640, |height| = 480 produce
+ // the same result as |width| = 480, |height| = 640.
// TODO(magjed/perkj): Document what this function does. Change name?
public void onOutputFormatRequest(final int width, final int height, final int framerate) {
cameraThreadHandler.post(new Runnable() {
@@ -303,7 +299,7 @@
// Helper function to retrieve the current camera id synchronously. Note that the camera id might
// change at any point by switchCamera() calls.
- private int getCurrentCameraId() {
+ int getCurrentCameraId() {
synchronized (cameraIdLock) {
return id;
}
@@ -329,20 +325,19 @@
}
private VideoCapturerAndroid(int cameraId, CameraEventsHandler eventsHandler,
- EGLContext sharedContext) {
- Logging.d(TAG, "VideoCapturerAndroid");
+ EglBase.Context sharedContext) {
this.id = cameraId;
this.eventsHandler = eventsHandler;
cameraThread = new HandlerThread(TAG);
cameraThread.start();
cameraThreadHandler = new Handler(cameraThread.getLooper());
- videoBuffers = new FramePool(cameraThread);
isCapturingToTexture = (sharedContext != null);
- surfaceHelper = SurfaceTextureHelper.create(
- isCapturingToTexture ? sharedContext : EGL10.EGL_NO_CONTEXT, cameraThreadHandler);
+ cameraStatistics = new CameraStatistics();
+ surfaceHelper = SurfaceTextureHelper.create(sharedContext, cameraThreadHandler);
if (isCapturingToTexture) {
surfaceHelper.setListener(this);
}
+ Logging.d(TAG, "VideoCapturerAndroid isCapturingToTexture : " + isCapturingToTexture);
}
private void checkIsOnCameraThread() {
@@ -355,13 +350,13 @@
// found. If |deviceName| is empty, the first available device is used.
private static int lookupDeviceName(String deviceName) {
Logging.d(TAG, "lookupDeviceName: " + deviceName);
- if (deviceName == null || Camera.getNumberOfCameras() == 0) {
+ if (deviceName == null || android.hardware.Camera.getNumberOfCameras() == 0) {
return -1;
}
if (deviceName.isEmpty()) {
return 0;
}
- for (int i = 0; i < Camera.getNumberOfCameras(); ++i) {
+ for (int i = 0; i < android.hardware.Camera.getNumberOfCameras(); ++i) {
if (deviceName.equals(CameraEnumerationAndroid.getDeviceName(i))) {
return i;
}
@@ -382,14 +377,9 @@
if (camera != null) {
throw new IllegalStateException("Release called while camera is running");
}
- if (cameraStatistics.pendingFramesCount() != 0) {
- throw new IllegalStateException("Release called with pending frames left");
- }
}
});
- surfaceHelper.disconnect();
- cameraThread.quit();
- ThreadUtils.joinUninterruptibly(cameraThread);
+ surfaceHelper.disconnect(cameraThreadHandler);
cameraThread = null;
}
@@ -413,6 +403,7 @@
if (frameObserver == null) {
throw new RuntimeException("frameObserver not set.");
}
+
cameraThreadHandler.post(new Runnable() {
@Override public void run() {
startCaptureOnCameraThread(width, height, framerate, frameObserver,
@@ -422,8 +413,8 @@
}
private void startCaptureOnCameraThread(
- int width, int height, int framerate, CapturerObserver frameObserver,
- Context applicationContext) {
+ final int width, final int height, final int framerate, final CapturerObserver frameObserver,
+ final Context applicationContext) {
Throwable error = null;
checkIsOnCameraThread();
if (camera != null) {
@@ -431,17 +422,36 @@
}
this.applicationContext = applicationContext;
this.frameObserver = frameObserver;
+ this.firstFrameReported = false;
+
try {
- synchronized (cameraIdLock) {
- Logging.d(TAG, "Opening camera " + id);
- firstFrameReported = false;
- if (eventsHandler != null) {
- eventsHandler.onCameraOpening(id);
+ try {
+ synchronized (cameraIdLock) {
+ Logging.d(TAG, "Opening camera " + id);
+ if (eventsHandler != null) {
+ eventsHandler.onCameraOpening(id);
+ }
+ camera = android.hardware.Camera.open(id);
+ info = new android.hardware.Camera.CameraInfo();
+ android.hardware.Camera.getCameraInfo(id, info);
}
- camera = Camera.open(id);
- info = new Camera.CameraInfo();
- Camera.getCameraInfo(id, info);
+ } catch (RuntimeException e) {
+ openCameraAttempts++;
+ if (openCameraAttempts < MAX_OPEN_CAMERA_ATTEMPTS) {
+ Logging.e(TAG, "Camera.open failed, retrying", e);
+ openCameraOnCodecThreadRunner = new Runnable() {
+ @Override public void run() {
+ startCaptureOnCameraThread(width, height, framerate, frameObserver,
+ applicationContext);
+ }
+ };
+ cameraThreadHandler.postDelayed(openCameraOnCodecThreadRunner, OPEN_CAMERA_DELAY_MS);
+ return;
+ }
+ openCameraAttempts = 0;
+ throw e;
}
+
try {
camera.setPreviewTexture(surfaceHelper.getSurfaceTexture());
} catch (IOException e) {
@@ -485,17 +495,18 @@
requestedFramerate = framerate;
// Find closest supported format for |width| x |height| @ |framerate|.
- final Camera.Parameters parameters = camera.getParameters();
+ final android.hardware.Camera.Parameters parameters = camera.getParameters();
final int[] range = CameraEnumerationAndroid.getFramerateRange(parameters, framerate * 1000);
- final Camera.Size previewSize = CameraEnumerationAndroid.getClosestSupportedSize(
- parameters.getSupportedPreviewSizes(), width, height);
+ final android.hardware.Camera.Size previewSize =
+ CameraEnumerationAndroid.getClosestSupportedSize(
+ parameters.getSupportedPreviewSizes(), width, height);
final CaptureFormat captureFormat = new CaptureFormat(
previewSize.width, previewSize.height,
- range[Camera.Parameters.PREVIEW_FPS_MIN_INDEX],
- range[Camera.Parameters.PREVIEW_FPS_MAX_INDEX]);
+ range[android.hardware.Camera.Parameters.PREVIEW_FPS_MIN_INDEX],
+ range[android.hardware.Camera.Parameters.PREVIEW_FPS_MAX_INDEX]);
// Check if we are already using this capture format, then we don't need to do anything.
- if (captureFormat.equals(this.captureFormat)) {
+ if (captureFormat.isSameFormat(this.captureFormat)) {
return;
}
@@ -511,11 +522,15 @@
parameters.setPreviewFpsRange(captureFormat.minFramerate, captureFormat.maxFramerate);
}
parameters.setPreviewSize(captureFormat.width, captureFormat.height);
- parameters.setPreviewFormat(captureFormat.imageFormat);
+
+ if (!isCapturingToTexture) {
+ parameters.setPreviewFormat(captureFormat.imageFormat);
+ }
// Picture size is for taking pictures and not for preview/video, but we need to set it anyway
// as a workaround for an aspect ratio problem on Nexus 7.
- final Camera.Size pictureSize = CameraEnumerationAndroid.getClosestSupportedSize(
- parameters.getSupportedPictureSizes(), width, height);
+ final android.hardware.Camera.Size pictureSize =
+ CameraEnumerationAndroid.getClosestSupportedSize(
+ parameters.getSupportedPictureSizes(), width, height);
parameters.setPictureSize(pictureSize.width, pictureSize.height);
// Temporarily stop preview if it's already running.
@@ -532,13 +547,19 @@
this.captureFormat = captureFormat;
List<String> focusModes = parameters.getSupportedFocusModes();
- if (focusModes.contains(Camera.Parameters.FOCUS_MODE_CONTINUOUS_VIDEO)) {
- parameters.setFocusMode(Camera.Parameters.FOCUS_MODE_CONTINUOUS_VIDEO);
+ if (focusModes.contains(android.hardware.Camera.Parameters.FOCUS_MODE_CONTINUOUS_VIDEO)) {
+ parameters.setFocusMode(android.hardware.Camera.Parameters.FOCUS_MODE_CONTINUOUS_VIDEO);
}
camera.setParameters(parameters);
if (!isCapturingToTexture) {
- videoBuffers.queueCameraBuffers(captureFormat.frameSize(), camera);
+ queuedBuffers.clear();
+ final int frameSize = captureFormat.frameSize();
+ for (int i = 0; i < NUMBER_OF_CAPTURE_BUFFERS; ++i) {
+ final ByteBuffer buffer = ByteBuffer.allocateDirect(frameSize);
+ queuedBuffers.add(buffer.array());
+ camera.addCallbackBuffer(buffer.array());
+ }
camera.setPreviewCallbackWithBuffer(this);
}
camera.startPreview();
@@ -561,6 +582,10 @@
private void stopCaptureOnCameraThread() {
checkIsOnCameraThread();
Logging.d(TAG, "stopCaptureOnCameraThread");
+ if (openCameraOnCodecThreadRunner != null) {
+ cameraThreadHandler.removeCallbacks(openCameraOnCodecThreadRunner);
+ }
+ openCameraAttempts = 0;
if (camera == null) {
Logging.e(TAG, "Calling stopCapture() for already stopped camera.");
return;
@@ -571,13 +596,7 @@
Logging.d(TAG, "Stop preview.");
camera.stopPreview();
camera.setPreviewCallbackWithBuffer(null);
- if (!isCapturingToTexture()) {
- videoBuffers.stopReturnBuffersToCamera();
- Logging.d(TAG, "stopReturnBuffersToCamera called."
- + (cameraStatistics.pendingFramesCount() == 0?
- " All buffers have been returned."
- : " Pending buffers: " + cameraStatistics.pendingFramesTimeStamps() + "."));
- }
+ queuedBuffers.clear();
captureFormat = null;
Logging.d(TAG, "Release camera.");
@@ -593,7 +612,7 @@
Logging.d(TAG, "switchCameraOnCameraThread");
stopCaptureOnCameraThread();
synchronized (cameraIdLock) {
- id = (id + 1) % Camera.getNumberOfCameras();
+ id = (id + 1) % android.hardware.Camera.getNumberOfCameras();
}
dropNextFrame = true;
startCaptureOnCameraThread(requestedWidth, requestedHeight, requestedFramerate, frameObserver,
@@ -612,17 +631,9 @@
frameObserver.onOutputFormatRequest(width, height, framerate);
}
- public void returnBuffer(final long timeStamp) {
- cameraThreadHandler.post(new Runnable() {
- @Override public void run() {
- cameraStatistics.frameReturned(timeStamp);
- if (isCapturingToTexture) {
- surfaceHelper.returnTextureFrame();
- } else {
- videoBuffers.returnBuffer(timeStamp);
- }
- }
- });
+ // Exposed for testing purposes only.
+ Handler getCameraThreadHandler() {
+ return cameraThreadHandler;
}
private int getDeviceOrientation() {
@@ -650,7 +661,7 @@
private int getFrameOrientation() {
int rotation = getDeviceOrientation();
- if (info.facing == Camera.CameraInfo.CAMERA_FACING_BACK) {
+ if (info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_BACK) {
rotation = 360 - rotation;
}
return (info.orientation + rotation) % 360;
@@ -658,9 +669,10 @@
// Called on cameraThread so must not "synchronized".
@Override
- public void onPreviewFrame(byte[] data, Camera callbackCamera) {
+ public void onPreviewFrame(byte[] data, android.hardware.Camera callbackCamera) {
checkIsOnCameraThread();
- if (camera == null) {
+ if (camera == null || !queuedBuffers.contains(data)) {
+ // The camera has been stopped or |data| is an old invalid buffer.
return;
}
if (camera != callbackCamera) {
@@ -675,16 +687,10 @@
firstFrameReported = true;
}
- // Mark the frame owning |data| as used.
- // Note that since data is directBuffer,
- // data.length >= videoBuffers.frameSize.
- if (videoBuffers.reserveByteBuffer(data, captureTimeNs)) {
- cameraStatistics.addPendingFrame(captureTimeNs);
- frameObserver.onByteBufferFrameCaptured(data, videoBuffers.frameSize, captureFormat.width,
- captureFormat.height, getFrameOrientation(), captureTimeNs);
- } else {
- Logging.w(TAG, "reserveByteBuffer failed - dropping frame.");
- }
+ cameraStatistics.addFrame();
+ frameObserver.onByteBufferFrameCaptured(data, captureFormat.width, captureFormat.height,
+ getFrameOrientation(), captureTimeNs);
+ camera.addCallbackBuffer(data);
}
@Override
@@ -696,135 +702,22 @@
surfaceHelper.returnTextureFrame();
return;
}
- if (!dropNextFrame) {
+ if (dropNextFrame) {
surfaceHelper.returnTextureFrame();
- dropNextFrame = true;
+ dropNextFrame = false;
return;
}
int rotation = getFrameOrientation();
- if (info.facing == Camera.CameraInfo.CAMERA_FACING_FRONT) {
+ if (info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_FRONT) {
// Undo the mirror that the OS "helps" us with.
// http://developer.android.com/reference/android/hardware/Camera.html#setDisplayOrientation(int)
transformMatrix =
RendererCommon.multiplyMatrices(transformMatrix, RendererCommon.horizontalFlipMatrix());
}
- transformMatrix = RendererCommon.rotateTextureMatrix(transformMatrix, rotation);
-
- final int rotatedWidth = (rotation % 180 == 0) ? captureFormat.width : captureFormat.height;
- final int rotatedHeight = (rotation % 180 == 0) ? captureFormat.height : captureFormat.width;
- cameraStatistics.addPendingFrame(timestampNs);
- frameObserver.onTextureFrameCaptured(rotatedWidth, rotatedHeight, oesTextureId,
- transformMatrix, timestampNs);
- }
-
- // Class used for allocating and bookkeeping video frames. All buffers are
- // direct allocated so that they can be directly used from native code. This class is
- // not thread-safe, and enforces single thread use.
- private static class FramePool {
- // Thread that all calls should be made on.
- private final Thread thread;
- // Arbitrary queue depth. Higher number means more memory allocated & held,
- // lower number means more sensitivity to processing time in the client (and
- // potentially stalling the capturer if it runs out of buffers to write to).
- private static final int numCaptureBuffers = 3;
- // This container tracks the buffers added as camera callback buffers. It is needed for finding
- // the corresponding ByteBuffer given a byte[].
- private final Map<byte[], ByteBuffer> queuedBuffers = new IdentityHashMap<byte[], ByteBuffer>();
- // This container tracks the frames that have been sent but not returned. It is needed for
- // keeping the buffers alive and for finding the corresponding ByteBuffer given a timestamp.
- private final Map<Long, ByteBuffer> pendingBuffers = new HashMap<Long, ByteBuffer>();
- private int frameSize = 0;
- private Camera camera;
-
- public FramePool(Thread thread) {
- this.thread = thread;
- }
-
- private void checkIsOnValidThread() {
- if (Thread.currentThread() != thread) {
- throw new IllegalStateException("Wrong thread");
- }
- }
-
- // Discards previous queued buffers and adds new callback buffers to camera.
- public void queueCameraBuffers(int frameSize, Camera camera) {
- checkIsOnValidThread();
- this.camera = camera;
- this.frameSize = frameSize;
-
- queuedBuffers.clear();
- for (int i = 0; i < numCaptureBuffers; ++i) {
- final ByteBuffer buffer = ByteBuffer.allocateDirect(frameSize);
- camera.addCallbackBuffer(buffer.array());
- queuedBuffers.put(buffer.array(), buffer);
- }
- Logging.d(TAG, "queueCameraBuffers enqueued " + numCaptureBuffers
- + " buffers of size " + frameSize + ".");
- }
-
- public void stopReturnBuffersToCamera() {
- checkIsOnValidThread();
- this.camera = null;
- queuedBuffers.clear();
- // Frames in |pendingBuffers| need to be kept alive until they are returned.
- }
-
- public boolean reserveByteBuffer(byte[] data, long timeStamp) {
- checkIsOnValidThread();
- final ByteBuffer buffer = queuedBuffers.remove(data);
- if (buffer == null) {
- // Frames might be posted to |onPreviewFrame| with the previous format while changing
- // capture format in |startPreviewOnCameraThread|. Drop these old frames.
- Logging.w(TAG, "Received callback buffer from previous configuration with length: "
- + (data == null ? "null" : data.length));
- return false;
- }
- if (buffer.capacity() != frameSize) {
- throw new IllegalStateException("Callback buffer has unexpected frame size");
- }
- if (pendingBuffers.containsKey(timeStamp)) {
- Logging.e(TAG, "Timestamp already present in pending buffers - they need to be unique");
- return false;
- }
- pendingBuffers.put(timeStamp, buffer);
- if (queuedBuffers.isEmpty()) {
- Logging.d(TAG, "Camera is running out of capture buffers.");
- }
- return true;
- }
-
- public void returnBuffer(long timeStamp) {
- checkIsOnValidThread();
- final ByteBuffer returnedFrame = pendingBuffers.remove(timeStamp);
- if (returnedFrame == null) {
- throw new RuntimeException("unknown data buffer with time stamp "
- + timeStamp + "returned?!?");
- }
-
- if (camera != null && returnedFrame.capacity() == frameSize) {
- camera.addCallbackBuffer(returnedFrame.array());
- if (queuedBuffers.isEmpty()) {
- Logging.d(TAG, "Frame returned when camera is running out of capture"
- + " buffers for TS " + TimeUnit.NANOSECONDS.toMillis(timeStamp));
- }
- queuedBuffers.put(returnedFrame.array(), returnedFrame);
- return;
- }
-
- if (returnedFrame.capacity() != frameSize) {
- Logging.d(TAG, "returnBuffer with time stamp "
- + TimeUnit.NANOSECONDS.toMillis(timeStamp)
- + " called with old frame size, " + returnedFrame.capacity() + ".");
- // Since this frame has the wrong size, don't requeue it. Frames with the correct size are
- // created in queueCameraBuffers so this must be an old buffer.
- return;
- }
-
- Logging.d(TAG, "returnBuffer with time stamp "
- + TimeUnit.NANOSECONDS.toMillis(timeStamp)
- + " called after camera has been stopped.");
- }
+ cameraStatistics.addFrame();
+ frameObserver.onTextureFrameCaptured(captureFormat.width, captureFormat.height, oesTextureId,
+ transformMatrix, rotation, timestampNs);
}
// Interface used for providing callbacks to an observer.
@@ -835,13 +728,14 @@
// Delivers a captured frame. Called on a Java thread owned by
// VideoCapturerAndroid.
- abstract void onByteBufferFrameCaptured(byte[] data, int length, int width, int height,
- int rotation, long timeStamp);
+ abstract void onByteBufferFrameCaptured(byte[] data, int width, int height, int rotation,
+ long timeStamp);
// Delivers a captured frame in a texture with id |oesTextureId|. Called on a Java thread
// owned by VideoCapturerAndroid.
abstract void onTextureFrameCaptured(
- int width, int height, int oesTextureId, float[] transformMatrix, long timestamp);
+ int width, int height, int oesTextureId, float[] transformMatrix, int rotation,
+ long timestamp);
// Requests an output format from the video capturer. Captured frames
// by the camera will be scaled/or dropped by the video capturer.
@@ -864,17 +758,18 @@
}
@Override
- public void onByteBufferFrameCaptured(byte[] data, int length, int width, int height,
+ public void onByteBufferFrameCaptured(byte[] data, int width, int height,
int rotation, long timeStamp) {
- nativeOnByteBufferFrameCaptured(nativeCapturer, data, length, width, height, rotation,
+ nativeOnByteBufferFrameCaptured(nativeCapturer, data, data.length, width, height, rotation,
timeStamp);
}
@Override
public void onTextureFrameCaptured(
- int width, int height, int oesTextureId, float[] transformMatrix, long timestamp) {
+ int width, int height, int oesTextureId, float[] transformMatrix, int rotation,
+ long timestamp) {
nativeOnTextureFrameCaptured(nativeCapturer, width, height, oesTextureId, transformMatrix,
- timestamp);
+ rotation, timestamp);
}
@Override
@@ -887,10 +782,12 @@
private native void nativeOnByteBufferFrameCaptured(long nativeCapturer,
byte[] data, int length, int width, int height, int rotation, long timeStamp);
private native void nativeOnTextureFrameCaptured(long nativeCapturer, int width, int height,
- int oesTextureId, float[] transformMatrix, long timestamp);
+ int oesTextureId, float[] transformMatrix, int rotation, long timestamp);
private native void nativeOnOutputFormatRequest(long nativeCapturer,
int width, int height, int framerate);
}
- private static native long nativeCreateVideoCapturer(VideoCapturerAndroid videoCapturer);
+ private static native long nativeCreateVideoCapturer(
+ VideoCapturerAndroid videoCapturer,
+ SurfaceTextureHelper surfaceHelper);
}
diff --git a/talk/app/webrtc/java/android/org/webrtc/VideoRendererGui.java b/talk/app/webrtc/java/android/org/webrtc/VideoRendererGui.java
index bacd0cf..bb6f01c 100644
--- a/talk/app/webrtc/java/android/org/webrtc/VideoRendererGui.java
+++ b/talk/app/webrtc/java/android/org/webrtc/VideoRendererGui.java
@@ -38,7 +38,7 @@
import android.annotation.SuppressLint;
import android.graphics.Point;
import android.graphics.Rect;
-import android.graphics.SurfaceTexture;
+import android.opengl.EGL14;
import android.opengl.GLES20;
import android.opengl.GLSurfaceView;
@@ -59,7 +59,7 @@
private static Runnable eglContextReady = null;
private static final String TAG = "VideoRendererGui";
private GLSurfaceView surface;
- private static EGLContext eglContext = null;
+ private static EglBase.Context eglContext = null;
// Indicates if SurfaceView.Renderer.onSurfaceCreated was called.
// If true then for every newly created yuv image renderer createTexture()
// should be called. The variable is accessed on multiple threads and
@@ -69,8 +69,6 @@
private int screenHeight;
// List of yuv renderers.
private final ArrayList<YuvImageRenderer> yuvImageRenderers;
- // |drawer| is synchronized on |yuvImageRenderers|.
- private GlRectDrawer drawer;
// Render and draw threads.
private static Thread renderFrameThread;
private static Thread drawThread;
@@ -99,6 +97,8 @@
// currently leaking resources to avoid a rare crash in release() where the EGLContext has
// become invalid beforehand.
private int[] yuvTextures = { 0, 0, 0 };
+ private final RendererCommon.YuvUploader yuvUploader = new RendererCommon.YuvUploader();
+ private final RendererCommon.GlDrawer drawer;
// Resources for making a deep copy of incoming OES texture frame.
private GlTextureFrameBuffer textureCopy;
@@ -157,12 +157,13 @@
private YuvImageRenderer(
GLSurfaceView surface, int id,
int x, int y, int width, int height,
- RendererCommon.ScalingType scalingType, boolean mirror) {
+ RendererCommon.ScalingType scalingType, boolean mirror, RendererCommon.GlDrawer drawer) {
Logging.d(TAG, "YuvImageRenderer.Create id: " + id);
this.surface = surface;
this.id = id;
this.scalingType = scalingType;
this.mirror = mirror;
+ this.drawer = drawer;
layoutInPercentage = new Rect(x, y, Math.min(100, x + width), Math.min(100, y + height));
updateLayoutProperties = false;
rotationDegree = 0;
@@ -174,6 +175,7 @@
private synchronized void release() {
surface = null;
+ drawer.release();
synchronized (pendingFrameLock) {
if (pendingFrame != null) {
VideoRenderer.renderFrameDone(pendingFrame);
@@ -226,7 +228,7 @@
}
}
- private void draw(GlRectDrawer drawer) {
+ private void draw() {
if (!seenFrame) {
// No frame received yet - nothing to render.
return;
@@ -241,29 +243,15 @@
}
if (isNewFrame) {
+ rotatedSamplingMatrix = RendererCommon.rotateTextureMatrix(
+ pendingFrame.samplingMatrix, pendingFrame.rotationDegree);
if (pendingFrame.yuvFrame) {
rendererType = RendererType.RENDERER_YUV;
- drawer.uploadYuvData(yuvTextures, pendingFrame.width, pendingFrame.height,
+ yuvUploader.uploadYuvData(yuvTextures, pendingFrame.width, pendingFrame.height,
pendingFrame.yuvStrides, pendingFrame.yuvPlanes);
- // The convention in WebRTC is that the first element in a ByteBuffer corresponds to the
- // top-left corner of the image, but in glTexImage2D() the first element corresponds to
- // the bottom-left corner. We correct this discrepancy by setting a vertical flip as
- // sampling matrix.
- final float[] samplingMatrix = RendererCommon.verticalFlipMatrix();
- rotatedSamplingMatrix =
- RendererCommon.rotateTextureMatrix(samplingMatrix, pendingFrame.rotationDegree);
} else {
rendererType = RendererType.RENDERER_TEXTURE;
- // External texture rendering. Update texture image to latest and make a deep copy of
- // the external texture.
- // TODO(magjed): Move updateTexImage() to the video source instead.
- final SurfaceTexture surfaceTexture = (SurfaceTexture) pendingFrame.textureObject;
- surfaceTexture.updateTexImage();
- final float[] samplingMatrix = new float[16];
- surfaceTexture.getTransformMatrix(samplingMatrix);
- rotatedSamplingMatrix =
- RendererCommon.rotateTextureMatrix(samplingMatrix, pendingFrame.rotationDegree);
-
+ // External texture rendering. Make a deep copy of the external texture.
// Reallocate offscreen texture if necessary.
textureCopy.setSize(pendingFrame.rotatedWidth(), pendingFrame.rotatedHeight());
@@ -272,12 +260,13 @@
GlUtil.checkNoGLES2Error("glBindFramebuffer");
// Copy the OES texture content. This will also normalize the sampling matrix.
- GLES20.glViewport(0, 0, textureCopy.getWidth(), textureCopy.getHeight());
- drawer.drawOes(pendingFrame.textureId, rotatedSamplingMatrix);
+ drawer.drawOes(pendingFrame.textureId, rotatedSamplingMatrix,
+ 0, 0, textureCopy.getWidth(), textureCopy.getHeight());
rotatedSamplingMatrix = RendererCommon.identityMatrix();
// Restore normal framebuffer.
GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, 0);
+ GLES20.glFinish();
}
copyTimeNs += (System.nanoTime() - now);
VideoRenderer.renderFrameDone(pendingFrame);
@@ -285,17 +274,17 @@
}
}
- // OpenGL defaults to lower left origin - flip vertically.
- GLES20.glViewport(displayLayout.left, screenHeight - displayLayout.bottom,
- displayLayout.width(), displayLayout.height());
-
updateLayoutMatrix();
final float[] texMatrix =
RendererCommon.multiplyMatrices(rotatedSamplingMatrix, layoutMatrix);
+ // OpenGL defaults to lower left origin - flip viewport position vertically.
+ final int viewportY = screenHeight - displayLayout.bottom;
if (rendererType == RendererType.RENDERER_YUV) {
- drawer.drawYuv(yuvTextures, texMatrix);
+ drawer.drawYuv(yuvTextures, texMatrix,
+ displayLayout.left, viewportY, displayLayout.width(), displayLayout.height());
} else {
- drawer.drawRgb(textureCopy.getTextureId(), texMatrix);
+ drawer.drawRgb(textureCopy.getTextureId(), texMatrix,
+ displayLayout.left, viewportY, displayLayout.width(), displayLayout.height());
}
if (isNewFrame) {
@@ -314,7 +303,7 @@
". Dropped: " + framesDropped + ". Rendered: " + framesRendered);
if (framesReceived > 0 && framesRendered > 0) {
Logging.d(TAG, "Duration: " + (int)(timeSinceFirstFrameNs / 1e6) +
- " ms. FPS: " + (float)framesRendered * 1e9 / timeSinceFirstFrameNs);
+ " ms. FPS: " + framesRendered * 1e9 / timeSinceFirstFrameNs);
Logging.d(TAG, "Draw time: " +
(int) (drawTimeNs / (1000 * framesRendered)) + " us. Copy time: " +
(int) (copyTimeNs / (1000 * framesReceived)) + " us");
@@ -429,7 +418,7 @@
eglContextReady = eglContextReadyCallback;
}
- public static synchronized EGLContext getEGLContext() {
+ public static synchronized EglBase.Context getEglBaseContext() {
return eglContext;
}
@@ -477,6 +466,16 @@
*/
public static synchronized YuvImageRenderer create(int x, int y, int width, int height,
RendererCommon.ScalingType scalingType, boolean mirror) {
+ return create(x, y, width, height, scalingType, mirror, new GlRectDrawer());
+ }
+
+ /**
+ * Creates VideoRenderer.Callbacks with top left corner at (x, y) and resolution (width, height).
+ * All parameters are in percentage of screen resolution. The custom |drawer| will be used for
+ * drawing frames on the EGLSurface. This class is responsible for calling release() on |drawer|.
+ */
+ public static synchronized YuvImageRenderer create(int x, int y, int width, int height,
+ RendererCommon.ScalingType scalingType, boolean mirror, RendererCommon.GlDrawer drawer) {
// Check display region parameters.
if (x < 0 || x > 100 || y < 0 || y > 100 ||
width < 0 || width > 100 || height < 0 || height > 100 ||
@@ -490,7 +489,7 @@
}
final YuvImageRenderer yuvImageRenderer = new YuvImageRenderer(
instance.surface, instance.yuvImageRenderers.size(),
- x, y, width, height, scalingType, mirror);
+ x, y, width, height, scalingType, mirror, drawer);
synchronized (instance.yuvImageRenderers) {
if (instance.onSurfaceCreatedCalled) {
// onSurfaceCreated has already been called for VideoRendererGui -
@@ -498,6 +497,7 @@
// rendering list.
final CountDownLatch countDownLatch = new CountDownLatch(1);
instance.surface.queueEvent(new Runnable() {
+ @Override
public void run() {
yuvImageRenderer.createTextures();
yuvImageRenderer.setScreenSize(
@@ -608,13 +608,16 @@
Logging.d(TAG, "VideoRendererGui.onSurfaceCreated");
// Store render EGL context.
synchronized (VideoRendererGui.class) {
- eglContext = ((EGL10) EGLContext.getEGL()).eglGetCurrentContext();
+ if (EglBase14.isEGL14Supported()) {
+ eglContext = new EglBase14.Context(EGL14.eglGetCurrentContext());
+ } else {
+ eglContext = new EglBase10.Context(((EGL10) EGLContext.getEGL()).eglGetCurrentContext());
+ }
+
Logging.d(TAG, "VideoRendererGui EGL Context: " + eglContext);
}
synchronized (yuvImageRenderers) {
- // Create drawer for YUV/OES frames.
- drawer = new GlRectDrawer();
// Create textures for all images.
for (YuvImageRenderer yuvImageRenderer : yuvImageRenderers) {
yuvImageRenderer.createTextures();
@@ -655,7 +658,7 @@
GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
synchronized (yuvImageRenderers) {
for (YuvImageRenderer yuvImageRenderer : yuvImageRenderers) {
- yuvImageRenderer.draw(drawer);
+ yuvImageRenderer.draw();
}
}
}
diff --git a/talk/app/webrtc/java/jni/androidmediacodeccommon.h b/talk/app/webrtc/java/jni/androidmediacodeccommon.h
index 348a716..92ea135 100644
--- a/talk/app/webrtc/java/jni/androidmediacodeccommon.h
+++ b/talk/app/webrtc/java/jni/androidmediacodeccommon.h
@@ -72,6 +72,8 @@
enum { kMediaCodecStatisticsIntervalMs = 3000 };
// Maximum amount of pending frames for VP8 decoder.
enum { kMaxPendingFramesVp8 = 1 };
+// Maximum amount of pending frames for VP9 decoder.
+enum { kMaxPendingFramesVp9 = 1 };
// Maximum amount of pending frames for H.264 decoder.
enum { kMaxPendingFramesH264 = 30 };
// Maximum amount of decoded frames for which per-frame logging is enabled.
diff --git a/talk/app/webrtc/java/jni/androidmediadecoder_jni.cc b/talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
index b664f16..c3d287c 100644
--- a/talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
+++ b/talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
@@ -33,14 +33,15 @@
#include "talk/app/webrtc/java/jni/androidmediacodeccommon.h"
#include "talk/app/webrtc/java/jni/classreferenceholder.h"
#include "talk/app/webrtc/java/jni/native_handle_impl.h"
+#include "talk/app/webrtc/java/jni/surfacetexturehelper_jni.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
-#include "webrtc/common_video/interface/i420_buffer_pool.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
+#include "webrtc/common_video/include/i420_buffer_pool.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
#include "webrtc/system_wrappers/include/logcat_trace_context.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "third_party/libyuv/include/libyuv/convert.h"
@@ -62,6 +63,7 @@
using webrtc::VideoCodecType;
using webrtc::kVideoCodecH264;
using webrtc::kVideoCodecVP8;
+using webrtc::kVideoCodecVP9;
namespace webrtc_jni {
@@ -87,9 +89,14 @@
int32_t Release() override;
int32_t Reset() override;
+
+ bool PrefersLateDecoding() const override { return true; }
+
// rtc::MessageHandler implementation.
void OnMessage(rtc::Message* msg) override;
+ const char* ImplementationName() const override;
+
private:
// CHECK-fail if not running on |codec_thread_|.
void CheckOnCodecThread();
@@ -105,13 +112,17 @@
// Type of video codec.
VideoCodecType codecType_;
+ // Render EGL context - owned by factory, should not be allocated/destroyed
+ // by VideoDecoder.
+ jobject render_egl_context_;
+
bool key_frame_required_;
bool inited_;
bool sw_fallback_required_;
bool use_surface_;
VideoCodec codec_;
webrtc::I420BufferPool decoded_frame_pool_;
- NativeHandleImpl native_handle_;
+ rtc::scoped_refptr<SurfaceTextureHelper> surface_texture_helper_;
DecodedImageCallback* callback_;
int frames_received_; // Number of frames received by decoder.
int frames_decoded_; // Number of frames decoded by decoder.
@@ -120,10 +131,6 @@
int current_bytes_; // Encoded bytes in the current statistics interval.
int current_decoding_time_ms_; // Overall decoding time in the current second
uint32_t max_pending_frames_; // Maximum number of pending input frames
- std::vector<int32_t> timestamps_;
- std::vector<int64_t> ntp_times_ms_;
- std::vector<int64_t> frame_rtc_times_ms_; // Time when video frame is sent to
- // decoder input.
// State that is constant for the lifetime of this object once the ctor
// returns.
@@ -134,7 +141,8 @@
jmethodID j_release_method_;
jmethodID j_dequeue_input_buffer_method_;
jmethodID j_queue_input_buffer_method_;
- jmethodID j_dequeue_output_buffer_method_;
+ jmethodID j_dequeue_byte_buffer_method_;
+ jmethodID j_dequeue_texture_buffer_method_;
jmethodID j_return_decoded_byte_buffer_method_;
// MediaCodecVideoDecoder fields.
jfieldID j_input_buffers_field_;
@@ -144,24 +152,23 @@
jfieldID j_height_field_;
jfieldID j_stride_field_;
jfieldID j_slice_height_field_;
- jfieldID j_surface_texture_field_;
// MediaCodecVideoDecoder.DecodedTextureBuffer fields.
- jfieldID j_textureID_field_;
- jfieldID j_texture_presentation_timestamp_us_field_;
- // MediaCodecVideoDecoder.DecodedByteBuffer fields.
+ jfieldID j_texture_id_field_;
+ jfieldID j_transform_matrix_field_;
+ jfieldID j_texture_timestamp_ms_field_;
+ jfieldID j_texture_ntp_timestamp_ms_field_;
+ jfieldID j_texture_decode_time_ms_field_;
+ jfieldID j_texture_frame_delay_ms_field_;
+ // MediaCodecVideoDecoder.DecodedOutputBuffer fields.
jfieldID j_info_index_field_;
jfieldID j_info_offset_field_;
jfieldID j_info_size_field_;
- jfieldID j_info_presentation_timestamp_us_field_;
+ jfieldID j_info_timestamp_ms_field_;
+ jfieldID j_info_ntp_timestamp_ms_field_;
+ jfieldID j_byte_buffer_decode_time_ms_field_;
// Global references; must be deleted in Release().
std::vector<jobject> input_buffers_;
- jobject surface_texture_;
- jobject previous_surface_texture_;
-
- // Render EGL context - owned by factory, should not be allocated/destroyed
- // by VideoDecoder.
- jobject render_egl_context_;
};
MediaCodecVideoDecoder::MediaCodecVideoDecoder(
@@ -171,8 +178,6 @@
key_frame_required_(true),
inited_(false),
sw_fallback_required_(false),
- surface_texture_(NULL),
- previous_surface_texture_(NULL),
codec_thread_(new Thread()),
j_media_codec_video_decoder_class_(
jni,
@@ -191,19 +196,22 @@
j_init_decode_method_ = GetMethodID(
jni, *j_media_codec_video_decoder_class_, "initDecode",
"(Lorg/webrtc/MediaCodecVideoDecoder$VideoCodecType;"
- "IILjavax/microedition/khronos/egl/EGLContext;)Z");
+ "IILorg/webrtc/SurfaceTextureHelper;)Z");
j_release_method_ =
GetMethodID(jni, *j_media_codec_video_decoder_class_, "release", "()V");
j_dequeue_input_buffer_method_ = GetMethodID(
jni, *j_media_codec_video_decoder_class_, "dequeueInputBuffer", "()I");
j_queue_input_buffer_method_ = GetMethodID(
- jni, *j_media_codec_video_decoder_class_, "queueInputBuffer", "(IIJ)Z");
- j_dequeue_output_buffer_method_ = GetMethodID(
+ jni, *j_media_codec_video_decoder_class_, "queueInputBuffer", "(IIJJJ)Z");
+ j_dequeue_byte_buffer_method_ = GetMethodID(
jni, *j_media_codec_video_decoder_class_, "dequeueOutputBuffer",
- "(I)Ljava/lang/Object;");
+ "(I)Lorg/webrtc/MediaCodecVideoDecoder$DecodedOutputBuffer;");
+ j_dequeue_texture_buffer_method_ = GetMethodID(
+ jni, *j_media_codec_video_decoder_class_, "dequeueTextureBuffer",
+ "(I)Lorg/webrtc/MediaCodecVideoDecoder$DecodedTextureBuffer;");
j_return_decoded_byte_buffer_method_ =
GetMethodID(jni, *j_media_codec_video_decoder_class_,
- "returnDecodedByteBuffer", "(I)V");
+ "returnDecodedOutputBuffer", "(I)V");
j_input_buffers_field_ = GetFieldID(
jni, *j_media_codec_video_decoder_class_,
@@ -221,28 +229,36 @@
jni, *j_media_codec_video_decoder_class_, "stride", "I");
j_slice_height_field_ = GetFieldID(
jni, *j_media_codec_video_decoder_class_, "sliceHeight", "I");
- j_surface_texture_field_ = GetFieldID(
- jni, *j_media_codec_video_decoder_class_, "surfaceTexture",
- "Landroid/graphics/SurfaceTexture;");
- jclass j_decoder_decoded_texture_buffer_class = FindClass(jni,
+ jclass j_decoded_texture_buffer_class = FindClass(jni,
"org/webrtc/MediaCodecVideoDecoder$DecodedTextureBuffer");
- j_textureID_field_ = GetFieldID(
- jni, j_decoder_decoded_texture_buffer_class, "textureID", "I");
- j_texture_presentation_timestamp_us_field_ =
- GetFieldID(jni, j_decoder_decoded_texture_buffer_class,
- "presentationTimestampUs", "J");
+ j_texture_id_field_ = GetFieldID(
+ jni, j_decoded_texture_buffer_class, "textureID", "I");
+ j_transform_matrix_field_ = GetFieldID(
+ jni, j_decoded_texture_buffer_class, "transformMatrix", "[F");
+ j_texture_timestamp_ms_field_ = GetFieldID(
+ jni, j_decoded_texture_buffer_class, "timeStampMs", "J");
+ j_texture_ntp_timestamp_ms_field_ = GetFieldID(
+ jni, j_decoded_texture_buffer_class, "ntpTimeStampMs", "J");
+ j_texture_decode_time_ms_field_ = GetFieldID(
+ jni, j_decoded_texture_buffer_class, "decodeTimeMs", "J");
+ j_texture_frame_delay_ms_field_ = GetFieldID(
+ jni, j_decoded_texture_buffer_class, "frameDelayMs", "J");
- jclass j_decoder_decoded_byte_buffer_class = FindClass(jni,
- "org/webrtc/MediaCodecVideoDecoder$DecodedByteBuffer");
+ jclass j_decoded_output_buffer_class = FindClass(jni,
+ "org/webrtc/MediaCodecVideoDecoder$DecodedOutputBuffer");
j_info_index_field_ = GetFieldID(
- jni, j_decoder_decoded_byte_buffer_class, "index", "I");
+ jni, j_decoded_output_buffer_class, "index", "I");
j_info_offset_field_ = GetFieldID(
- jni, j_decoder_decoded_byte_buffer_class, "offset", "I");
+ jni, j_decoded_output_buffer_class, "offset", "I");
j_info_size_field_ = GetFieldID(
- jni, j_decoder_decoded_byte_buffer_class, "size", "I");
- j_info_presentation_timestamp_us_field_ = GetFieldID(
- jni, j_decoder_decoded_byte_buffer_class, "presentationTimestampUs", "J");
+ jni, j_decoded_output_buffer_class, "size", "I");
+ j_info_timestamp_ms_field_ = GetFieldID(
+ jni, j_decoded_output_buffer_class, "timeStampMs", "J");
+ j_info_ntp_timestamp_ms_field_ = GetFieldID(
+ jni, j_decoded_output_buffer_class, "ntpTimeStampMs", "J");
+ j_byte_buffer_decode_time_ms_field_ = GetFieldID(
+ jni, j_decoded_output_buffer_class, "decodeTimeMs", "J");
CHECK_EXCEPTION(jni) << "MediaCodecVideoDecoder ctor failed";
use_surface_ = (render_egl_context_ != NULL);
@@ -254,14 +270,6 @@
MediaCodecVideoDecoder::~MediaCodecVideoDecoder() {
// Call Release() to ensure no more callbacks to us after we are deleted.
Release();
- // Delete global references.
- JNIEnv* jni = AttachCurrentThreadIfNeeded();
- if (previous_surface_texture_ != NULL) {
- jni->DeleteGlobalRef(previous_surface_texture_);
- }
- if (surface_texture_ != NULL) {
- jni->DeleteGlobalRef(surface_texture_);
- }
}
int32_t MediaCodecVideoDecoder::InitDecode(const VideoCodec* inst,
@@ -312,6 +320,21 @@
frames_received_ = 0;
frames_decoded_ = 0;
+ jobject java_surface_texture_helper_ = nullptr;
+ if (use_surface_) {
+ java_surface_texture_helper_ = jni->CallStaticObjectMethod(
+ FindClass(jni, "org/webrtc/SurfaceTextureHelper"),
+ GetStaticMethodID(jni,
+ FindClass(jni, "org/webrtc/SurfaceTextureHelper"),
+ "create",
+ "(Lorg/webrtc/EglBase$Context;)"
+ "Lorg/webrtc/SurfaceTextureHelper;"),
+ render_egl_context_);
+ RTC_CHECK(java_surface_texture_helper_ != nullptr);
+ surface_texture_helper_ = new rtc::RefCountedObject<SurfaceTextureHelper>(
+ jni, java_surface_texture_helper_);
+ }
+
jobject j_video_codec_enum = JavaEnumFromIndex(
jni, "MediaCodecVideoDecoder$VideoCodecType", codecType_);
bool success = jni->CallBooleanMethod(
@@ -320,7 +343,7 @@
j_video_codec_enum,
codec_.width,
codec_.height,
- use_surface_ ? render_egl_context_ : nullptr);
+ java_surface_texture_helper_);
if (CheckException(jni) || !success) {
ALOGE << "Codec initialization error - fallback to SW codec.";
sw_fallback_required_ = true;
@@ -332,6 +355,9 @@
case kVideoCodecVP8:
max_pending_frames_ = kMaxPendingFramesVp8;
break;
+ case kVideoCodecVP9:
+ max_pending_frames_ = kMaxPendingFramesVp9;
+ break;
case kVideoCodecH264:
max_pending_frames_ = kMaxPendingFramesH264;
break;
@@ -342,9 +368,6 @@
current_frames_ = 0;
current_bytes_ = 0;
current_decoding_time_ms_ = 0;
- timestamps_.clear();
- ntp_times_ms_.clear();
- frame_rtc_times_ms_.clear();
jobjectArray input_buffers = (jobjectArray)GetObjectField(
jni, *j_media_codec_video_decoder_, j_input_buffers_field_);
@@ -361,15 +384,6 @@
}
}
- if (use_surface_) {
- jobject surface_texture = GetObjectField(
- jni, *j_media_codec_video_decoder_, j_surface_texture_field_);
- if (previous_surface_texture_ != NULL) {
- jni->DeleteGlobalRef(previous_surface_texture_);
- }
- previous_surface_texture_ = surface_texture_;
- surface_texture_ = jni->NewGlobalRef(surface_texture);
- }
codec_thread_->PostDelayed(kMediaCodecPollMs, this);
return WEBRTC_VIDEO_CODEC_OK;
@@ -395,6 +409,7 @@
}
input_buffers_.clear();
jni->CallVoidMethod(*j_media_codec_video_decoder_, j_release_method_);
+ surface_texture_helper_ = nullptr;
inited_ = false;
rtc::MessageQueueManager::Clear(this);
if (CheckException(jni)) {
@@ -501,19 +516,21 @@
// Try to drain the decoder and wait until output is not too
// much behind the input.
- if (frames_received_ > frames_decoded_ + max_pending_frames_) {
+ const int64 drain_start = GetCurrentTimeMs();
+ while ((frames_received_ > frames_decoded_ + max_pending_frames_) &&
+ (GetCurrentTimeMs() - drain_start) < kMediaCodecTimeoutMs) {
ALOGV("Received: %d. Decoded: %d. Wait for output...",
frames_received_, frames_decoded_);
- if (!DeliverPendingOutputs(jni, kMediaCodecTimeoutMs * 1000)) {
+ if (!DeliverPendingOutputs(jni, kMediaCodecPollMs)) {
ALOGE << "DeliverPendingOutputs error. Frames received: " <<
frames_received_ << ". Frames decoded: " << frames_decoded_;
return ProcessHWErrorOnCodecThread();
}
- if (frames_received_ > frames_decoded_ + max_pending_frames_) {
- ALOGE << "Output buffer dequeue timeout. Frames received: " <<
- frames_received_ << ". Frames decoded: " << frames_decoded_;
- return ProcessHWErrorOnCodecThread();
- }
+ }
+ if (frames_received_ > frames_decoded_ + max_pending_frames_) {
+ ALOGE << "Output buffer dequeue timeout. Frames received: " <<
+ frames_received_ << ". Frames decoded: " << frames_decoded_;
+ return ProcessHWErrorOnCodecThread();
}
// Get input buffer.
@@ -535,11 +552,14 @@
" is bigger than buffer size " << buffer_capacity;
return ProcessHWErrorOnCodecThread();
}
- jlong timestamp_us = (frames_received_ * 1000000) / codec_.maxFramerate;
+ jlong presentation_timestamp_us =
+ (frames_received_ * 1000000) / codec_.maxFramerate;
if (frames_decoded_ < kMaxDecodedLogFrames) {
ALOGD << "Decoder frame in # " << frames_received_ << ". Type: "
<< inputImage._frameType << ". Buffer # " <<
- j_input_buffer_index << ". TS: " << (int)(timestamp_us / 1000)
+ j_input_buffer_index << ". pTS: "
+ << (int)(presentation_timestamp_us / 1000)
+ << ". TS: " << inputImage._timeStamp
<< ". Size: " << inputImage._length;
}
memcpy(buffer, inputImage._buffer, inputImage._length);
@@ -547,16 +567,16 @@
// Save input image timestamps for later output.
frames_received_++;
current_bytes_ += inputImage._length;
- timestamps_.push_back(inputImage._timeStamp);
- ntp_times_ms_.push_back(inputImage.ntp_time_ms_);
- frame_rtc_times_ms_.push_back(GetCurrentTimeMs());
// Feed input to decoder.
- bool success = jni->CallBooleanMethod(*j_media_codec_video_decoder_,
- j_queue_input_buffer_method_,
- j_input_buffer_index,
- inputImage._length,
- timestamp_us);
+ bool success = jni->CallBooleanMethod(
+ *j_media_codec_video_decoder_,
+ j_queue_input_buffer_method_,
+ j_input_buffer_index,
+ inputImage._length,
+ presentation_timestamp_us,
+ static_cast<int64_t> (inputImage._timeStamp),
+ inputImage.ntp_time_ms_);
if (CheckException(jni) || !success) {
ALOGE << "queueInputBuffer error";
return ProcessHWErrorOnCodecThread();
@@ -572,16 +592,18 @@
}
bool MediaCodecVideoDecoder::DeliverPendingOutputs(
- JNIEnv* jni, int dequeue_timeout_us) {
+ JNIEnv* jni, int dequeue_timeout_ms) {
if (frames_received_ <= frames_decoded_) {
// No need to query for output buffers - decoder is drained.
return true;
}
// Get decoder output.
- jobject j_decoder_output_buffer = jni->CallObjectMethod(
- *j_media_codec_video_decoder_,
- j_dequeue_output_buffer_method_,
- dequeue_timeout_us);
+ jobject j_decoder_output_buffer =
+ jni->CallObjectMethod(*j_media_codec_video_decoder_,
+ use_surface_ ? j_dequeue_texture_buffer_method_
+ : j_dequeue_byte_buffer_method_,
+ dequeue_timeout_ms);
+
if (CheckException(jni)) {
ALOGE << "dequeueOutputBuffer() error";
return false;
@@ -601,19 +623,35 @@
j_slice_height_field_);
rtc::scoped_refptr<webrtc::VideoFrameBuffer> frame_buffer;
- long output_timestamps_ms = 0;
+ int64_t output_timestamps_ms = 0;
+ int64_t output_ntp_timestamps_ms = 0;
+ int decode_time_ms = 0;
+ int64_t frame_delayed_ms = 0;
if (use_surface_) {
// Extract data from Java DecodedTextureBuffer.
const int texture_id =
- GetIntField(jni, j_decoder_output_buffer, j_textureID_field_);
- const int64_t timestamp_us =
- GetLongField(jni, j_decoder_output_buffer,
- j_texture_presentation_timestamp_us_field_);
- output_timestamps_ms = timestamp_us / rtc::kNumMicrosecsPerMillisec;
- // Create webrtc::VideoFrameBuffer with native texture handle.
- native_handle_.SetTextureObject(surface_texture_, texture_id);
- frame_buffer = new rtc::RefCountedObject<JniNativeHandleBuffer>(
- &native_handle_, width, height);
+ GetIntField(jni, j_decoder_output_buffer, j_texture_id_field_);
+ if (texture_id != 0) { // |texture_id| == 0 represents a dropped frame.
+ const jfloatArray j_transform_matrix =
+ reinterpret_cast<jfloatArray>(GetObjectField(
+ jni, j_decoder_output_buffer, j_transform_matrix_field_));
+ const int64_t timestamp_us =
+ GetLongField(jni, j_decoder_output_buffer,
+ j_texture_timestamp_ms_field_);
+ output_timestamps_ms = GetLongField(jni, j_decoder_output_buffer,
+ j_texture_timestamp_ms_field_);
+ output_ntp_timestamps_ms =
+ GetLongField(jni, j_decoder_output_buffer,
+ j_texture_ntp_timestamp_ms_field_);
+ decode_time_ms = GetLongField(jni, j_decoder_output_buffer,
+ j_texture_decode_time_ms_field_);
+ frame_delayed_ms = GetLongField(jni, j_decoder_output_buffer,
+ j_texture_frame_delay_ms_field_);
+
+ // Create webrtc::VideoFrameBuffer with native texture handle.
+ frame_buffer = surface_texture_helper_->CreateTextureFrame(
+ width, height, NativeHandleImpl(jni, texture_id, j_transform_matrix));
+ }
} else {
// Extract data from Java ByteBuffer and create output yuv420 frame -
// for non surface decoding only.
@@ -623,9 +661,14 @@
GetIntField(jni, j_decoder_output_buffer, j_info_offset_field_);
const int output_buffer_size =
GetIntField(jni, j_decoder_output_buffer, j_info_size_field_);
- const int64_t timestamp_us = GetLongField(
- jni, j_decoder_output_buffer, j_info_presentation_timestamp_us_field_);
- output_timestamps_ms = timestamp_us / rtc::kNumMicrosecsPerMillisec;
+ output_timestamps_ms = GetLongField(jni, j_decoder_output_buffer,
+ j_info_timestamp_ms_field_);
+ output_ntp_timestamps_ms =
+ GetLongField(jni, j_decoder_output_buffer,
+ j_info_ntp_timestamp_ms_field_);
+
+ decode_time_ms = GetLongField(jni, j_decoder_output_buffer,
+ j_byte_buffer_decode_time_ms_field_);
if (output_buffer_size < width * height * 3 / 2) {
ALOGE << "Insufficient output buffer size: " << output_buffer_size;
@@ -683,41 +726,31 @@
j_return_decoded_byte_buffer_method_,
output_buffer_index);
if (CheckException(jni)) {
- ALOGE << "returnDecodedByteBuffer error";
+ ALOGE << "returnDecodedOutputBuffer error";
return false;
}
}
VideoFrame decoded_frame(frame_buffer, 0, 0, webrtc::kVideoRotation_0);
+ decoded_frame.set_timestamp(output_timestamps_ms);
+ decoded_frame.set_ntp_time_ms(output_ntp_timestamps_ms);
- // Get frame timestamps from a queue.
- if (timestamps_.size() > 0) {
- decoded_frame.set_timestamp(timestamps_.front());
- timestamps_.erase(timestamps_.begin());
- }
- if (ntp_times_ms_.size() > 0) {
- decoded_frame.set_ntp_time_ms(ntp_times_ms_.front());
- ntp_times_ms_.erase(ntp_times_ms_.begin());
- }
- int64_t frame_decoding_time_ms = 0;
- if (frame_rtc_times_ms_.size() > 0) {
- frame_decoding_time_ms = GetCurrentTimeMs() - frame_rtc_times_ms_.front();
- frame_rtc_times_ms_.erase(frame_rtc_times_ms_.begin());
- }
if (frames_decoded_ < kMaxDecodedLogFrames) {
ALOGD << "Decoder frame out # " << frames_decoded_ << ". " << width <<
" x " << height << ". " << stride << " x " << slice_height <<
- ". Color: " << color_format << ". TS:" << (int)output_timestamps_ms <<
- ". DecTime: " << (int)frame_decoding_time_ms;
+ ". Color: " << color_format << ". TS:" << decoded_frame.timestamp() <<
+ ". DecTime: " << (int)decode_time_ms <<
+ ". DelayTime: " << (int)frame_delayed_ms;
}
// Calculate and print decoding statistics - every 3 seconds.
frames_decoded_++;
current_frames_++;
- current_decoding_time_ms_ += frame_decoding_time_ms;
+ current_decoding_time_ms_ += decode_time_ms;
int statistic_time_ms = GetCurrentTimeMs() - start_time_ms_;
if (statistic_time_ms >= kMediaCodecStatisticsIntervalMs &&
current_frames_ > 0) {
- ALOGD << "Decoded frames: " << frames_decoded_ << ". Bitrate: " <<
+ ALOGD << "Decoded frames: " << frames_decoded_ << ". Received frames: "
+ << frames_received_ << ". Bitrate: " <<
(current_bytes_ * 8 / statistic_time_ms) << " kbps, fps: " <<
((current_frames_ * 1000 + statistic_time_ms / 2) / statistic_time_ms)
<< ". decTime: " << (current_decoding_time_ms_ / current_frames_) <<
@@ -728,12 +761,15 @@
current_decoding_time_ms_ = 0;
}
- // Callback - output decoded frame.
- const int32_t callback_status = callback_->Decoded(decoded_frame);
- if (callback_status > 0) {
- ALOGE << "callback error";
+ // |.IsZeroSize())| returns true when a frame has been dropped.
+ if (!decoded_frame.IsZeroSize()) {
+ // Callback - output decoded frame.
+ const int32_t callback_status =
+ callback_->Decoded(decoded_frame, decode_time_ms);
+ if (callback_status > 0) {
+ ALOGE << "callback error";
+ }
}
-
return true;
}
@@ -790,6 +826,17 @@
supported_codec_types_.push_back(kVideoCodecVP8);
}
+ bool is_vp9_hw_supported = jni->CallStaticBooleanMethod(
+ j_decoder_class,
+ GetStaticMethodID(jni, j_decoder_class, "isVp9HwSupported", "()Z"));
+ if (CheckException(jni)) {
+ is_vp9_hw_supported = false;
+ }
+ if (is_vp9_hw_supported) {
+ ALOGD << "VP9 HW Decoder supported.";
+ supported_codec_types_.push_back(kVideoCodecVP9);
+ }
+
bool is_h264_hw_supported = jni->CallStaticBooleanMethod(
j_decoder_class,
GetStaticMethodID(jni, j_decoder_class, "isH264HwSupported", "()Z"));
@@ -825,7 +872,7 @@
render_egl_context_ = NULL;
} else {
jclass j_egl_context_class =
- FindClass(jni, "javax/microedition/khronos/egl/EGLContext");
+ FindClass(jni, "org/webrtc/EglBase$Context");
if (!jni->IsInstanceOf(render_egl_context_, j_egl_context_class)) {
ALOGE << "Wrong EGL Context.";
jni->DeleteGlobalRef(render_egl_context_);
@@ -841,7 +888,7 @@
webrtc::VideoDecoder* MediaCodecVideoDecoderFactory::CreateVideoDecoder(
VideoCodecType type) {
if (supported_codec_types_.empty()) {
- ALOGE << "No HW video decoder for type " << (int)type;
+ ALOGW << "No HW video decoder for type " << (int)type;
return NULL;
}
for (VideoCodecType codec_type : supported_codec_types_) {
@@ -851,7 +898,7 @@
AttachCurrentThreadIfNeeded(), type, render_egl_context_);
}
}
- ALOGE << "Can not find HW video decoder for type " << (int)type;
+ ALOGW << "Can not find HW video decoder for type " << (int)type;
return NULL;
}
@@ -861,5 +908,9 @@
delete decoder;
}
+const char* MediaCodecVideoDecoder::ImplementationName() const {
+ return "MediaCodec";
+}
+
} // namespace webrtc_jni
diff --git a/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc b/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
index ac349e7..64831c3 100644
--- a/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
+++ b/talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
@@ -29,14 +29,16 @@
#include "talk/app/webrtc/java/jni/androidmediaencoder_jni.h"
#include "talk/app/webrtc/java/jni/classreferenceholder.h"
#include "talk/app/webrtc/java/jni/androidmediacodeccommon.h"
+#include "talk/app/webrtc/java/jni/native_handle_impl.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
+#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/rtp_rtcp/source/h264_bitstream_parser.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "webrtc/modules/video_coding/utility/include/quality_scaler.h"
-#include "webrtc/modules/video_coding/utility/include/vp8_header_parser.h"
+#include "webrtc/modules/video_coding/include/video_codec_interface.h"
+#include "webrtc/modules/video_coding/utility/quality_scaler.h"
+#include "webrtc/modules/video_coding/utility/vp8_header_parser.h"
#include "webrtc/system_wrappers/include/field_trial.h"
#include "webrtc/system_wrappers/include/logcat_trace_context.h"
#include "third_party/libyuv/include/libyuv/convert.h"
@@ -56,6 +58,7 @@
using webrtc::VideoCodecType;
using webrtc::kVideoCodecH264;
using webrtc::kVideoCodecVP8;
+using webrtc::kVideoCodecVP9;
namespace webrtc_jni {
@@ -79,7 +82,9 @@
public rtc::MessageHandler {
public:
virtual ~MediaCodecVideoEncoder();
- explicit MediaCodecVideoEncoder(JNIEnv* jni, VideoCodecType codecType);
+ MediaCodecVideoEncoder(JNIEnv* jni,
+ VideoCodecType codecType,
+ jobject egl_context);
// webrtc::VideoEncoder implementation. Everything trampolines to
// |codec_thread_| for execution.
@@ -103,13 +108,18 @@
int GetTargetFramerate() override;
+ bool SupportsNativeHandle() const override { return true; }
+ const char* ImplementationName() const override;
+
private:
// CHECK-fail if not running on |codec_thread_|.
void CheckOnCodecThread();
- // Release() and InitEncode() in an attempt to restore the codec to an
+ private:
+ // ResetCodecOnCodecThread() calls ReleaseOnCodecThread() and
+ // InitEncodeOnCodecThread() in an attempt to restore the codec to an
// operable state. Necessary after all manner of OMX-layer errors.
- void ResetCodec();
+ bool ResetCodecOnCodecThread();
// Implementation of webrtc::VideoEncoder methods above, all running on the
// codec thread exclusively.
@@ -117,10 +127,20 @@
// If width==0 then this is assumed to be a re-initialization and the
// previously-current values are reused instead of the passed parameters
// (makes it easier to reason about thread-safety).
- int32_t InitEncodeOnCodecThread(int width, int height, int kbps, int fps);
+ int32_t InitEncodeOnCodecThread(int width, int height, int kbps, int fps,
+ bool use_surface);
+ // Reconfigure to match |frame| in width, height. Also reconfigures the
+ // encoder if |frame| is a texture/byte buffer and the encoder is initialized
+ // for byte buffer/texture. Returns false if reconfiguring fails.
+ bool MaybeReconfigureEncoderOnCodecThread(const webrtc::VideoFrame& frame);
int32_t EncodeOnCodecThread(
const webrtc::VideoFrame& input_image,
const std::vector<webrtc::FrameType>* frame_types);
+ bool EncodeByteBufferOnCodecThread(JNIEnv* jni,
+ bool key_frame, const webrtc::VideoFrame& frame, int input_buffer_index);
+ bool EncodeTextureOnCodecThread(JNIEnv* jni,
+ bool key_frame, const webrtc::VideoFrame& frame);
+
int32_t RegisterEncodeCompleteCallbackOnCodecThread(
webrtc::EncodedImageCallback* callback);
int32_t ReleaseOnCodecThread();
@@ -150,11 +170,14 @@
// State that is constant for the lifetime of this object once the ctor
// returns.
scoped_ptr<Thread> codec_thread_; // Thread on which to operate MediaCodec.
+ rtc::ThreadChecker codec_thread_checker_;
ScopedGlobalRef<jclass> j_media_codec_video_encoder_class_;
ScopedGlobalRef<jobject> j_media_codec_video_encoder_;
jmethodID j_init_encode_method_;
+ jmethodID j_get_input_buffers_method_;
jmethodID j_dequeue_input_buffer_method_;
- jmethodID j_encode_method_;
+ jmethodID j_encode_buffer_method_;
+ jmethodID j_encode_texture_method_;
jmethodID j_release_method_;
jmethodID j_set_rates_method_;
jmethodID j_dequeue_output_buffer_method_;
@@ -170,6 +193,7 @@
int width_; // Frame width in pixels.
int height_; // Frame height in pixels.
bool inited_;
+ bool use_surface_;
uint16_t picture_id_;
enum libyuv::FourCC encoder_fourcc_; // Encoder color space format.
int last_set_bitrate_kbps_; // Last-requested bitrate in kbps.
@@ -205,6 +229,16 @@
// H264 bitstream parser, used to extract QP from encoded bitstreams.
webrtc::H264BitstreamParser h264_bitstream_parser_;
+
+ // VP9 variables to populate codec specific structure.
+ webrtc::GofInfoVP9 gof_; // Contains each frame's temporal information for
+ // non-flexible VP9 mode.
+ uint8_t tl0_pic_idx_;
+ size_t gof_idx_;
+
+ // EGL context - owned by factory, should not be allocated/destroyed
+ // by MediaCodecVideoEncoder.
+ jobject egl_context_;
};
MediaCodecVideoEncoder::~MediaCodecVideoEncoder() {
@@ -213,11 +247,9 @@
}
MediaCodecVideoEncoder::MediaCodecVideoEncoder(
- JNIEnv* jni, VideoCodecType codecType) :
+ JNIEnv* jni, VideoCodecType codecType, jobject egl_context) :
codecType_(codecType),
callback_(NULL),
- inited_(false),
- picture_id_(0),
codec_thread_(new Thread()),
j_media_codec_video_encoder_class_(
jni,
@@ -228,7 +260,11 @@
GetMethodID(jni,
*j_media_codec_video_encoder_class_,
"<init>",
- "()V"))) {
+ "()V"))),
+ inited_(false),
+ use_surface_(false),
+ picture_id_(0),
+ egl_context_(egl_context) {
ScopedLocalRefFrame local_ref_frame(jni);
// It would be nice to avoid spinning up a new thread per MediaCodec, and
// instead re-use e.g. the PeerConnectionFactory's |worker_thread_|, but bug
@@ -239,19 +275,27 @@
// thread.
codec_thread_->SetName("MediaCodecVideoEncoder", NULL);
RTC_CHECK(codec_thread_->Start()) << "Failed to start MediaCodecVideoEncoder";
-
+ codec_thread_checker_.DetachFromThread();
jclass j_output_buffer_info_class =
FindClass(jni, "org/webrtc/MediaCodecVideoEncoder$OutputBufferInfo");
j_init_encode_method_ = GetMethodID(
jni,
*j_media_codec_video_encoder_class_,
"initEncode",
- "(Lorg/webrtc/MediaCodecVideoEncoder$VideoCodecType;IIII)"
- "[Ljava/nio/ByteBuffer;");
+ "(Lorg/webrtc/MediaCodecVideoEncoder$VideoCodecType;"
+ "IIIILorg/webrtc/EglBase14$Context;)Z");
+ j_get_input_buffers_method_ = GetMethodID(
+ jni,
+ *j_media_codec_video_encoder_class_,
+ "getInputBuffers",
+ "()[Ljava/nio/ByteBuffer;");
j_dequeue_input_buffer_method_ = GetMethodID(
jni, *j_media_codec_video_encoder_class_, "dequeueInputBuffer", "()I");
- j_encode_method_ = GetMethodID(
- jni, *j_media_codec_video_encoder_class_, "encode", "(ZIIJ)Z");
+ j_encode_buffer_method_ = GetMethodID(
+ jni, *j_media_codec_video_encoder_class_, "encodeBuffer", "(ZIIJ)Z");
+ j_encode_texture_method_ = GetMethodID(
+ jni, *j_media_codec_video_encoder_class_, "encodeTexture",
+ "(ZI[FJ)Z");
j_release_method_ =
GetMethodID(jni, *j_media_codec_video_encoder_class_, "release", "()V");
j_set_rates_method_ = GetMethodID(
@@ -275,6 +319,7 @@
j_info_presentation_timestamp_us_field_ = GetFieldID(
jni, j_output_buffer_info_class, "presentationTimestampUs", "J");
CHECK_EXCEPTION(jni) << "MediaCodecVideoEncoder ctor failed";
+ srand(time(NULL));
AllowBlockingCalls();
}
@@ -295,8 +340,8 @@
<< codecType_;
ALOGD << "InitEncode request";
- scale_ = webrtc::field_trial::FindFullName(
- "WebRTC-MediaCodecVideoEncoder-AutomaticResize") == "Enabled";
+ scale_ = (codecType_ != kVideoCodecVP9) && (webrtc::field_trial::FindFullName(
+ "WebRTC-MediaCodecVideoEncoder-AutomaticResize") == "Enabled");
ALOGD << "Encoder automatic resize " << (scale_ ? "enabled" : "disabled");
if (scale_) {
if (codecType_ == kVideoCodecVP8) {
@@ -331,7 +376,8 @@
codec_settings->width,
codec_settings->height,
codec_settings->startBitrate,
- codec_settings->maxFramerate));
+ codec_settings->maxFramerate,
+ false /* use_surface */));
}
int32_t MediaCodecVideoEncoder::Encode(
@@ -374,6 +420,7 @@
}
void MediaCodecVideoEncoder::OnMessage(rtc::Message* msg) {
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
@@ -381,7 +428,6 @@
// functor), so expect no ID/data.
RTC_CHECK(!msg->message_id) << "Unexpected message!";
RTC_CHECK(!msg->pdata) << "Unexpected message!";
- CheckOnCodecThread();
if (!inited_) {
return;
}
@@ -393,26 +439,24 @@
codec_thread_->PostDelayed(kMediaCodecPollMs, this);
}
-void MediaCodecVideoEncoder::CheckOnCodecThread() {
- RTC_CHECK(codec_thread_ == ThreadManager::Instance()->CurrentThread())
- << "Running on wrong thread!";
-}
-
-void MediaCodecVideoEncoder::ResetCodec() {
- ALOGE << "ResetCodec";
- if (Release() != WEBRTC_VIDEO_CODEC_OK ||
- codec_thread_->Invoke<int32_t>(Bind(
- &MediaCodecVideoEncoder::InitEncodeOnCodecThread, this,
- width_, height_, 0, 0)) != WEBRTC_VIDEO_CODEC_OK) {
+bool MediaCodecVideoEncoder::ResetCodecOnCodecThread() {
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
+ ALOGE << "ResetOnCodecThread";
+ if (ReleaseOnCodecThread() != WEBRTC_VIDEO_CODEC_OK ||
+ InitEncodeOnCodecThread(width_, height_, 0, 0, false) !=
+ WEBRTC_VIDEO_CODEC_OK) {
// TODO(fischman): wouldn't it be nice if there was a way to gracefully
// degrade to a SW encoder at this point? There isn't one AFAICT :(
// https://code.google.com/p/webrtc/issues/detail?id=2920
+ return false;
}
+ return true;
}
int32_t MediaCodecVideoEncoder::InitEncodeOnCodecThread(
- int width, int height, int kbps, int fps) {
- CheckOnCodecThread();
+ int width, int height, int kbps, int fps, bool use_surface) {
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
+ RTC_CHECK(!use_surface || egl_context_ != nullptr) << "EGL context not set.";
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
@@ -448,52 +492,63 @@
render_times_ms_.clear();
frame_rtc_times_ms_.clear();
drop_next_input_frame_ = false;
+ use_surface_ = use_surface;
picture_id_ = static_cast<uint16_t>(rand()) & 0x7FFF;
+ gof_.SetGofInfoVP9(webrtc::TemporalStructureMode::kTemporalStructureMode1);
+ tl0_pic_idx_ = static_cast<uint8_t>(rand());
+ gof_idx_ = 0;
+
// We enforce no extra stride/padding in the format creation step.
jobject j_video_codec_enum = JavaEnumFromIndex(
jni, "MediaCodecVideoEncoder$VideoCodecType", codecType_);
- jobjectArray input_buffers = reinterpret_cast<jobjectArray>(
- jni->CallObjectMethod(*j_media_codec_video_encoder_,
- j_init_encode_method_,
- j_video_codec_enum,
- width_,
- height_,
- kbps,
- fps));
- CHECK_EXCEPTION(jni);
- if (IsNull(jni, input_buffers)) {
+ const bool encode_status = jni->CallBooleanMethod(
+ *j_media_codec_video_encoder_, j_init_encode_method_,
+ j_video_codec_enum, width, height, kbps, fps,
+ (use_surface ? egl_context_ : nullptr));
+ if (!encode_status) {
+ ALOGE << "Failed to configure encoder.";
return WEBRTC_VIDEO_CODEC_ERROR;
}
+ CHECK_EXCEPTION(jni);
+
+ if (!use_surface) {
+ jobjectArray input_buffers = reinterpret_cast<jobjectArray>(
+ jni->CallObjectMethod(*j_media_codec_video_encoder_,
+ j_get_input_buffers_method_));
+ CHECK_EXCEPTION(jni);
+ if (IsNull(jni, input_buffers)) {
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+
+ switch (GetIntField(jni, *j_media_codec_video_encoder_,
+ j_color_format_field_)) {
+ case COLOR_FormatYUV420Planar:
+ encoder_fourcc_ = libyuv::FOURCC_YU12;
+ break;
+ case COLOR_FormatYUV420SemiPlanar:
+ case COLOR_QCOM_FormatYUV420SemiPlanar:
+ case COLOR_QCOM_FORMATYUV420PackedSemiPlanar32m:
+ encoder_fourcc_ = libyuv::FOURCC_NV12;
+ break;
+ default:
+ LOG(LS_ERROR) << "Wrong color format.";
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ size_t num_input_buffers = jni->GetArrayLength(input_buffers);
+ RTC_CHECK(input_buffers_.empty())
+ << "Unexpected double InitEncode without Release";
+ input_buffers_.resize(num_input_buffers);
+ for (size_t i = 0; i < num_input_buffers; ++i) {
+ input_buffers_[i] =
+ jni->NewGlobalRef(jni->GetObjectArrayElement(input_buffers, i));
+ int64_t yuv_buffer_capacity =
+ jni->GetDirectBufferCapacity(input_buffers_[i]);
+ CHECK_EXCEPTION(jni);
+ RTC_CHECK(yuv_buffer_capacity >= yuv_size_) << "Insufficient capacity";
+ }
+ }
inited_ = true;
- switch (GetIntField(jni, *j_media_codec_video_encoder_,
- j_color_format_field_)) {
- case COLOR_FormatYUV420Planar:
- encoder_fourcc_ = libyuv::FOURCC_YU12;
- break;
- case COLOR_FormatYUV420SemiPlanar:
- case COLOR_QCOM_FormatYUV420SemiPlanar:
- case COLOR_QCOM_FORMATYUV420PackedSemiPlanar32m:
- encoder_fourcc_ = libyuv::FOURCC_NV12;
- break;
- default:
- LOG(LS_ERROR) << "Wrong color format.";
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- size_t num_input_buffers = jni->GetArrayLength(input_buffers);
- RTC_CHECK(input_buffers_.empty())
- << "Unexpected double InitEncode without Release";
- input_buffers_.resize(num_input_buffers);
- for (size_t i = 0; i < num_input_buffers; ++i) {
- input_buffers_[i] =
- jni->NewGlobalRef(jni->GetObjectArrayElement(input_buffers, i));
- int64_t yuv_buffer_capacity =
- jni->GetDirectBufferCapacity(input_buffers_[i]);
- CHECK_EXCEPTION(jni);
- RTC_CHECK(yuv_buffer_capacity >= yuv_size_) << "Insufficient capacity";
- }
- CHECK_EXCEPTION(jni);
-
codec_thread_->PostDelayed(kMediaCodecPollMs, this);
return WEBRTC_VIDEO_CODEC_OK;
}
@@ -501,40 +556,53 @@
int32_t MediaCodecVideoEncoder::EncodeOnCodecThread(
const webrtc::VideoFrame& frame,
const std::vector<webrtc::FrameType>* frame_types) {
- CheckOnCodecThread();
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
if (!inited_) {
return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
}
+
frames_received_++;
if (!DeliverPendingOutputs(jni)) {
- ResetCodec();
- // Continue as if everything's fine.
+ if (!ResetCodecOnCodecThread())
+ return WEBRTC_VIDEO_CODEC_ERROR;
}
if (drop_next_input_frame_) {
- ALOGV("Encoder drop frame - failed callback.");
+ ALOGW << "Encoder drop frame - failed callback.";
drop_next_input_frame_ = false;
return WEBRTC_VIDEO_CODEC_OK;
}
RTC_CHECK(frame_types->size() == 1) << "Unexpected stream count";
- // Check framerate before spatial resolution change.
- if (scale_)
+
+ VideoFrame input_frame = frame;
+ if (scale_) {
+ // Check framerate before spatial resolution change.
quality_scaler_.OnEncodeFrame(frame);
+ const webrtc::QualityScaler::Resolution scaled_resolution =
+ quality_scaler_.GetScaledResolution();
+ if (scaled_resolution.width != frame.width() ||
+ scaled_resolution.height != frame.height()) {
+ if (frame.native_handle() != nullptr) {
+ rtc::scoped_refptr<webrtc::VideoFrameBuffer> scaled_buffer(
+ static_cast<AndroidTextureBuffer*>(
+ frame.video_frame_buffer().get())->ScaleAndRotate(
+ scaled_resolution.width,
+ scaled_resolution.height,
+ webrtc::kVideoRotation_0));
+ input_frame.set_video_frame_buffer(scaled_buffer);
+ } else {
+ input_frame = quality_scaler_.GetScaledFrame(frame);
+ }
+ }
+ }
- const VideoFrame& input_frame =
- scale_ ? quality_scaler_.GetScaledFrame(frame) : frame;
-
- if (input_frame.width() != width_ || input_frame.height() != height_) {
- ALOGD << "Frame resolution change from " << width_ << " x " << height_ <<
- " to " << input_frame.width() << " x " << input_frame.height();
- width_ = input_frame.width();
- height_ = input_frame.height();
- ResetCodec();
- return WEBRTC_VIDEO_CODEC_OK;
+ if (!MaybeReconfigureEncoderOnCodecThread(input_frame)) {
+ ALOGE << "Failed to reconfigure encoder.";
+ return WEBRTC_VIDEO_CODEC_ERROR;
}
// Check if we accumulated too many frames in encoder input buffers
@@ -552,65 +620,138 @@
}
}
- int j_input_buffer_index = jni->CallIntMethod(*j_media_codec_video_encoder_,
- j_dequeue_input_buffer_method_);
- CHECK_EXCEPTION(jni);
- if (j_input_buffer_index == -1) {
- // Video codec falls behind - no input buffer available.
- ALOGV("Encoder drop frame - no input buffers available");
- frames_dropped_++;
- // Report dropped frame to quality_scaler_.
- OnDroppedFrame();
- return WEBRTC_VIDEO_CODEC_OK; // TODO(fischman): see webrtc bug 2887.
+ const bool key_frame = frame_types->front() != webrtc::kVideoFrameDelta;
+ bool encode_status = true;
+ if (!input_frame.native_handle()) {
+ int j_input_buffer_index = jni->CallIntMethod(*j_media_codec_video_encoder_,
+ j_dequeue_input_buffer_method_);
+ CHECK_EXCEPTION(jni);
+ if (j_input_buffer_index == -1) {
+ // Video codec falls behind - no input buffer available.
+ ALOGW << "Encoder drop frame - no input buffers available";
+ frames_dropped_++;
+ // Report dropped frame to quality_scaler_.
+ OnDroppedFrame();
+ return WEBRTC_VIDEO_CODEC_OK; // TODO(fischman): see webrtc bug 2887.
+ }
+ if (j_input_buffer_index == -2) {
+ ResetCodecOnCodecThread();
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ encode_status = EncodeByteBufferOnCodecThread(jni, key_frame, input_frame,
+ j_input_buffer_index);
+ } else {
+ encode_status = EncodeTextureOnCodecThread(jni, key_frame, input_frame);
}
- if (j_input_buffer_index == -2) {
- ResetCodec();
+
+ if (!encode_status) {
+ ALOGE << "Failed encode frame with timestamp: " << input_frame.timestamp();
+ ResetCodecOnCodecThread();
return WEBRTC_VIDEO_CODEC_ERROR;
}
- ALOGV("Encoder frame in # %d. TS: %lld. Q: %d",
- frames_received_ - 1, current_timestamp_us_ / 1000, frames_in_queue_);
-
- jobject j_input_buffer = input_buffers_[j_input_buffer_index];
- uint8_t* yuv_buffer =
- reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(j_input_buffer));
- CHECK_EXCEPTION(jni);
- RTC_CHECK(yuv_buffer) << "Indirect buffer??";
- RTC_CHECK(!libyuv::ConvertFromI420(
- input_frame.buffer(webrtc::kYPlane), input_frame.stride(webrtc::kYPlane),
- input_frame.buffer(webrtc::kUPlane), input_frame.stride(webrtc::kUPlane),
- input_frame.buffer(webrtc::kVPlane), input_frame.stride(webrtc::kVPlane),
- yuv_buffer, width_, width_, height_, encoder_fourcc_))
- << "ConvertFromI420 failed";
- last_input_timestamp_ms_ = current_timestamp_us_ / 1000;
+ last_input_timestamp_ms_ =
+ current_timestamp_us_ / rtc::kNumMicrosecsPerMillisec;
frames_in_queue_++;
// Save input image timestamps for later output
timestamps_.push_back(input_frame.timestamp());
render_times_ms_.push_back(input_frame.render_time_ms());
frame_rtc_times_ms_.push_back(GetCurrentTimeMs());
+ current_timestamp_us_ += rtc::kNumMicrosecsPerSec / last_set_fps_;
- bool key_frame = frame_types->front() != webrtc::kVideoFrameDelta;
+ if (!DeliverPendingOutputs(jni)) {
+ ALOGE << "Failed deliver pending outputs.";
+ ResetCodecOnCodecThread();
+ return WEBRTC_VIDEO_CODEC_ERROR;
+ }
+ return WEBRTC_VIDEO_CODEC_OK;
+}
+
+bool MediaCodecVideoEncoder::MaybeReconfigureEncoderOnCodecThread(
+ const webrtc::VideoFrame& frame) {
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
+
+ const bool is_texture_frame = frame.native_handle() != nullptr;
+ const bool reconfigure_due_to_format = is_texture_frame != use_surface_;
+ const bool reconfigure_due_to_size =
+ frame.width() != width_ || frame.height() != height_;
+
+ if (reconfigure_due_to_format) {
+ ALOGD << "Reconfigure encoder due to format change. "
+ << (use_surface_ ?
+ "Reconfiguring to encode from byte buffer." :
+ "Reconfiguring to encode from texture.");
+ }
+ if (reconfigure_due_to_size) {
+ ALOGD << "Reconfigure encoder due to frame resolution change from "
+ << width_ << " x " << height_ << " to " << frame.width() << " x "
+ << frame.height();
+ width_ = frame.width();
+ height_ = frame.height();
+ }
+
+ if (!reconfigure_due_to_format && !reconfigure_due_to_size)
+ return true;
+
+ ReleaseOnCodecThread();
+
+ return InitEncodeOnCodecThread(width_, height_, 0, 0 , is_texture_frame) ==
+ WEBRTC_VIDEO_CODEC_OK;
+}
+
+bool MediaCodecVideoEncoder::EncodeByteBufferOnCodecThread(JNIEnv* jni,
+ bool key_frame, const webrtc::VideoFrame& frame, int input_buffer_index) {
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
+ RTC_CHECK(!use_surface_);
+
+ ALOGV("Encoder frame in # %d. TS: %lld. Q: %d",
+ frames_received_ - 1, current_timestamp_us_ / 1000, frames_in_queue_);
+
+ jobject j_input_buffer = input_buffers_[input_buffer_index];
+ uint8_t* yuv_buffer =
+ reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(j_input_buffer));
+ CHECK_EXCEPTION(jni);
+ RTC_CHECK(yuv_buffer) << "Indirect buffer??";
+ RTC_CHECK(!libyuv::ConvertFromI420(
+ frame.buffer(webrtc::kYPlane), frame.stride(webrtc::kYPlane),
+ frame.buffer(webrtc::kUPlane), frame.stride(webrtc::kUPlane),
+ frame.buffer(webrtc::kVPlane), frame.stride(webrtc::kVPlane),
+ yuv_buffer, width_, width_, height_, encoder_fourcc_))
+ << "ConvertFromI420 failed";
+
bool encode_status = jni->CallBooleanMethod(*j_media_codec_video_encoder_,
- j_encode_method_,
+ j_encode_buffer_method_,
key_frame,
- j_input_buffer_index,
+ input_buffer_index,
yuv_size_,
current_timestamp_us_);
CHECK_EXCEPTION(jni);
- current_timestamp_us_ += 1000000 / last_set_fps_;
+ return encode_status;
+}
- if (!encode_status || !DeliverPendingOutputs(jni)) {
- ResetCodec();
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
+bool MediaCodecVideoEncoder::EncodeTextureOnCodecThread(JNIEnv* jni,
+ bool key_frame, const webrtc::VideoFrame& frame) {
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
+ RTC_CHECK(use_surface_);
+ NativeHandleImpl* handle =
+ static_cast<NativeHandleImpl*>(frame.native_handle());
+ jfloatArray sampling_matrix = jni->NewFloatArray(16);
+ jni->SetFloatArrayRegion(sampling_matrix, 0, 16, handle->sampling_matrix);
- return WEBRTC_VIDEO_CODEC_OK;
+ bool encode_status = jni->CallBooleanMethod(*j_media_codec_video_encoder_,
+ j_encode_texture_method_,
+ key_frame,
+ handle->oes_texture_id,
+ sampling_matrix,
+ current_timestamp_us_);
+ CHECK_EXCEPTION(jni);
+ return encode_status;
}
int32_t MediaCodecVideoEncoder::RegisterEncodeCompleteCallbackOnCodecThread(
webrtc::EncodedImageCallback* callback) {
- CheckOnCodecThread();
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
callback_ = callback;
@@ -618,10 +759,10 @@
}
int32_t MediaCodecVideoEncoder::ReleaseOnCodecThread() {
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
if (!inited_) {
return WEBRTC_VIDEO_CODEC_OK;
}
- CheckOnCodecThread();
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ALOGD << "EncoderReleaseOnCodecThread: Frames received: " <<
frames_received_ << ". Encoded: " << frames_encoded_ <<
@@ -634,13 +775,14 @@
CHECK_EXCEPTION(jni);
rtc::MessageQueueManager::Clear(this);
inited_ = false;
+ use_surface_ = false;
ALOGD << "EncoderReleaseOnCodecThread done.";
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t MediaCodecVideoEncoder::SetRatesOnCodecThread(uint32_t new_bit_rate,
uint32_t frame_rate) {
- CheckOnCodecThread();
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
if (last_set_bitrate_kbps_ == new_bit_rate &&
last_set_fps_ == frame_rate) {
return WEBRTC_VIDEO_CODEC_OK;
@@ -659,7 +801,7 @@
last_set_fps_);
CHECK_EXCEPTION(jni);
if (!ret) {
- ResetCodec();
+ ResetCodecOnCodecThread();
return WEBRTC_VIDEO_CODEC_ERROR;
}
return WEBRTC_VIDEO_CODEC_OK;
@@ -691,6 +833,7 @@
}
bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
+ RTC_DCHECK(codec_thread_checker_.CalledOnValidThread());
while (true) {
jobject j_output_buffer_info = jni->CallObjectMethod(
*j_media_codec_video_encoder_, j_dequeue_output_buffer_method_);
@@ -702,7 +845,7 @@
int output_buffer_index =
GetOutputBufferInfoIndex(jni, j_output_buffer_info);
if (output_buffer_index == -1) {
- ResetCodec();
+ ResetCodecOnCodecThread();
return false;
}
@@ -786,19 +929,42 @@
info.codecSpecific.VP8.layerSync = false;
info.codecSpecific.VP8.tl0PicIdx = webrtc::kNoTl0PicIdx;
info.codecSpecific.VP8.keyIdx = webrtc::kNoKeyIdx;
- picture_id_ = (picture_id_ + 1) & 0x7FFF;
+ } else if (codecType_ == kVideoCodecVP9) {
+ if (key_frame) {
+ gof_idx_ = 0;
+ }
+ info.codecSpecific.VP9.picture_id = picture_id_;
+ info.codecSpecific.VP9.inter_pic_predicted = key_frame ? false : true;
+ info.codecSpecific.VP9.flexible_mode = false;
+ info.codecSpecific.VP9.ss_data_available = key_frame ? true : false;
+ info.codecSpecific.VP9.tl0_pic_idx = tl0_pic_idx_++;
+ info.codecSpecific.VP9.temporal_idx = webrtc::kNoTemporalIdx;
+ info.codecSpecific.VP9.spatial_idx = webrtc::kNoSpatialIdx;
+ info.codecSpecific.VP9.temporal_up_switch = true;
+ info.codecSpecific.VP9.inter_layer_predicted = false;
+ info.codecSpecific.VP9.gof_idx =
+ static_cast<uint8_t>(gof_idx_++ % gof_.num_frames_in_gof);
+ info.codecSpecific.VP9.num_spatial_layers = 1;
+ info.codecSpecific.VP9.spatial_layer_resolution_present = false;
+ if (info.codecSpecific.VP9.ss_data_available) {
+ info.codecSpecific.VP9.spatial_layer_resolution_present = true;
+ info.codecSpecific.VP9.width[0] = width_;
+ info.codecSpecific.VP9.height[0] = height_;
+ info.codecSpecific.VP9.gof.CopyGofInfoVP9(gof_);
+ }
}
+ picture_id_ = (picture_id_ + 1) & 0x7FFF;
// Generate a header describing a single fragment.
webrtc::RTPFragmentationHeader header;
memset(&header, 0, sizeof(header));
- if (codecType_ == kVideoCodecVP8) {
+ if (codecType_ == kVideoCodecVP8 || codecType_ == kVideoCodecVP9) {
header.VerifyAndAllocateFragmentationHeader(1);
header.fragmentationOffset[0] = 0;
header.fragmentationLength[0] = image->_length;
header.fragmentationPlType[0] = 0;
header.fragmentationTimeDiff[0] = 0;
- if (scale_) {
+ if (codecType_ == kVideoCodecVP8 && scale_) {
int qp;
if (webrtc::vp8::GetQp(payload, payload_size, &qp))
quality_scaler_.ReportQP(qp);
@@ -829,7 +995,7 @@
ALOGE << "Data:" << image->_buffer[0] << " " << image->_buffer[1]
<< " " << image->_buffer[2] << " " << image->_buffer[3]
<< " " << image->_buffer[4] << " " << image->_buffer[5];
- ResetCodec();
+ ResetCodecOnCodecThread();
return false;
}
scPositions[scPositionsLength] = payload_size;
@@ -852,7 +1018,7 @@
output_buffer_index);
CHECK_EXCEPTION(jni);
if (!success) {
- ResetCodec();
+ ResetCodecOnCodecThread();
return false;
}
@@ -907,7 +1073,12 @@
return scale_ ? quality_scaler_.GetTargetFramerate() : -1;
}
-MediaCodecVideoEncoderFactory::MediaCodecVideoEncoderFactory() {
+const char* MediaCodecVideoEncoder::ImplementationName() const {
+ return "MediaCodec";
+}
+
+MediaCodecVideoEncoderFactory::MediaCodecVideoEncoderFactory()
+ : egl_context_(nullptr) {
JNIEnv* jni = AttachCurrentThreadIfNeeded();
ScopedLocalRefFrame local_ref_frame(jni);
jclass j_encoder_class = FindClass(jni, "org/webrtc/MediaCodecVideoEncoder");
@@ -923,6 +1094,16 @@
MAX_VIDEO_WIDTH, MAX_VIDEO_HEIGHT, MAX_VIDEO_FPS));
}
+ bool is_vp9_hw_supported = jni->CallStaticBooleanMethod(
+ j_encoder_class,
+ GetStaticMethodID(jni, j_encoder_class, "isVp9HwSupported", "()Z"));
+ CHECK_EXCEPTION(jni);
+ if (is_vp9_hw_supported) {
+ ALOGD << "VP9 HW Encoder supported.";
+ supported_codecs_.push_back(VideoCodec(kVideoCodecVP9, "VP9",
+ MAX_VIDEO_WIDTH, MAX_VIDEO_HEIGHT, MAX_VIDEO_FPS));
+ }
+
bool is_h264_hw_supported = jni->CallStaticBooleanMethod(
j_encoder_class,
GetStaticMethodID(jni, j_encoder_class, "isH264HwSupported", "()Z"));
@@ -936,9 +1117,37 @@
MediaCodecVideoEncoderFactory::~MediaCodecVideoEncoderFactory() {}
+void MediaCodecVideoEncoderFactory::SetEGLContext(
+ JNIEnv* jni, jobject render_egl_context) {
+ ALOGD << "MediaCodecVideoEncoderFactory::SetEGLContext";
+ if (egl_context_) {
+ jni->DeleteGlobalRef(egl_context_);
+ egl_context_ = NULL;
+ }
+ if (!IsNull(jni, render_egl_context)) {
+ egl_context_ = jni->NewGlobalRef(render_egl_context);
+ if (CheckException(jni)) {
+ ALOGE << "error calling NewGlobalRef for EGL Context.";
+ egl_context_ = NULL;
+ } else {
+ jclass j_egl_context_class =
+ FindClass(jni, "org/webrtc/EglBase14$Context");
+ if (!jni->IsInstanceOf(egl_context_, j_egl_context_class)) {
+ ALOGE << "Wrong EGL Context.";
+ jni->DeleteGlobalRef(egl_context_);
+ egl_context_ = NULL;
+ }
+ }
+ }
+ if (egl_context_ == NULL) {
+ ALOGW << "NULL VideoDecoder EGL context - HW surface encoding is disabled.";
+ }
+}
+
webrtc::VideoEncoder* MediaCodecVideoEncoderFactory::CreateVideoEncoder(
VideoCodecType type) {
if (supported_codecs_.empty()) {
+ ALOGW << "No HW video encoder for type " << (int)type;
return NULL;
}
for (std::vector<VideoCodec>::const_iterator it = supported_codecs_.begin();
@@ -946,9 +1155,11 @@
if (it->type == type) {
ALOGD << "Create HW video encoder for type " << (int)type <<
" (" << it->name << ").";
- return new MediaCodecVideoEncoder(AttachCurrentThreadIfNeeded(), type);
+ return new MediaCodecVideoEncoder(AttachCurrentThreadIfNeeded(), type,
+ egl_context_);
}
}
+ ALOGW << "Can not find HW video encoder for type " << (int)type;
return NULL;
}
diff --git a/talk/app/webrtc/java/jni/androidmediaencoder_jni.h b/talk/app/webrtc/java/jni/androidmediaencoder_jni.h
index ff124aa..8ff8164 100644
--- a/talk/app/webrtc/java/jni/androidmediaencoder_jni.h
+++ b/talk/app/webrtc/java/jni/androidmediaencoder_jni.h
@@ -43,6 +43,8 @@
MediaCodecVideoEncoderFactory();
virtual ~MediaCodecVideoEncoderFactory();
+ void SetEGLContext(JNIEnv* jni, jobject render_egl_context);
+
// WebRtcVideoEncoderFactory implementation.
webrtc::VideoEncoder* CreateVideoEncoder(webrtc::VideoCodecType type)
override;
@@ -50,6 +52,7 @@
void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) override;
private:
+ jobject egl_context_;
// Empty if platform support is lacking, const after ctor returns.
std::vector<VideoCodec> supported_codecs_;
};
diff --git a/talk/app/webrtc/java/jni/androidvideocapturer_jni.cc b/talk/app/webrtc/java/jni/androidvideocapturer_jni.cc
index 02b9f22..8813c89 100644
--- a/talk/app/webrtc/java/jni/androidvideocapturer_jni.cc
+++ b/talk/app/webrtc/java/jni/androidvideocapturer_jni.cc
@@ -29,8 +29,9 @@
#include "talk/app/webrtc/java/jni/androidvideocapturer_jni.h"
#include "talk/app/webrtc/java/jni/classreferenceholder.h"
#include "talk/app/webrtc/java/jni/native_handle_impl.h"
+#include "talk/app/webrtc/java/jni/surfacetexturehelper_jni.h"
+#include "third_party/libyuv/include/libyuv/convert.h"
#include "webrtc/base/bind.h"
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
namespace webrtc_jni {
@@ -47,15 +48,19 @@
return 0;
}
-AndroidVideoCapturerJni::AndroidVideoCapturerJni(JNIEnv* jni,
- jobject j_video_capturer)
- : j_capturer_global_(jni, j_video_capturer),
+AndroidVideoCapturerJni::AndroidVideoCapturerJni(
+ JNIEnv* jni,
+ jobject j_video_capturer,
+ jobject j_surface_texture_helper)
+ : j_video_capturer_(jni, j_video_capturer),
j_video_capturer_class_(
jni, FindClass(jni, "org/webrtc/VideoCapturerAndroid")),
j_observer_class_(
jni,
FindClass(jni,
"org/webrtc/VideoCapturerAndroid$NativeObserver")),
+ surface_texture_helper_(new rtc::RefCountedObject<SurfaceTextureHelper>(
+ jni, j_surface_texture_helper)),
capturer_(nullptr) {
LOG(LS_INFO) << "AndroidVideoCapturerJni ctor";
thread_checker_.DetachFromThread();
@@ -64,7 +69,7 @@
AndroidVideoCapturerJni::~AndroidVideoCapturerJni() {
LOG(LS_INFO) << "AndroidVideoCapturerJni dtor";
jni()->CallVoidMethod(
- *j_capturer_global_,
+ *j_video_capturer_,
GetMethodID(jni(), *j_video_capturer_class_, "release", "()V"));
CHECK_EXCEPTION(jni()) << "error during VideoCapturerAndroid.release()";
}
@@ -90,7 +95,7 @@
jni(), *j_video_capturer_class_, "startCapture",
"(IIILandroid/content/Context;"
"Lorg/webrtc/VideoCapturerAndroid$CapturerObserver;)V");
- jni()->CallVoidMethod(*j_capturer_global_,
+ jni()->CallVoidMethod(*j_video_capturer_,
m, width, height,
framerate,
application_context_,
@@ -109,7 +114,7 @@
}
jmethodID m = GetMethodID(jni(), *j_video_capturer_class_,
"stopCapture", "()V");
- jni()->CallVoidMethod(*j_capturer_global_, m);
+ jni()->CallVoidMethod(*j_video_capturer_, m);
CHECK_EXCEPTION(jni()) << "error during VideoCapturerAndroid.stopCapture";
LOG(LS_INFO) << "AndroidVideoCapturerJni stop done";
}
@@ -127,19 +132,12 @@
invoker_->AsyncInvoke<void>(rtc::Bind(method, capturer_, args...));
}
-void AndroidVideoCapturerJni::ReturnBuffer(int64_t time_stamp) {
- jmethodID m = GetMethodID(jni(), *j_video_capturer_class_,
- "returnBuffer", "(J)V");
- jni()->CallVoidMethod(*j_capturer_global_, m, time_stamp);
- CHECK_EXCEPTION(jni()) << "error during VideoCapturerAndroid.returnBuffer";
-}
-
std::string AndroidVideoCapturerJni::GetSupportedFormats() {
jmethodID m =
GetMethodID(jni(), *j_video_capturer_class_,
"getSupportedFormatsAsJson", "()Ljava/lang/String;");
jstring j_json_caps =
- (jstring) jni()->CallObjectMethod(*j_capturer_global_, m);
+ (jstring) jni()->CallObjectMethod(*j_video_capturer_, m);
CHECK_EXCEPTION(jni()) << "error during supportedFormatsAsJson";
return JavaToStdString(jni(), j_json_caps);
}
@@ -158,46 +156,33 @@
int rotation,
int64_t timestamp_ns) {
const uint8_t* y_plane = static_cast<uint8_t*>(video_frame);
- // Android guarantees that the stride is a multiple of 16.
- // http://developer.android.com/reference/android/hardware/Camera.Parameters.html#setPreviewFormat%28int%29
- int y_stride;
- int uv_stride;
- webrtc::Calc16ByteAlignedStride(width, &y_stride, &uv_stride);
- const uint8_t* v_plane = y_plane + y_stride * height;
- const uint8_t* u_plane =
- v_plane + uv_stride * webrtc::AlignInt(height, 2) / 2;
+ const uint8_t* vu_plane = y_plane + width * height;
- // Wrap the Java buffer, and call ReturnBuffer() in the wrapped
- // VideoFrameBuffer destructor.
- rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer(
- new rtc::RefCountedObject<webrtc::WrappedI420Buffer>(
- width, height, y_plane, y_stride, u_plane, uv_stride, v_plane,
- uv_stride,
- rtc::Bind(&AndroidVideoCapturerJni::ReturnBuffer, this,
- timestamp_ns)));
+ rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer =
+ buffer_pool_.CreateBuffer(width, height);
+ libyuv::NV21ToI420(
+ y_plane, width,
+ vu_plane, width,
+ buffer->MutableData(webrtc::kYPlane), buffer->stride(webrtc::kYPlane),
+ buffer->MutableData(webrtc::kUPlane), buffer->stride(webrtc::kUPlane),
+ buffer->MutableData(webrtc::kVPlane), buffer->stride(webrtc::kVPlane),
+ width, height);
AsyncCapturerInvoke("OnIncomingFrame",
&webrtc::AndroidVideoCapturer::OnIncomingFrame,
buffer, rotation, timestamp_ns);
}
-void AndroidVideoCapturerJni::OnTextureFrame(
- int width,
- int height,
- int64_t timestamp_ns,
- const NativeTextureHandleImpl& handle) {
- // TODO(magjed): Fix this. See bug webrtc:4993.
- RTC_NOTREACHED()
- << "The rest of the stack for Android expects the native "
- "handle to be a NativeHandleImpl with a SurfaceTexture, not a "
- "NativeTextureHandleImpl";
+void AndroidVideoCapturerJni::OnTextureFrame(int width,
+ int height,
+ int rotation,
+ int64_t timestamp_ns,
+ const NativeHandleImpl& handle) {
rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer(
- new rtc::RefCountedObject<AndroidTextureBuffer>(
- width, height, handle,
- rtc::Bind(&AndroidVideoCapturerJni::ReturnBuffer, this,
- timestamp_ns)));
+ surface_texture_helper_->CreateTextureFrame(width, height, handle));
+
AsyncCapturerInvoke("OnIncomingFrame",
&webrtc::AndroidVideoCapturer::OnIncomingFrame,
- buffer, 0, timestamp_ns);
+ buffer, rotation, timestamp_ns);
}
void AndroidVideoCapturerJni::OnOutputFormatRequest(int width,
@@ -216,13 +201,6 @@
jint width, jint height, jint rotation, jlong timestamp) {
jboolean is_copy = true;
jbyte* bytes = jni->GetByteArrayElements(j_frame, &is_copy);
- // If this is a copy of the original frame, it means that the memory
- // is not direct memory and thus VideoCapturerAndroid does not guarantee
- // that the memory is valid when we have released |j_frame|.
- // TODO(magjed): Move ReleaseByteArrayElements() into ReturnBuffer() and
- // remove this check.
- RTC_CHECK(!is_copy)
- << "NativeObserver_nativeOnFrameCaptured: frame is a copy";
reinterpret_cast<AndroidVideoCapturerJni*>(j_capturer)
->OnMemoryBufferFrame(bytes, length, width, height, rotation, timestamp);
jni->ReleaseByteArrayElements(j_frame, bytes, JNI_ABORT);
@@ -231,11 +209,11 @@
JOW(void, VideoCapturerAndroid_00024NativeObserver_nativeOnTextureFrameCaptured)
(JNIEnv* jni, jclass, jlong j_capturer, jint j_width, jint j_height,
jint j_oes_texture_id, jfloatArray j_transform_matrix,
- jlong j_timestamp) {
+ jint j_rotation, jlong j_timestamp) {
reinterpret_cast<AndroidVideoCapturerJni*>(j_capturer)
- ->OnTextureFrame(j_width, j_height, j_timestamp,
- NativeTextureHandleImpl(jni, j_oes_texture_id,
- j_transform_matrix));
+ ->OnTextureFrame(j_width, j_height, j_rotation, j_timestamp,
+ NativeHandleImpl(jni, j_oes_texture_id,
+ j_transform_matrix));
}
JOW(void, VideoCapturerAndroid_00024NativeObserver_nativeCapturerStarted)
@@ -254,9 +232,11 @@
}
JOW(jlong, VideoCapturerAndroid_nativeCreateVideoCapturer)
- (JNIEnv* jni, jclass, jobject j_video_capturer) {
+ (JNIEnv* jni, jclass,
+ jobject j_video_capturer, jobject j_surface_texture_helper) {
rtc::scoped_refptr<webrtc::AndroidVideoCapturerDelegate> delegate =
- new rtc::RefCountedObject<AndroidVideoCapturerJni>(jni, j_video_capturer);
+ new rtc::RefCountedObject<AndroidVideoCapturerJni>(
+ jni, j_video_capturer, j_surface_texture_helper);
rtc::scoped_ptr<cricket::VideoCapturer> capturer(
new webrtc::AndroidVideoCapturer(delegate));
// Caller takes ownership of the cricket::VideoCapturer* pointer.
diff --git a/talk/app/webrtc/java/jni/androidvideocapturer_jni.h b/talk/app/webrtc/java/jni/androidvideocapturer_jni.h
index d1eb3a0..89ecacb 100644
--- a/talk/app/webrtc/java/jni/androidvideocapturer_jni.h
+++ b/talk/app/webrtc/java/jni/androidvideocapturer_jni.h
@@ -36,10 +36,12 @@
#include "webrtc/base/asyncinvoker.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_checker.h"
+#include "webrtc/common_video/include/i420_buffer_pool.h"
namespace webrtc_jni {
-class NativeTextureHandleImpl;
+struct NativeHandleImpl;
+class SurfaceTextureHelper;
// AndroidVideoCapturerJni implements AndroidVideoCapturerDelegate.
// The purpose of the delegate is to hide the JNI specifics from the C++ only
@@ -48,7 +50,9 @@
public:
static int SetAndroidObjects(JNIEnv* jni, jobject appliction_context);
- AndroidVideoCapturerJni(JNIEnv* jni, jobject j_video_capturer);
+ AndroidVideoCapturerJni(JNIEnv* jni,
+ jobject j_video_capturer,
+ jobject j_surface_texture_helper);
void Start(int width, int height, int framerate,
webrtc::AndroidVideoCapturer* capturer) override;
@@ -60,15 +64,14 @@
void OnCapturerStarted(bool success);
void OnMemoryBufferFrame(void* video_frame, int length, int width,
int height, int rotation, int64_t timestamp_ns);
- void OnTextureFrame(int width, int height, int64_t timestamp_ns,
- const NativeTextureHandleImpl& handle);
+ void OnTextureFrame(int width, int height, int rotation, int64_t timestamp_ns,
+ const NativeHandleImpl& handle);
void OnOutputFormatRequest(int width, int height, int fps);
protected:
~AndroidVideoCapturerJni();
private:
- void ReturnBuffer(int64_t time_stamp);
JNIEnv* jni();
// To avoid deducing Args from the 3rd parameter of AsyncCapturerInvoke.
@@ -85,10 +88,13 @@
void (webrtc::AndroidVideoCapturer::*method)(Args...),
typename Identity<Args>::type... args);
- const ScopedGlobalRef<jobject> j_capturer_global_;
+ const ScopedGlobalRef<jobject> j_video_capturer_;
const ScopedGlobalRef<jclass> j_video_capturer_class_;
const ScopedGlobalRef<jclass> j_observer_class_;
+ // Used on the Java thread running the camera.
+ webrtc::I420BufferPool buffer_pool_;
+ rtc::scoped_refptr<SurfaceTextureHelper> surface_texture_helper_;
rtc::ThreadChecker thread_checker_;
// |capturer| is a guaranteed to be a valid pointer between a call to
diff --git a/talk/app/webrtc/java/jni/classreferenceholder.cc b/talk/app/webrtc/java/jni/classreferenceholder.cc
index 4c836f8..5fe8ec7 100644
--- a/talk/app/webrtc/java/jni/classreferenceholder.cc
+++ b/talk/app/webrtc/java/jni/classreferenceholder.cc
@@ -72,20 +72,21 @@
LoadClass(jni, "org/webrtc/IceCandidate");
#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
LoadClass(jni, "android/graphics/SurfaceTexture");
- LoadClass(jni, "javax/microedition/khronos/egl/EGLContext");
LoadClass(jni, "org/webrtc/CameraEnumerator");
LoadClass(jni, "org/webrtc/Camera2Enumerator");
LoadClass(jni, "org/webrtc/CameraEnumerationAndroid");
LoadClass(jni, "org/webrtc/VideoCapturerAndroid");
LoadClass(jni, "org/webrtc/VideoCapturerAndroid$NativeObserver");
LoadClass(jni, "org/webrtc/EglBase");
+ LoadClass(jni, "org/webrtc/EglBase$Context");
+ LoadClass(jni, "org/webrtc/EglBase14$Context");
LoadClass(jni, "org/webrtc/NetworkMonitor");
LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder");
LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder$OutputBufferInfo");
LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder$VideoCodecType");
LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder");
LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$DecodedTextureBuffer");
- LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$DecodedByteBuffer");
+ LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$DecodedOutputBuffer");
LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$VideoCodecType");
LoadClass(jni, "org/webrtc/SurfaceTextureHelper");
#endif
diff --git a/talk/app/webrtc/java/jni/jni_helpers.cc b/talk/app/webrtc/java/jni/jni_helpers.cc
index 755698e..3a7ff21 100644
--- a/talk/app/webrtc/java/jni/jni_helpers.cc
+++ b/talk/app/webrtc/java/jni/jni_helpers.cc
@@ -1,4 +1,3 @@
-
/*
* libjingle
* Copyright 2015 Google Inc.
@@ -33,8 +32,6 @@
#include <sys/syscall.h>
#include <unistd.h>
-#include "unicode/unistr.h"
-
namespace webrtc_jni {
static JavaVM* g_jvm = nullptr;
@@ -46,8 +43,6 @@
// were attached by the JVM because of a Java->native call.
static pthread_key_t g_jni_ptr;
-using icu::UnicodeString;
-
JavaVM *GetJVM() {
RTC_CHECK(g_jvm) << "JNI_OnLoad failed to run?";
return g_jvm;
@@ -232,22 +227,20 @@
// Given a UTF-8 encoded |native| string return a new (UTF-16) jstring.
jstring JavaStringFromStdString(JNIEnv* jni, const std::string& native) {
- UnicodeString ustr(UnicodeString::fromUTF8(native));
- jstring jstr = jni->NewString(ustr.getBuffer(), ustr.length());
- CHECK_EXCEPTION(jni) << "error during NewString";
+ jstring jstr = jni->NewStringUTF(native.c_str());
+ CHECK_EXCEPTION(jni) << "error during NewStringUTF";
return jstr;
}
// Given a (UTF-16) jstring return a new UTF-8 native string.
std::string JavaToStdString(JNIEnv* jni, const jstring& j_string) {
- const jchar* jchars = jni->GetStringChars(j_string, NULL);
- CHECK_EXCEPTION(jni) << "Error during GetStringChars";
- UnicodeString ustr(jchars, jni->GetStringLength(j_string));
- CHECK_EXCEPTION(jni) << "Error during GetStringLength";
- jni->ReleaseStringChars(j_string, jchars);
- CHECK_EXCEPTION(jni) << "Error during ReleaseStringChars";
- std::string ret;
- return ustr.toUTF8String(ret);
+ const char* chars = jni->GetStringUTFChars(j_string, NULL);
+ CHECK_EXCEPTION(jni) << "Error during GetStringUTFChars";
+ std::string str(chars, jni->GetStringUTFLength(j_string));
+ CHECK_EXCEPTION(jni) << "Error during GetStringUTFLength";
+ jni->ReleaseStringUTFChars(j_string, chars);
+ CHECK_EXCEPTION(jni) << "Error during ReleaseStringUTFChars";
+ return str;
}
// Return the (singleton) Java Enum object corresponding to |index|;
diff --git a/talk/app/webrtc/java/jni/jni_onload.cc b/talk/app/webrtc/java/jni/jni_onload.cc
new file mode 100644
index 0000000..9664ecd
--- /dev/null
+++ b/talk/app/webrtc/java/jni/jni_onload.cc
@@ -0,0 +1,55 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include <jni.h>
+#undef JNIEXPORT
+#define JNIEXPORT __attribute__((visibility("default")))
+
+#include "talk/app/webrtc/java/jni/classreferenceholder.h"
+#include "talk/app/webrtc/java/jni/jni_helpers.h"
+#include "webrtc/base/ssladapter.h"
+
+namespace webrtc_jni {
+
+extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM *jvm, void *reserved) {
+ jint ret = InitGlobalJniVariables(jvm);
+ RTC_DCHECK_GE(ret, 0);
+ if (ret < 0)
+ return -1;
+
+ RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
+ LoadGlobalClassReferenceHolder();
+
+ return ret;
+}
+
+extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM *jvm, void *reserved) {
+ FreeGlobalClassReferenceHolder();
+ RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
+}
+
+} // namespace webrtc_jni
diff --git a/talk/app/webrtc/java/jni/native_handle_impl.cc b/talk/app/webrtc/java/jni/native_handle_impl.cc
index ac3e045..1757184 100644
--- a/talk/app/webrtc/java/jni/native_handle_impl.cc
+++ b/talk/app/webrtc/java/jni/native_handle_impl.cc
@@ -27,14 +27,65 @@
#include "talk/app/webrtc/java/jni/native_handle_impl.h"
+#include "talk/app/webrtc/java/jni/jni_helpers.h"
+#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/keep_ref_until_done.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/logging.h"
+
+using webrtc::NativeHandleBuffer;
+
+namespace {
+
+void RotateMatrix(float a[16], webrtc::VideoRotation rotation) {
+ // Texture coordinates are in the range 0 to 1. The transformation of the last
+ // row in each rotation matrix is needed for proper translation, e.g, to
+ // mirror x, we don't replace x by -x, but by 1-x.
+ switch (rotation) {
+ case webrtc::kVideoRotation_0:
+ break;
+ case webrtc::kVideoRotation_90: {
+ const float ROTATE_90[16] =
+ { a[4], a[5], a[6], a[7],
+ -a[0], -a[1], -a[2], -a[3],
+ a[8], a[9], a[10], a[11],
+ a[0] + a[12], a[1] + a[13], a[2] + a[14], a[3] + a[15]};
+ memcpy(a, ROTATE_90, sizeof(ROTATE_90));
+ } break;
+ case webrtc::kVideoRotation_180: {
+ const float ROTATE_180[16] =
+ { -a[0], -a[1], -a[2], -a[3],
+ -a[4], -a[5], -a[6], -a[7],
+ a[8], a[9], a[10], a[11],
+ a[0] + a[4] + a[12], a[1] +a[5] + a[13], a[2] + a[6] + a[14],
+ a[3] + a[11]+ a[15]};
+ memcpy(a, ROTATE_180, sizeof(ROTATE_180));
+ }
+ break;
+ case webrtc::kVideoRotation_270: {
+ const float ROTATE_270[16] =
+ { -a[4], -a[5], -a[6], -a[7],
+ a[0], a[1], a[2], a[3],
+ a[8], a[9], a[10], a[11],
+ a[4] + a[12], a[5] + a[13], a[6] + a[14], a[7] + a[15]};
+ memcpy(a, ROTATE_270, sizeof(ROTATE_270));
+ } break;
+ }
+}
+
+} // anonymouse namespace
namespace webrtc_jni {
-NativeTextureHandleImpl::NativeTextureHandleImpl(JNIEnv* jni,
- jint j_oes_texture_id,
- jfloatArray j_transform_matrix)
- : oes_texture_id(j_oes_texture_id) {
+// Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
+static const int kBufferAlignment = 64;
+
+NativeHandleImpl::NativeHandleImpl(JNIEnv* jni,
+ jint j_oes_texture_id,
+ jfloatArray j_transform_matrix)
+ : oes_texture_id(j_oes_texture_id) {
RTC_CHECK_EQ(16, jni->GetArrayLength(j_transform_matrix));
jfloat* transform_matrix_ptr =
jni->GetFloatArrayElements(j_transform_matrix, nullptr);
@@ -44,41 +95,15 @@
jni->ReleaseFloatArrayElements(j_transform_matrix, transform_matrix_ptr, 0);
}
-NativeHandleImpl::NativeHandleImpl() : texture_object_(NULL), texture_id_(-1) {}
-
-void* NativeHandleImpl::GetHandle() {
- return texture_object_;
-}
-
-int NativeHandleImpl::GetTextureId() {
- return texture_id_;
-}
-
-void NativeHandleImpl::SetTextureObject(void* texture_object, int texture_id) {
- texture_object_ = reinterpret_cast<jobject>(texture_object);
- texture_id_ = texture_id;
-}
-
-JniNativeHandleBuffer::JniNativeHandleBuffer(void* native_handle,
- int width,
- int height)
- : NativeHandleBuffer(native_handle, width, height) {}
-
-rtc::scoped_refptr<webrtc::VideoFrameBuffer>
-JniNativeHandleBuffer::NativeToI420Buffer() {
- // TODO(pbos): Implement before using this in the encoder pipeline (or
- // remove the RTC_CHECK() in VideoCapture).
- RTC_NOTREACHED();
- return nullptr;
-}
-
AndroidTextureBuffer::AndroidTextureBuffer(
int width,
int height,
- const NativeTextureHandleImpl& native_handle,
+ const NativeHandleImpl& native_handle,
+ jobject surface_texture_helper,
const rtc::Callback0<void>& no_longer_used)
: webrtc::NativeHandleBuffer(&native_handle_, width, height),
native_handle_(native_handle),
+ surface_texture_helper_(surface_texture_helper),
no_longer_used_cb_(no_longer_used) {}
AndroidTextureBuffer::~AndroidTextureBuffer() {
@@ -87,9 +112,75 @@
rtc::scoped_refptr<webrtc::VideoFrameBuffer>
AndroidTextureBuffer::NativeToI420Buffer() {
- RTC_NOTREACHED()
- << "AndroidTextureBuffer::NativeToI420Buffer not implemented.";
- return nullptr;
+ int uv_width = (width()+7) / 8;
+ int stride = 8 * uv_width;
+ int uv_height = (height()+1)/2;
+ size_t size = stride * (height() + uv_height);
+ // The data is owned by the frame, and the normal case is that the
+ // data is deleted by the frame's destructor callback.
+ //
+ // TODO(nisse): Use an I420BufferPool. We then need to extend that
+ // class, and I420Buffer, to support our memory layout.
+ rtc::scoped_ptr<uint8_t, webrtc::AlignedFreeDeleter> yuv_data(
+ static_cast<uint8_t*>(webrtc::AlignedMalloc(size, kBufferAlignment)));
+ // See SurfaceTextureHelper.java for the required layout.
+ uint8_t* y_data = yuv_data.get();
+ uint8_t* u_data = y_data + height() * stride;
+ uint8_t* v_data = u_data + stride/2;
+
+ rtc::scoped_refptr<webrtc::VideoFrameBuffer> copy =
+ new rtc::RefCountedObject<webrtc::WrappedI420Buffer>(
+ width(), height(),
+ y_data, stride,
+ u_data, stride,
+ v_data, stride,
+ rtc::Bind(&webrtc::AlignedFree, yuv_data.release()));
+
+ JNIEnv* jni = AttachCurrentThreadIfNeeded();
+ ScopedLocalRefFrame local_ref_frame(jni);
+
+ jmethodID transform_mid = GetMethodID(
+ jni,
+ GetObjectClass(jni, surface_texture_helper_),
+ "textureToYUV",
+ "(Ljava/nio/ByteBuffer;IIII[F)V");
+
+ jobject byte_buffer = jni->NewDirectByteBuffer(y_data, size);
+
+ // TODO(nisse): Keep java transform matrix around.
+ jfloatArray sampling_matrix = jni->NewFloatArray(16);
+ jni->SetFloatArrayRegion(sampling_matrix, 0, 16,
+ native_handle_.sampling_matrix);
+
+ jni->CallVoidMethod(surface_texture_helper_,
+ transform_mid,
+ byte_buffer, width(), height(), stride,
+ native_handle_.oes_texture_id, sampling_matrix);
+ CHECK_EXCEPTION(jni) << "textureToYUV throwed an exception";
+
+ return copy;
+}
+
+rtc::scoped_refptr<AndroidTextureBuffer>
+AndroidTextureBuffer::ScaleAndRotate(int dst_widht,
+ int dst_height,
+ webrtc::VideoRotation rotation) {
+ if (width() == dst_widht && height() == dst_height &&
+ rotation == webrtc::kVideoRotation_0) {
+ return this;
+ }
+ int rotated_width = (rotation % 180 == 0) ? dst_widht : dst_height;
+ int rotated_height = (rotation % 180 == 0) ? dst_height : dst_widht;
+
+ // Here we use Bind magic to add a reference count to |this| until the newly
+ // created AndroidTextureBuffer is destructed
+ rtc::scoped_refptr<AndroidTextureBuffer> buffer(
+ new rtc::RefCountedObject<AndroidTextureBuffer>(
+ rotated_width, rotated_height, native_handle_,
+ surface_texture_helper_, rtc::KeepRefUntilDone(this)));
+
+ RotateMatrix(buffer->native_handle_.sampling_matrix, rotation);
+ return buffer;
}
} // namespace webrtc_jni
diff --git a/talk/app/webrtc/java/jni/native_handle_impl.h b/talk/app/webrtc/java/jni/native_handle_impl.h
index dd04bc2..1d0f601 100644
--- a/talk/app/webrtc/java/jni/native_handle_impl.h
+++ b/talk/app/webrtc/java/jni/native_handle_impl.h
@@ -31,56 +31,44 @@
#include <jni.h>
-#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
+#include "webrtc/common_video/rotation.h"
namespace webrtc_jni {
// Wrapper for texture object.
-struct NativeTextureHandleImpl {
- NativeTextureHandleImpl(JNIEnv* jni,
- jint j_oes_texture_id,
- jfloatArray j_transform_matrix);
+struct NativeHandleImpl {
+ NativeHandleImpl(JNIEnv* jni,
+ jint j_oes_texture_id,
+ jfloatArray j_transform_matrix);
const int oes_texture_id;
float sampling_matrix[16];
};
-// Native handle for SurfaceTexture + texture id.
-class NativeHandleImpl {
- public:
- NativeHandleImpl();
-
- void* GetHandle();
- int GetTextureId();
- void SetTextureObject(void* texture_object, int texture_id);
-
- private:
- jobject texture_object_;
- int32_t texture_id_;
-};
-
-class JniNativeHandleBuffer : public webrtc::NativeHandleBuffer {
- public:
- JniNativeHandleBuffer(void* native_handle, int width, int height);
-
- // TODO(pbos): Override destructor to release native handle, at the moment the
- // native handle is not released based on refcount.
-
- private:
- rtc::scoped_refptr<webrtc::VideoFrameBuffer> NativeToI420Buffer() override;
-};
-
class AndroidTextureBuffer : public webrtc::NativeHandleBuffer {
public:
AndroidTextureBuffer(int width,
int height,
- const NativeTextureHandleImpl& native_handle,
+ const NativeHandleImpl& native_handle,
+ jobject surface_texture_helper,
const rtc::Callback0<void>& no_longer_used);
~AndroidTextureBuffer();
rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override;
+ rtc::scoped_refptr<AndroidTextureBuffer> ScaleAndRotate(
+ int dst_widht,
+ int dst_height,
+ webrtc::VideoRotation rotation);
+
private:
- NativeTextureHandleImpl native_handle_;
+ NativeHandleImpl native_handle_;
+ // Raw object pointer, relying on the caller, i.e.,
+ // AndroidVideoCapturerJni or the C++ SurfaceTextureHelper, to keep
+ // a global reference. TODO(nisse): Make this a reference to the C++
+ // SurfaceTextureHelper instead, but that requires some refactoring
+ // of AndroidVideoCapturerJni.
+ jobject surface_texture_helper_;
rtc::Callback0<void> no_longer_used_cb_;
};
diff --git a/talk/app/webrtc/java/jni/peerconnection_jni.cc b/talk/app/webrtc/java/jni/peerconnection_jni.cc
index e75cd55..5ea63f7 100644
--- a/talk/app/webrtc/java/jni/peerconnection_jni.cc
+++ b/talk/app/webrtc/java/jni/peerconnection_jni.cc
@@ -57,6 +57,7 @@
#define JNIEXPORT __attribute__((visibility("default")))
#include <limits>
+#include <utility>
#include "talk/app/webrtc/java/jni/classreferenceholder.h"
#include "talk/app/webrtc/java/jni/jni_helpers.h"
@@ -74,10 +75,11 @@
#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/event_tracer.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/logsinks.h"
-#include "webrtc/base/networkmonitor.h"
#include "webrtc/base/messagequeue.h"
+#include "webrtc/base/networkmonitor.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/system_wrappers/include/field_trial_default.h"
@@ -141,22 +143,6 @@
static bool video_hw_acceleration_enabled = true;
#endif
-extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM *jvm, void *reserved) {
- jint ret = InitGlobalJniVariables(jvm);
- if (ret < 0)
- return -1;
-
- RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
- LoadGlobalClassReferenceHolder();
-
- return ret;
-}
-
-extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM *jvm, void *reserved) {
- FreeGlobalClassReferenceHolder();
- RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
-}
-
// Return the (singleton) Java Enum object corresponding to |index|;
// |state_class_fragment| is something like "MediaSource$State".
static jobject JavaEnumFromIndex(
@@ -545,7 +531,7 @@
protected:
// Common implementation for failure of Set & Create types, distinguished by
// |op| being "Set" or "Create".
- void OnFailure(const std::string& op, const std::string& error) {
+ void DoOnFailure(const std::string& op, const std::string& error) {
jmethodID m = GetMethodID(jni(), *j_observer_class_, "on" + op + "Failure",
"(Ljava/lang/String;)V");
jstring j_error_string = JavaStringFromStdString(jni(), error);
@@ -572,7 +558,7 @@
void OnFailure(const std::string& error) override {
ScopedLocalRefFrame local_ref_frame(jni());
- SdpObserverWrapper::OnFailure(std::string("Create"), error);
+ SdpObserverWrapper::DoOnFailure(std::string("Create"), error);
}
};
@@ -585,7 +571,7 @@
void OnFailure(const std::string& error) override {
ScopedLocalRefFrame local_ref_frame(jni());
- SdpObserverWrapper::OnFailure(std::string("Set"), error);
+ SdpObserverWrapper::DoOnFailure(std::string("Set"), error);
}
};
@@ -773,7 +759,7 @@
jni, *j_frame_class_, "<init>", "(III[I[Ljava/nio/ByteBuffer;J)V")),
j_texture_frame_ctor_id_(GetMethodID(
jni, *j_frame_class_, "<init>",
- "(IIILjava/lang/Object;IJ)V")),
+ "(IIII[FJ)V")),
j_byte_buffer_class_(jni, FindClass(jni, "java/nio/ByteBuffer")) {
CHECK_EXCEPTION(jni);
}
@@ -829,13 +815,13 @@
jobject CricketToJavaTextureFrame(const cricket::VideoFrame* frame) {
NativeHandleImpl* handle =
reinterpret_cast<NativeHandleImpl*>(frame->GetNativeHandle());
- jobject texture_object = reinterpret_cast<jobject>(handle->GetHandle());
- int texture_id = handle->GetTextureId();
+ jfloatArray sampling_matrix = jni()->NewFloatArray(16);
+ jni()->SetFloatArrayRegion(sampling_matrix, 0, 16, handle->sampling_matrix);
return jni()->NewObject(
*j_frame_class_, j_texture_frame_ctor_id_,
frame->GetWidth(), frame->GetHeight(),
static_cast<int>(frame->GetVideoRotation()),
- texture_object, texture_id, javaShallowCopy(frame));
+ handle->oes_texture_id, sampling_matrix, javaShallowCopy(frame));
}
JNIEnv* jni() {
@@ -1054,6 +1040,32 @@
webrtc::field_trial::InitFieldTrialsFromString(field_trials_init_string);
}
+JOW(void, PeerConnectionFactory_initializeInternalTracer)(JNIEnv* jni, jclass) {
+ rtc::tracing::SetupInternalTracer();
+}
+
+JOW(jboolean, PeerConnectionFactory_startInternalTracingCapture)(
+ JNIEnv* jni, jclass, jstring j_event_tracing_filename) {
+ if (!j_event_tracing_filename)
+ return false;
+
+ const char* init_string =
+ jni->GetStringUTFChars(j_event_tracing_filename, NULL);
+ LOG(LS_INFO) << "Starting internal tracing to: " << init_string;
+ bool ret = rtc::tracing::StartInternalCapture(init_string);
+ jni->ReleaseStringUTFChars(j_event_tracing_filename, init_string);
+ return ret;
+}
+
+JOW(void, PeerConnectionFactory_stopInternalTracingCapture)(
+ JNIEnv* jni, jclass) {
+ rtc::tracing::StopInternalCapture();
+}
+
+JOW(void, PeerConnectionFactory_shutdownInternalTracer)(JNIEnv* jni, jclass) {
+ rtc::tracing::ShutdownInternalTracer();
+}
+
// Helper struct for working around the fact that CreatePeerConnectionFactory()
// comes in two flavors: either entirely automagical (constructing its own
// threads and deleting them on teardown, but no external codec factory support)
@@ -1251,6 +1263,46 @@
return (jlong)track.release();
}
+JOW(jboolean, PeerConnectionFactory_nativeStartAecDump)(
+ JNIEnv* jni, jclass, jlong native_factory, jint file) {
+#if defined(ANDROID)
+ rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
+ factoryFromJava(native_factory));
+ return factory->StartAecDump(file);
+#else
+ return false;
+#endif
+}
+
+JOW(void, PeerConnectionFactory_nativeStopAecDump)(
+ JNIEnv* jni, jclass, jlong native_factory) {
+#if defined(ANDROID)
+ rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
+ factoryFromJava(native_factory));
+ factory->StopAecDump();
+#endif
+}
+
+JOW(jboolean, PeerConnectionFactory_nativeStartRtcEventLog)(
+ JNIEnv* jni, jclass, jlong native_factory, jint file) {
+#if defined(ANDROID)
+ rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
+ factoryFromJava(native_factory));
+ return factory->StartRtcEventLog(file);
+#else
+ return false;
+#endif
+}
+
+JOW(void, PeerConnectionFactory_nativeStopRtcEventLog)(
+ JNIEnv* jni, jclass, jlong native_factory) {
+#if defined(ANDROID)
+ rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
+ factoryFromJava(native_factory));
+ factory->StopRtcEventLog();
+#endif
+}
+
JOW(void, PeerConnectionFactory_nativeSetOptions)(
JNIEnv* jni, jclass, jlong native_factory, jobject options) {
rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
@@ -1292,21 +1344,35 @@
}
JOW(void, PeerConnectionFactory_nativeSetVideoHwAccelerationOptions)(
- JNIEnv* jni, jclass, jlong native_factory, jobject render_egl_context) {
+ JNIEnv* jni, jclass, jlong native_factory, jobject local_egl_context,
+ jobject remote_egl_context) {
#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
OwnedFactoryAndThreads* owned_factory =
reinterpret_cast<OwnedFactoryAndThreads*>(native_factory);
+
+ jclass j_eglbase14_context_class =
+ FindClass(jni, "org/webrtc/EglBase14$Context");
+
+ MediaCodecVideoEncoderFactory* encoder_factory =
+ static_cast<MediaCodecVideoEncoderFactory*>
+ (owned_factory->encoder_factory());
+ if (encoder_factory &&
+ jni->IsInstanceOf(local_egl_context, j_eglbase14_context_class)) {
+ LOG(LS_INFO) << "Set EGL context for HW encoding.";
+ encoder_factory->SetEGLContext(jni, local_egl_context);
+ }
+
MediaCodecVideoDecoderFactory* decoder_factory =
static_cast<MediaCodecVideoDecoderFactory*>
(owned_factory->decoder_factory());
- if (decoder_factory) {
- LOG(LS_INFO) << "Set EGL context for HW acceleration.";
- decoder_factory->SetEGLContext(jni, render_egl_context);
+ if (decoder_factory &&
+ jni->IsInstanceOf(remote_egl_context, j_eglbase14_context_class)) {
+ LOG(LS_INFO) << "Set EGL context for HW decoding.";
+ decoder_factory->SetEGLContext(jni, remote_egl_context);
}
#endif
}
-
static std::string
GetJavaEnumName(JNIEnv* jni, const std::string& className, jobject j_enum) {
jclass enumClass = FindClass(jni, className.c_str());
@@ -1503,6 +1569,9 @@
jfieldID j_ice_connection_receiving_timeout_id =
GetFieldID(jni, j_rtc_config_class, "iceConnectionReceivingTimeout", "I");
+ jfieldID j_ice_backup_candidate_pair_ping_interval_id = GetFieldID(
+ jni, j_rtc_config_class, "iceBackupCandidatePairPingInterval", "I");
+
jfieldID j_continual_gathering_policy_id =
GetFieldID(jni, j_rtc_config_class, "continualGatheringPolicy",
"Lorg/webrtc/PeerConnection$ContinualGatheringPolicy;");
@@ -1524,6 +1593,8 @@
jni, j_rtc_config, j_audio_jitter_buffer_fast_accelerate_id);
rtc_config->ice_connection_receiving_timeout =
GetIntField(jni, j_rtc_config, j_ice_connection_receiving_timeout_id);
+ rtc_config->ice_backup_candidate_pair_ping_interval = GetIntField(
+ jni, j_rtc_config, j_ice_backup_candidate_pair_ping_interval_id);
rtc_config->continual_gathering_policy =
JavaContinualGatheringPolicyToNativeType(
jni, j_continual_gathering_policy);
@@ -1550,7 +1621,7 @@
rtc::SSLIdentity::Generate(webrtc::kIdentityName, rtc::KT_ECDSA));
if (ssl_identity.get()) {
rtc_config.certificates.push_back(
- rtc::RTCCertificate::Create(ssl_identity.Pass()));
+ rtc::RTCCertificate::Create(std::move(ssl_identity)));
LOG(LS_INFO) << "ECDSA certificate created.";
} else {
// Failing to create certificate should not abort peer connection
@@ -1704,6 +1775,29 @@
reinterpret_cast<MediaStreamInterface*>(native_stream));
}
+JOW(jobject, PeerConnection_nativeCreateSender)(
+ JNIEnv* jni, jobject j_pc, jstring j_kind, jstring j_stream_id) {
+ jclass j_rtp_sender_class = FindClass(jni, "org/webrtc/RtpSender");
+ jmethodID j_rtp_sender_ctor =
+ GetMethodID(jni, j_rtp_sender_class, "<init>", "(J)V");
+
+ std::string kind = JavaToStdString(jni, j_kind);
+ std::string stream_id = JavaToStdString(jni, j_stream_id);
+ rtc::scoped_refptr<RtpSenderInterface> sender =
+ ExtractNativePC(jni, j_pc)->CreateSender(kind, stream_id);
+ if (!sender.get()) {
+ return nullptr;
+ }
+ jlong nativeSenderPtr = jlongFromPointer(sender.get());
+ jobject j_sender =
+ jni->NewObject(j_rtp_sender_class, j_rtp_sender_ctor, nativeSenderPtr);
+ CHECK_EXCEPTION(jni) << "error during NewObject";
+ // Sender is now owned by the Java object, and will be freed from
+ // RtpSender.dispose(), called by PeerConnection.dispose() or getSenders().
+ sender->AddRef();
+ return j_sender;
+}
+
JOW(jobject, PeerConnection_nativeGetSenders)(JNIEnv* jni, jobject j_pc) {
jclass j_array_list_class = FindClass(jni, "java/util/ArrayList");
jmethodID j_array_list_ctor =
@@ -1723,7 +1817,8 @@
jobject j_sender =
jni->NewObject(j_rtp_sender_class, j_rtp_sender_ctor, nativeSenderPtr);
CHECK_EXCEPTION(jni) << "error during NewObject";
- // Sender is now owned by Java object, and will be freed from there.
+ // Sender is now owned by the Java object, and will be freed from
+ // RtpSender.dispose(), called by PeerConnection.dispose() or getSenders().
sender->AddRef();
jni->CallBooleanMethod(j_senders, j_array_list_add, j_sender);
CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
@@ -1802,6 +1897,7 @@
// Since we can't create platform specific java implementations in Java, we
// defer the creation to C land.
#if defined(ANDROID)
+ // TODO(nisse): This case is intended to be deleted.
jclass j_video_capturer_class(
FindClass(jni, "org/webrtc/VideoCapturerAndroid"));
const int camera_id = jni->CallStaticIntMethod(
@@ -1816,8 +1912,13 @@
j_video_capturer_class,
GetMethodID(jni, j_video_capturer_class, "<init>", "(I)V"), camera_id);
CHECK_EXCEPTION(jni) << "error during creation of VideoCapturerAndroid";
+ jfieldID helper_fid = GetFieldID(jni, j_video_capturer_class, "surfaceHelper",
+ "Lorg/webrtc/SurfaceTextureHelper;");
+
rtc::scoped_refptr<webrtc::AndroidVideoCapturerDelegate> delegate =
- new rtc::RefCountedObject<AndroidVideoCapturerJni>(jni, j_video_capturer);
+ new rtc::RefCountedObject<AndroidVideoCapturerJni>(
+ jni, j_video_capturer,
+ GetObjectField(jni, j_video_capturer, helper_fid));
rtc::scoped_ptr<cricket::VideoCapturer> capturer(
new webrtc::AndroidVideoCapturer(delegate));
@@ -2003,11 +2104,11 @@
return result;
}
-JOW(void, RtpSender_nativeSetTrack)(JNIEnv* jni,
+JOW(jboolean, RtpSender_nativeSetTrack)(JNIEnv* jni,
jclass,
jlong j_rtp_sender_pointer,
jlong j_track_pointer) {
- reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
+ return reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
->SetTrack(reinterpret_cast<MediaStreamTrackInterface*>(j_track_pointer));
}
diff --git a/talk/app/webrtc/java/jni/surfacetexturehelper_jni.cc b/talk/app/webrtc/java/jni/surfacetexturehelper_jni.cc
index 05f1b23..3e32b9a 100644
--- a/talk/app/webrtc/java/jni/surfacetexturehelper_jni.cc
+++ b/talk/app/webrtc/java/jni/surfacetexturehelper_jni.cc
@@ -35,25 +35,14 @@
namespace webrtc_jni {
-SurfaceTextureHelper::SurfaceTextureHelper(JNIEnv* jni,
- jobject egl_shared_context)
- : j_surface_texture_helper_class_(
- jni,
- FindClass(jni, "org/webrtc/SurfaceTextureHelper")),
- j_surface_texture_helper_(
- jni,
- jni->CallStaticObjectMethod(
- *j_surface_texture_helper_class_,
- GetStaticMethodID(jni,
- *j_surface_texture_helper_class_,
- "create",
- "(Ljavax/microedition/khronos/egl/EGLContext;)"
- "Lorg/webrtc/SurfaceTextureHelper;"),
- egl_shared_context)),
- j_return_texture_method_(GetMethodID(jni,
- *j_surface_texture_helper_class_,
- "returnTextureFrame",
- "()V")) {
+SurfaceTextureHelper::SurfaceTextureHelper(
+ JNIEnv* jni, jobject surface_texture_helper)
+ : j_surface_texture_helper_(jni, surface_texture_helper),
+ j_return_texture_method_(
+ GetMethodID(jni,
+ FindClass(jni, "org/webrtc/SurfaceTextureHelper"),
+ "returnTextureFrame",
+ "()V")) {
CHECK_EXCEPTION(jni) << "error during initialization of SurfaceTextureHelper";
}
@@ -70,9 +59,9 @@
rtc::scoped_refptr<webrtc::VideoFrameBuffer>
SurfaceTextureHelper::CreateTextureFrame(int width, int height,
- const NativeTextureHandleImpl& native_handle) {
+ const NativeHandleImpl& native_handle) {
return new rtc::RefCountedObject<AndroidTextureBuffer>(
- width, height, native_handle,
+ width, height, native_handle, *j_surface_texture_helper_,
rtc::Bind(&SurfaceTextureHelper::ReturnTextureFrame, this));
}
diff --git a/talk/app/webrtc/java/jni/surfacetexturehelper_jni.h b/talk/app/webrtc/java/jni/surfacetexturehelper_jni.h
index dc9d2b8..8dde2b5 100644
--- a/talk/app/webrtc/java/jni/surfacetexturehelper_jni.h
+++ b/talk/app/webrtc/java/jni/surfacetexturehelper_jni.h
@@ -35,7 +35,7 @@
#include "talk/app/webrtc/java/jni/native_handle_impl.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
namespace webrtc_jni {
@@ -49,24 +49,19 @@
// destroyed while a VideoFrameBuffer is in use.
// This class is the C++ counterpart of the java class SurfaceTextureHelper.
// Usage:
-// 1. Create an instance of this class.
-// 2. Call GetJavaSurfaceTextureHelper to get the Java SurfaceTextureHelper.
+// 1. Create an java instance of SurfaceTextureHelper.
+// 2. Create an instance of this class.
// 3. Register a listener to the Java SurfaceListener and start producing
// new buffers.
-// 3. Call CreateTextureFrame to wrap the Java texture in a VideoFrameBuffer.
+// 4. Call CreateTextureFrame to wrap the Java texture in a VideoFrameBuffer.
class SurfaceTextureHelper : public rtc::RefCountInterface {
public:
- SurfaceTextureHelper(JNIEnv* jni, jobject shared_egl_context);
-
- // Returns the Java SurfaceTextureHelper.
- jobject GetJavaSurfaceTextureHelper() const {
- return *j_surface_texture_helper_;
- }
+ SurfaceTextureHelper(JNIEnv* jni, jobject surface_texture_helper);
rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateTextureFrame(
int width,
int height,
- const NativeTextureHandleImpl& native_handle);
+ const NativeHandleImpl& native_handle);
protected:
~SurfaceTextureHelper();
@@ -75,7 +70,6 @@
// May be called on arbitrary thread.
void ReturnTextureFrame() const;
- const ScopedGlobalRef<jclass> j_surface_texture_helper_class_;
const ScopedGlobalRef<jobject> j_surface_texture_helper_;
const jmethodID j_return_texture_method_;
};
diff --git a/talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java b/talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
index 42af9c7..19002f7 100644
--- a/talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
+++ b/talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java
@@ -33,23 +33,23 @@
import android.media.MediaCodecInfo.CodecCapabilities;
import android.media.MediaCodecList;
import android.media.MediaFormat;
-import android.opengl.GLES11Ext;
-import android.opengl.GLES20;
import android.os.Build;
+import android.os.SystemClock;
import android.view.Surface;
import org.webrtc.Logging;
import java.nio.ByteBuffer;
import java.util.Arrays;
+import java.util.LinkedList;
import java.util.List;
-
-import javax.microedition.khronos.egl.EGLContext;
+import java.util.concurrent.CountDownLatch;
+import java.util.Queue;
+import java.util.concurrent.TimeUnit;
// Java-side of peerconnection_jni.cc:MediaCodecVideoDecoder.
// This class is an implementation detail of the Java PeerConnection API.
-// MediaCodec is thread-hostile so this class must be operated on a single
-// thread.
+@SuppressWarnings("deprecation")
public class MediaCodecVideoDecoder {
// This class is constructed, operated, and destroyed by its C++ incarnation,
// so the class and its methods have non-public visibility. The API this
@@ -66,18 +66,26 @@
}
private static final int DEQUEUE_INPUT_TIMEOUT = 500000; // 500 ms timeout.
+ private static final int MEDIA_CODEC_RELEASE_TIMEOUT_MS = 5000; // Timeout for codec releasing.
// Active running decoder instance. Set in initDecode() (called from native code)
// and reset to null in release() call.
private static MediaCodecVideoDecoder runningInstance = null;
+ private static MediaCodecVideoDecoderErrorCallback errorCallback = null;
+ private static int codecErrors = 0;
+
private Thread mediaCodecThread;
private MediaCodec mediaCodec;
private ByteBuffer[] inputBuffers;
private ByteBuffer[] outputBuffers;
private static final String VP8_MIME_TYPE = "video/x-vnd.on2.vp8";
+ private static final String VP9_MIME_TYPE = "video/x-vnd.on2.vp9";
private static final String H264_MIME_TYPE = "video/avc";
// List of supported HW VP8 decoders.
private static final String[] supportedVp8HwCodecPrefixes =
{"OMX.qcom.", "OMX.Nvidia.", "OMX.Exynos.", "OMX.Intel." };
+ // List of supported HW VP9 decoders.
+ private static final String[] supportedVp9HwCodecPrefixes =
+ {"OMX.qcom.", "OMX.Exynos." };
// List of supported HW H.264 decoders.
private static final String[] supportedH264HwCodecPrefixes =
{"OMX.qcom.", "OMX.Intel." };
@@ -96,13 +104,29 @@
private int height;
private int stride;
private int sliceHeight;
+ private boolean hasDecodedFirstFrame;
+ private final Queue<TimeStamps> decodeStartTimeMs = new LinkedList<TimeStamps>();
private boolean useSurface;
- private int textureID = 0;
- private SurfaceTexture surfaceTexture = null;
- private Surface surface = null;
- private EglBase eglBase;
- private MediaCodecVideoDecoder() {
+ // The below variables are only used when decoding to a Surface.
+ private TextureListener textureListener;
+ // Max number of output buffers queued before starting to drop decoded frames.
+ private static final int MAX_QUEUED_OUTPUTBUFFERS = 3;
+ private int droppedFrames;
+ private Surface surface = null;
+ private final Queue<DecodedOutputBuffer>
+ dequeuedSurfaceOutputBuffers = new LinkedList<DecodedOutputBuffer>();
+
+ // MediaCodec error handler - invoked when critical error happens which may prevent
+ // further use of media codec API. Now it means that one of media codec instances
+ // is hanging and can no longer be used in the next call.
+ public static interface MediaCodecVideoDecoderErrorCallback {
+ void onMediaCodecVideoDecoderCriticalError(int codecErrors);
+ }
+
+ public static void setErrorCallback(MediaCodecVideoDecoderErrorCallback errorCallback) {
+ Logging.d(TAG, "Set error callback");
+ MediaCodecVideoDecoder.errorCallback = errorCallback;
}
// Helper struct for findVp8Decoder() below.
@@ -120,6 +144,7 @@
if (Build.VERSION.SDK_INT < Build.VERSION_CODES.KITKAT) {
return null; // MediaCodec.setParameters is missing.
}
+ Logging.d(TAG, "Trying to find HW decoder for mime " + mime);
for (int i = 0; i < MediaCodecList.getCodecCount(); ++i) {
MediaCodecInfo info = MediaCodecList.getCodecInfoAt(i);
if (info.isEncoder()) {
@@ -135,7 +160,7 @@
if (name == null) {
continue; // No HW support in this codec; try the next one.
}
- Logging.v(TAG, "Found candidate decoder " + name);
+ Logging.d(TAG, "Found candidate decoder " + name);
// Check if this is supported decoder.
boolean supportedCodec = false;
@@ -166,6 +191,7 @@
}
}
}
+ Logging.d(TAG, "No HW decoder found for mime " + mime);
return null; // No HW decoder.
}
@@ -173,6 +199,10 @@
return findDecoder(VP8_MIME_TYPE, supportedVp8HwCodecPrefixes) != null;
}
+ public static boolean isVp9HwSupported() {
+ return findDecoder(VP9_MIME_TYPE, supportedVp9HwCodecPrefixes) != null;
+ }
+
public static boolean isH264HwSupported() {
return findDecoder(H264_MIME_TYPE, supportedH264HwCodecPrefixes) != null;
}
@@ -197,17 +227,21 @@
}
}
- // Pass null in |sharedContext| to configure the codec for ByteBuffer output.
- private boolean initDecode(VideoCodecType type, int width, int height, EGLContext sharedContext) {
+ // Pass null in |surfaceTextureHelper| to configure the codec for ByteBuffer output.
+ private boolean initDecode(
+ VideoCodecType type, int width, int height, SurfaceTextureHelper surfaceTextureHelper) {
if (mediaCodecThread != null) {
throw new RuntimeException("Forgot to release()?");
}
- useSurface = (sharedContext != null);
+ useSurface = (surfaceTextureHelper != null);
String mime = null;
String[] supportedCodecPrefixes = null;
if (type == VideoCodecType.VIDEO_CODEC_VP8) {
mime = VP8_MIME_TYPE;
supportedCodecPrefixes = supportedVp8HwCodecPrefixes;
+ } else if (type == VideoCodecType.VIDEO_CODEC_VP9) {
+ mime = VP9_MIME_TYPE;
+ supportedCodecPrefixes = supportedVp9HwCodecPrefixes;
} else if (type == VideoCodecType.VIDEO_CODEC_H264) {
mime = H264_MIME_TYPE;
supportedCodecPrefixes = supportedH264HwCodecPrefixes;
@@ -221,9 +255,6 @@
Logging.d(TAG, "Java initDecode: " + type + " : "+ width + " x " + height +
". Color: 0x" + Integer.toHexString(properties.colorFormat) +
". Use Surface: " + useSurface);
- if (sharedContext != null) {
- Logging.d(TAG, "Decoder shared EGL Context: " + sharedContext);
- }
runningInstance = this; // Decoder is now running and can be queried for stack traces.
mediaCodecThread = Thread.currentThread();
try {
@@ -233,16 +264,8 @@
sliceHeight = height;
if (useSurface) {
- // Create shared EGL context.
- eglBase = new EglBase(sharedContext, EglBase.ConfigType.PIXEL_BUFFER);
- eglBase.createDummyPbufferSurface();
- eglBase.makeCurrent();
-
- // Create output surface
- textureID = GlUtil.generateTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES);
- Logging.d(TAG, "Video decoder TextureID = " + textureID);
- surfaceTexture = new SurfaceTexture(textureID);
- surface = new Surface(surfaceTexture);
+ textureListener = new TextureListener(surfaceTextureHelper);
+ surface = new Surface(surfaceTextureHelper.getSurfaceTexture());
}
MediaFormat format = MediaFormat.createVideoFormat(mime, width, height);
@@ -261,6 +284,10 @@
colorFormat = properties.colorFormat;
outputBuffers = mediaCodec.getOutputBuffers();
inputBuffers = mediaCodec.getInputBuffers();
+ decodeStartTimeMs.clear();
+ hasDecodedFirstFrame = false;
+ dequeuedSurfaceOutputBuffers.clear();
+ droppedFrames = 0;
Logging.d(TAG, "Input buffers: " + inputBuffers.length +
". Output buffers: " + outputBuffers.length);
return true;
@@ -271,25 +298,45 @@
}
private void release() {
- Logging.d(TAG, "Java releaseDecoder");
+ Logging.d(TAG, "Java releaseDecoder. Total number of dropped frames: " + droppedFrames);
checkOnMediaCodecThread();
- try {
- mediaCodec.stop();
- mediaCodec.release();
- } catch (IllegalStateException e) {
- Logging.e(TAG, "release failed", e);
+
+ // Run Mediacodec stop() and release() on separate thread since sometime
+ // Mediacodec.stop() may hang.
+ final CountDownLatch releaseDone = new CountDownLatch(1);
+
+ Runnable runMediaCodecRelease = new Runnable() {
+ @Override
+ public void run() {
+ try {
+ Logging.d(TAG, "Java releaseDecoder on release thread");
+ mediaCodec.stop();
+ mediaCodec.release();
+ Logging.d(TAG, "Java releaseDecoder on release thread done");
+ } catch (Exception e) {
+ Logging.e(TAG, "Media decoder release failed", e);
+ }
+ releaseDone.countDown();
+ }
+ };
+ new Thread(runMediaCodecRelease).start();
+
+ if (!ThreadUtils.awaitUninterruptibly(releaseDone, MEDIA_CODEC_RELEASE_TIMEOUT_MS)) {
+ Logging.e(TAG, "Media decoder release timeout");
+ codecErrors++;
+ if (errorCallback != null) {
+ Logging.e(TAG, "Invoke codec error callback. Errors: " + codecErrors);
+ errorCallback.onMediaCodecVideoDecoderCriticalError(codecErrors);
+ }
}
+
mediaCodec = null;
mediaCodecThread = null;
runningInstance = null;
if (useSurface) {
surface.release();
surface = null;
- Logging.d(TAG, "Delete video decoder TextureID " + textureID);
- GLES20.glDeleteTextures(1, new int[] {textureID}, 0);
- textureID = 0;
- eglBase.release();
- eglBase = null;
+ textureListener.release();
}
Logging.d(TAG, "Java releaseDecoder done");
}
@@ -306,13 +353,15 @@
}
}
- private boolean queueInputBuffer(
- int inputBufferIndex, int size, long timestampUs) {
+ private boolean queueInputBuffer(int inputBufferIndex, int size, long presentationTimeStamUs,
+ long timeStampMs, long ntpTimeStamp) {
checkOnMediaCodecThread();
try {
inputBuffers[inputBufferIndex].position(0);
inputBuffers[inputBufferIndex].limit(size);
- mediaCodec.queueInputBuffer(inputBufferIndex, 0, size, timestampUs, 0);
+ decodeStartTimeMs.add(new TimeStamps(SystemClock.elapsedRealtime(), timeStampMs,
+ ntpTimeStamp));
+ mediaCodec.queueInputBuffer(inputBufferIndex, 0, size, presentationTimeStamUs, 0);
return true;
}
catch (IllegalStateException e) {
@@ -321,56 +370,183 @@
}
}
- // Helper structs for dequeueOutputBuffer() below.
- private static class DecodedByteBuffer {
- public DecodedByteBuffer(int index, int offset, int size, long presentationTimestampUs) {
+ private static class TimeStamps {
+ public TimeStamps(long decodeStartTimeMs, long timeStampMs, long ntpTimeStampMs) {
+ this.decodeStartTimeMs = decodeStartTimeMs;
+ this.timeStampMs = timeStampMs;
+ this.ntpTimeStampMs = ntpTimeStampMs;
+ }
+ private final long decodeStartTimeMs; // Time when this frame was queued for decoding.
+ private final long timeStampMs; // Only used for bookkeeping in Java. Used in C++;
+ private final long ntpTimeStampMs; // Only used for bookkeeping in Java. Used in C++;
+ }
+
+ // Helper struct for dequeueOutputBuffer() below.
+ private static class DecodedOutputBuffer {
+ public DecodedOutputBuffer(int index, int offset, int size, long timeStampMs,
+ long ntpTimeStampMs, long decodeTime, long endDecodeTime) {
this.index = index;
this.offset = offset;
this.size = size;
- this.presentationTimestampUs = presentationTimestampUs;
+ this.timeStampMs = timeStampMs;
+ this.ntpTimeStampMs = ntpTimeStampMs;
+ this.decodeTimeMs = decodeTime;
+ this.endDecodeTimeMs = endDecodeTime;
}
private final int index;
private final int offset;
private final int size;
- private final long presentationTimestampUs;
+ private final long timeStampMs;
+ private final long ntpTimeStampMs;
+ // Number of ms it took to decode this frame.
+ private final long decodeTimeMs;
+ // System time when this frame finished decoding.
+ private final long endDecodeTimeMs;
}
+ // Helper struct for dequeueTextureBuffer() below.
private static class DecodedTextureBuffer {
private final int textureID;
- private final long presentationTimestampUs;
+ private final float[] transformMatrix;
+ private final long timeStampMs;
+ private final long ntpTimeStampMs;
+ private final long decodeTimeMs;
+ // Interval from when the frame finished decoding until this buffer has been created.
+ // Since there is only one texture, this interval depend on the time from when
+ // a frame is decoded and provided to C++ and until that frame is returned to the MediaCodec
+ // so that the texture can be updated with the next decoded frame.
+ private final long frameDelayMs;
- public DecodedTextureBuffer(int textureID, long presentationTimestampUs) {
+ // A DecodedTextureBuffer with zero |textureID| has special meaning and represents a frame
+ // that was dropped.
+ public DecodedTextureBuffer(int textureID, float[] transformMatrix, long timeStampMs,
+ long ntpTimeStampMs, long decodeTimeMs, long frameDelay) {
this.textureID = textureID;
- this.presentationTimestampUs = presentationTimestampUs;
+ this.transformMatrix = transformMatrix;
+ this.timeStampMs = timeStampMs;
+ this.ntpTimeStampMs = ntpTimeStampMs;
+ this.decodeTimeMs = decodeTimeMs;
+ this.frameDelayMs = frameDelay;
}
}
- // Returns null if no decoded buffer is available, and otherwise either a DecodedByteBuffer or
- // DecodedTexturebuffer depending on |useSurface| configuration.
+ // Poll based texture listener.
+ private static class TextureListener
+ implements SurfaceTextureHelper.OnTextureFrameAvailableListener {
+ private final SurfaceTextureHelper surfaceTextureHelper;
+ // |newFrameLock| is used to synchronize arrival of new frames with wait()/notifyAll().
+ private final Object newFrameLock = new Object();
+ // |bufferToRender| is non-null when waiting for transition between addBufferToRender() to
+ // onTextureFrameAvailable().
+ private DecodedOutputBuffer bufferToRender;
+ private DecodedTextureBuffer renderedBuffer;
+
+ public TextureListener(SurfaceTextureHelper surfaceTextureHelper) {
+ this.surfaceTextureHelper = surfaceTextureHelper;
+ surfaceTextureHelper.setListener(this);
+ }
+
+ public void addBufferToRender(DecodedOutputBuffer buffer) {
+ if (bufferToRender != null) {
+ Logging.e(TAG,
+ "Unexpected addBufferToRender() called while waiting for a texture.");
+ throw new IllegalStateException("Waiting for a texture.");
+ }
+ bufferToRender = buffer;
+ }
+
+ public boolean isWaitingForTexture() {
+ synchronized (newFrameLock) {
+ return bufferToRender != null;
+ }
+ }
+
+ // Callback from |surfaceTextureHelper|. May be called on an arbitrary thread.
+ @Override
+ public void onTextureFrameAvailable(
+ int oesTextureId, float[] transformMatrix, long timestampNs) {
+ synchronized (newFrameLock) {
+ if (renderedBuffer != null) {
+ Logging.e(TAG,
+ "Unexpected onTextureFrameAvailable() called while already holding a texture.");
+ throw new IllegalStateException("Already holding a texture.");
+ }
+ // |timestampNs| is always zero on some Android versions.
+ renderedBuffer = new DecodedTextureBuffer(oesTextureId, transformMatrix,
+ bufferToRender.timeStampMs, bufferToRender.ntpTimeStampMs, bufferToRender.decodeTimeMs,
+ SystemClock.elapsedRealtime() - bufferToRender.endDecodeTimeMs);
+ bufferToRender = null;
+ newFrameLock.notifyAll();
+ }
+ }
+
+ // Dequeues and returns a DecodedTextureBuffer if available, or null otherwise.
+ public DecodedTextureBuffer dequeueTextureBuffer(int timeoutMs) {
+ synchronized (newFrameLock) {
+ if (renderedBuffer == null && timeoutMs > 0 && isWaitingForTexture()) {
+ try {
+ newFrameLock.wait(timeoutMs);
+ } catch(InterruptedException e) {
+ // Restore the interrupted status by reinterrupting the thread.
+ Thread.currentThread().interrupt();
+ }
+ }
+ DecodedTextureBuffer returnedBuffer = renderedBuffer;
+ renderedBuffer = null;
+ return returnedBuffer;
+ }
+ }
+
+ public void release() {
+ // SurfaceTextureHelper.disconnect() will block until any onTextureFrameAvailable() in
+ // progress is done. Therefore, the call to disconnect() must be outside any synchronized
+ // statement that is also used in the onTextureFrameAvailable() above to avoid deadlocks.
+ surfaceTextureHelper.disconnect();
+ synchronized (newFrameLock) {
+ if (renderedBuffer != null) {
+ surfaceTextureHelper.returnTextureFrame();
+ renderedBuffer = null;
+ }
+ }
+ }
+ }
+
+ // Returns null if no decoded buffer is available, and otherwise a DecodedByteBuffer.
// Throws IllegalStateException if call is made on the wrong thread, if color format changes to an
// unsupported format, or if |mediaCodec| is not in the Executing state. Throws CodecException
// upon codec error.
- private Object dequeueOutputBuffer(int dequeueTimeoutUs)
- throws IllegalStateException, MediaCodec.CodecException {
+ private DecodedOutputBuffer dequeueOutputBuffer(int dequeueTimeoutMs) {
checkOnMediaCodecThread();
+ if (decodeStartTimeMs.isEmpty()) {
+ return null;
+ }
// Drain the decoder until receiving a decoded buffer or hitting
// MediaCodec.INFO_TRY_AGAIN_LATER.
final MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
while (true) {
- final int result = mediaCodec.dequeueOutputBuffer(info, dequeueTimeoutUs);
+ final int result = mediaCodec.dequeueOutputBuffer(
+ info, TimeUnit.MILLISECONDS.toMicros(dequeueTimeoutMs));
switch (result) {
- case MediaCodec.INFO_TRY_AGAIN_LATER:
- return null;
case MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED:
outputBuffers = mediaCodec.getOutputBuffers();
Logging.d(TAG, "Decoder output buffers changed: " + outputBuffers.length);
+ if (hasDecodedFirstFrame) {
+ throw new RuntimeException("Unexpected output buffer change event.");
+ }
break;
case MediaCodec.INFO_OUTPUT_FORMAT_CHANGED:
MediaFormat format = mediaCodec.getOutputFormat();
Logging.d(TAG, "Decoder format changed: " + format.toString());
+ int new_width = format.getInteger(MediaFormat.KEY_WIDTH);
+ int new_height = format.getInteger(MediaFormat.KEY_HEIGHT);
+ if (hasDecodedFirstFrame && (new_width != width || new_height != height)) {
+ throw new RuntimeException("Unexpected size change. Configured " + width + "*" +
+ height + ". New " + new_width + "*" + new_height);
+ }
width = format.getInteger(MediaFormat.KEY_WIDTH);
height = format.getInteger(MediaFormat.KEY_HEIGHT);
+
if (!useSurface && format.containsKey(MediaFormat.KEY_COLOR_FORMAT)) {
colorFormat = format.getInteger(MediaFormat.KEY_COLOR_FORMAT);
Logging.d(TAG, "Color: 0x" + Integer.toHexString(colorFormat));
@@ -388,30 +564,88 @@
stride = Math.max(width, stride);
sliceHeight = Math.max(height, sliceHeight);
break;
+ case MediaCodec.INFO_TRY_AGAIN_LATER:
+ return null;
default:
- // Output buffer decoded.
- if (useSurface) {
- mediaCodec.releaseOutputBuffer(result, true /* render */);
- // TODO(magjed): Wait for SurfaceTexture.onFrameAvailable() before returning a texture
- // frame.
- return new DecodedTextureBuffer(textureID, info.presentationTimeUs);
- } else {
- return new DecodedByteBuffer(result, info.offset, info.size, info.presentationTimeUs);
- }
- }
+ hasDecodedFirstFrame = true;
+ TimeStamps timeStamps = decodeStartTimeMs.remove();
+ return new DecodedOutputBuffer(result, info.offset, info.size, timeStamps.timeStampMs,
+ timeStamps.ntpTimeStampMs,
+ SystemClock.elapsedRealtime() - timeStamps.decodeStartTimeMs,
+ SystemClock.elapsedRealtime());
+ }
}
}
+ // Returns null if no decoded buffer is available, and otherwise a DecodedTextureBuffer.
+ // Throws IllegalStateException if call is made on the wrong thread, if color format changes to an
+ // unsupported format, or if |mediaCodec| is not in the Executing state. Throws CodecException
+ // upon codec error. If |dequeueTimeoutMs| > 0, the oldest decoded frame will be dropped if
+ // a frame can't be returned.
+ private DecodedTextureBuffer dequeueTextureBuffer(int dequeueTimeoutMs) {
+ checkOnMediaCodecThread();
+ if (!useSurface) {
+ throw new IllegalStateException("dequeueTexture() called for byte buffer decoding.");
+ }
+ DecodedOutputBuffer outputBuffer = dequeueOutputBuffer(dequeueTimeoutMs);
+ if (outputBuffer != null) {
+ dequeuedSurfaceOutputBuffers.add(outputBuffer);
+ }
+
+ MaybeRenderDecodedTextureBuffer();
+ // Check if there is texture ready now by waiting max |dequeueTimeoutMs|.
+ DecodedTextureBuffer renderedBuffer = textureListener.dequeueTextureBuffer(dequeueTimeoutMs);
+ if (renderedBuffer != null) {
+ MaybeRenderDecodedTextureBuffer();
+ return renderedBuffer;
+ }
+
+ if ((dequeuedSurfaceOutputBuffers.size()
+ >= Math.min(MAX_QUEUED_OUTPUTBUFFERS, outputBuffers.length)
+ || (dequeueTimeoutMs > 0 && !dequeuedSurfaceOutputBuffers.isEmpty()))) {
+ ++droppedFrames;
+ // Drop the oldest frame still in dequeuedSurfaceOutputBuffers.
+ // The oldest frame is owned by |textureListener| and can't be dropped since
+ // mediaCodec.releaseOutputBuffer has already been called.
+ final DecodedOutputBuffer droppedFrame = dequeuedSurfaceOutputBuffers.remove();
+ if (dequeueTimeoutMs > 0) {
+ // TODO(perkj): Re-add the below log when VideoRenderGUI has been removed or fixed to
+ // return the one and only texture even if it does not render.
+ // Logging.w(TAG, "Draining decoder. Dropping frame with TS: "
+ // + droppedFrame.timeStampMs + ". Total number of dropped frames: " + droppedFrames);
+ } else {
+ Logging.w(TAG, "Too many output buffers. Dropping frame with TS: "
+ + droppedFrame.timeStampMs + ". Total number of dropped frames: " + droppedFrames);
+ }
+
+ mediaCodec.releaseOutputBuffer(droppedFrame.index, false /* render */);
+ return new DecodedTextureBuffer(0, null, droppedFrame.timeStampMs,
+ droppedFrame.ntpTimeStampMs, droppedFrame.decodeTimeMs,
+ SystemClock.elapsedRealtime() - droppedFrame.endDecodeTimeMs);
+ }
+ return null;
+ }
+
+ private void MaybeRenderDecodedTextureBuffer() {
+ if (dequeuedSurfaceOutputBuffers.isEmpty() || textureListener.isWaitingForTexture()) {
+ return;
+ }
+ // Get the first frame in the queue and render to the decoder output surface.
+ final DecodedOutputBuffer buffer = dequeuedSurfaceOutputBuffers.remove();
+ textureListener.addBufferToRender(buffer);
+ mediaCodec.releaseOutputBuffer(buffer.index, true /* render */);
+ }
+
// Release a dequeued output byte buffer back to the codec for re-use. Should only be called for
// non-surface decoding.
// Throws IllegalStateException if the call is made on the wrong thread, if codec is configured
// for surface decoding, or if |mediaCodec| is not in the Executing state. Throws
// MediaCodec.CodecException upon codec error.
- private void returnDecodedByteBuffer(int index)
+ private void returnDecodedOutputBuffer(int index)
throws IllegalStateException, MediaCodec.CodecException {
checkOnMediaCodecThread();
if (useSurface) {
- throw new IllegalStateException("returnDecodedByteBuffer() called for surface decoding.");
+ throw new IllegalStateException("returnDecodedOutputBuffer() called for surface decoding.");
}
mediaCodec.releaseOutputBuffer(index, false /* render */);
}
diff --git a/talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java b/talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
index f3f03c1..5c8f9dc 100644
--- a/talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
+++ b/talk/app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java
@@ -27,24 +27,29 @@
package org.webrtc;
+import android.annotation.TargetApi;
import android.media.MediaCodec;
import android.media.MediaCodecInfo.CodecCapabilities;
import android.media.MediaCodecInfo;
import android.media.MediaCodecList;
import android.media.MediaFormat;
+import android.opengl.GLES20;
import android.os.Build;
import android.os.Bundle;
+import android.view.Surface;
import org.webrtc.Logging;
import java.nio.ByteBuffer;
import java.util.Arrays;
import java.util.List;
+import java.util.concurrent.CountDownLatch;
+import java.util.concurrent.TimeUnit;
// Java-side of peerconnection_jni.cc:MediaCodecVideoEncoder.
// This class is an implementation detail of the Java PeerConnection API.
-// MediaCodec is thread-hostile so this class must be operated on a single
-// thread.
+@TargetApi(19)
+@SuppressWarnings("deprecation")
public class MediaCodecVideoEncoder {
// This class is constructed, operated, and destroyed by its C++ incarnation,
// so the class and its methods have non-public visibility. The API this
@@ -60,18 +65,31 @@
VIDEO_CODEC_H264
}
+ private static final int MEDIA_CODEC_RELEASE_TIMEOUT_MS = 5000; // Timeout for codec releasing.
private static final int DEQUEUE_TIMEOUT = 0; // Non-blocking, no wait.
- // Active running encoder instance. Set in initDecode() (called from native code)
+ // Active running encoder instance. Set in initEncode() (called from native code)
// and reset to null in release() call.
private static MediaCodecVideoEncoder runningInstance = null;
+ private static MediaCodecVideoEncoderErrorCallback errorCallback = null;
+ private static int codecErrors = 0;
+
private Thread mediaCodecThread;
private MediaCodec mediaCodec;
private ByteBuffer[] outputBuffers;
+ private EglBase14 eglBase;
+ private int width;
+ private int height;
+ private Surface inputSurface;
+ private GlRectDrawer drawer;
private static final String VP8_MIME_TYPE = "video/x-vnd.on2.vp8";
+ private static final String VP9_MIME_TYPE = "video/x-vnd.on2.vp9";
private static final String H264_MIME_TYPE = "video/avc";
// List of supported HW VP8 codecs.
private static final String[] supportedVp8HwCodecPrefixes =
{"OMX.qcom.", "OMX.Intel." };
+ // List of supported HW VP9 decoders.
+ private static final String[] supportedVp9HwCodecPrefixes =
+ {"OMX.qcom."};
// List of supported HW H.264 codecs.
private static final String[] supportedH264HwCodecPrefixes =
{"OMX.qcom." };
@@ -99,13 +117,25 @@
CodecCapabilities.COLOR_QCOM_FormatYUV420SemiPlanar,
COLOR_QCOM_FORMATYUV420PackedSemiPlanar32m
};
- private int colorFormat;
- // Video encoder type.
+ private static final int[] supportedSurfaceColorList = {
+ CodecCapabilities.COLOR_FormatSurface
+ };
private VideoCodecType type;
+ private int colorFormat; // Used by native code.
+
// SPS and PPS NALs (Config frame) for H.264.
private ByteBuffer configData = null;
- private MediaCodecVideoEncoder() {
+ // MediaCodec error handler - invoked when critical error happens which may prevent
+ // further use of media codec API. Now it means that one of media codec instances
+ // is hanging and can no longer be used in the next call.
+ public static interface MediaCodecVideoEncoderErrorCallback {
+ void onMediaCodecVideoEncoderCriticalError(int codecErrors);
+ }
+
+ public static void setErrorCallback(MediaCodecVideoEncoderErrorCallback errorCallback) {
+ Logging.d(TAG, "Set error callback");
+ MediaCodecVideoEncoder.errorCallback = errorCallback;
}
// Helper struct for findHwEncoder() below.
@@ -119,7 +149,7 @@
}
private static EncoderProperties findHwEncoder(
- String mime, String[] supportedHwCodecPrefixes) {
+ String mime, String[] supportedHwCodecPrefixes, int[] colorList) {
// MediaCodec.setParameters is missing for JB and below, so bitrate
// can not be adjusted dynamically.
if (Build.VERSION.SDK_INT < Build.VERSION_CODES.KITKAT) {
@@ -130,8 +160,7 @@
if (mime.equals(H264_MIME_TYPE)) {
List<String> exceptionModels = Arrays.asList(H264_HW_EXCEPTION_MODELS);
if (exceptionModels.contains(Build.MODEL)) {
- Logging.w(TAG, "Model: " + Build.MODEL +
- " has black listed H.264 encoder.");
+ Logging.w(TAG, "Model: " + Build.MODEL + " has black listed H.264 encoder.");
return null;
}
}
@@ -170,8 +199,7 @@
Logging.v(TAG, " Color: 0x" + Integer.toHexString(colorFormat));
}
- // Check if codec supports either yuv420 or nv12.
- for (int supportedColorFormat : supportedColorList) {
+ for (int supportedColorFormat : colorList) {
for (int codecColorFormat : capabilities.colorFormats) {
if (codecColorFormat == supportedColorFormat) {
// Found supported HW encoder.
@@ -182,15 +210,34 @@
}
}
}
- return null; // No HW VP8 encoder.
+ return null; // No HW encoder.
}
public static boolean isVp8HwSupported() {
- return findHwEncoder(VP8_MIME_TYPE, supportedVp8HwCodecPrefixes) != null;
+ return findHwEncoder(VP8_MIME_TYPE, supportedVp8HwCodecPrefixes, supportedColorList) != null;
+ }
+
+ public static boolean isVp9HwSupported() {
+ return findHwEncoder(VP9_MIME_TYPE, supportedVp9HwCodecPrefixes, supportedColorList) != null;
}
public static boolean isH264HwSupported() {
- return findHwEncoder(H264_MIME_TYPE, supportedH264HwCodecPrefixes) != null;
+ return findHwEncoder(H264_MIME_TYPE, supportedH264HwCodecPrefixes, supportedColorList) != null;
+ }
+
+ public static boolean isVp8HwSupportedUsingTextures() {
+ return findHwEncoder(
+ VP8_MIME_TYPE, supportedVp8HwCodecPrefixes, supportedSurfaceColorList) != null;
+ }
+
+ public static boolean isVp9HwSupportedUsingTextures() {
+ return findHwEncoder(
+ VP9_MIME_TYPE, supportedVp9HwCodecPrefixes, supportedSurfaceColorList) != null;
+ }
+
+ public static boolean isH264HwSupportedUsingTextures() {
+ return findHwEncoder(
+ H264_MIME_TYPE, supportedH264HwCodecPrefixes, supportedSurfaceColorList) != null;
}
private void checkOnMediaCodecThread() {
@@ -223,32 +270,43 @@
}
}
- // Return the array of input buffers, or null on failure.
- private ByteBuffer[] initEncode(
- VideoCodecType type, int width, int height, int kbps, int fps) {
+ boolean initEncode(VideoCodecType type, int width, int height, int kbps, int fps,
+ EglBase14.Context sharedContext) {
+ final boolean useSurface = sharedContext != null;
Logging.d(TAG, "Java initEncode: " + type + " : " + width + " x " + height +
- ". @ " + kbps + " kbps. Fps: " + fps +
- ". Color: 0x" + Integer.toHexString(colorFormat));
+ ". @ " + kbps + " kbps. Fps: " + fps + ". Encode from texture : " + useSurface);
+
+ this.width = width;
+ this.height = height;
if (mediaCodecThread != null) {
throw new RuntimeException("Forgot to release()?");
}
- this.type = type;
EncoderProperties properties = null;
String mime = null;
int keyFrameIntervalSec = 0;
if (type == VideoCodecType.VIDEO_CODEC_VP8) {
mime = VP8_MIME_TYPE;
- properties = findHwEncoder(VP8_MIME_TYPE, supportedVp8HwCodecPrefixes);
+ properties = findHwEncoder(VP8_MIME_TYPE, supportedVp8HwCodecPrefixes,
+ useSurface ? supportedSurfaceColorList : supportedColorList);
+ keyFrameIntervalSec = 100;
+ } else if (type == VideoCodecType.VIDEO_CODEC_VP9) {
+ mime = VP9_MIME_TYPE;
+ properties = findHwEncoder(VP9_MIME_TYPE, supportedH264HwCodecPrefixes,
+ useSurface ? supportedSurfaceColorList : supportedColorList);
keyFrameIntervalSec = 100;
} else if (type == VideoCodecType.VIDEO_CODEC_H264) {
mime = H264_MIME_TYPE;
- properties = findHwEncoder(H264_MIME_TYPE, supportedH264HwCodecPrefixes);
+ properties = findHwEncoder(H264_MIME_TYPE, supportedH264HwCodecPrefixes,
+ useSurface ? supportedSurfaceColorList : supportedColorList);
keyFrameIntervalSec = 20;
}
if (properties == null) {
throw new RuntimeException("Can not find HW encoder for " + type);
}
runningInstance = this; // Encoder is now running and can be queried for stack traces.
+ colorFormat = properties.colorFormat;
+ Logging.d(TAG, "Color format: " + colorFormat);
+
mediaCodecThread = Thread.currentThread();
try {
MediaFormat format = MediaFormat.createVideoFormat(mime, width, height);
@@ -259,26 +317,39 @@
format.setInteger(MediaFormat.KEY_I_FRAME_INTERVAL, keyFrameIntervalSec);
Logging.d(TAG, " Format: " + format);
mediaCodec = createByCodecName(properties.codecName);
+ this.type = type;
if (mediaCodec == null) {
Logging.e(TAG, "Can not create media encoder");
- return null;
+ return false;
}
mediaCodec.configure(
format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
+
+ if (useSurface) {
+ eglBase = new EglBase14(sharedContext, EglBase.CONFIG_RECORDABLE);
+ // Create an input surface and keep a reference since we must release the surface when done.
+ inputSurface = mediaCodec.createInputSurface();
+ eglBase.createSurface(inputSurface);
+ drawer = new GlRectDrawer();
+ }
mediaCodec.start();
- colorFormat = properties.colorFormat;
outputBuffers = mediaCodec.getOutputBuffers();
- ByteBuffer[] inputBuffers = mediaCodec.getInputBuffers();
- Logging.d(TAG, "Input buffers: " + inputBuffers.length +
- ". Output buffers: " + outputBuffers.length);
- return inputBuffers;
+ Logging.d(TAG, "Output buffers: " + outputBuffers.length);
+
} catch (IllegalStateException e) {
Logging.e(TAG, "initEncode failed", e);
- return null;
+ return false;
}
+ return true;
}
- private boolean encode(
+ ByteBuffer[] getInputBuffers() {
+ ByteBuffer[] inputBuffers = mediaCodec.getInputBuffers();
+ Logging.d(TAG, "Input buffers: " + inputBuffers.length);
+ return inputBuffers;
+ }
+
+ boolean encodeBuffer(
boolean isKeyframe, int inputBuffer, int size,
long presentationTimestampUs) {
checkOnMediaCodecThread();
@@ -298,22 +369,82 @@
return true;
}
catch (IllegalStateException e) {
- Logging.e(TAG, "encode failed", e);
+ Logging.e(TAG, "encodeBuffer failed", e);
return false;
}
}
- private void release() {
- Logging.d(TAG, "Java releaseEncoder");
+ boolean encodeTexture(boolean isKeyframe, int oesTextureId, float[] transformationMatrix,
+ long presentationTimestampUs) {
checkOnMediaCodecThread();
try {
- mediaCodec.stop();
- mediaCodec.release();
- } catch (IllegalStateException e) {
- Logging.e(TAG, "release failed", e);
+ if (isKeyframe) {
+ Logging.d(TAG, "Sync frame request");
+ Bundle b = new Bundle();
+ b.putInt(MediaCodec.PARAMETER_KEY_REQUEST_SYNC_FRAME, 0);
+ mediaCodec.setParameters(b);
+ }
+ eglBase.makeCurrent();
+ // TODO(perkj): glClear() shouldn't be necessary since every pixel is covered anyway,
+ // but it's a workaround for bug webrtc:5147.
+ GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
+ drawer.drawOes(oesTextureId, transformationMatrix, 0, 0, width, height);
+ eglBase.swapBuffers(TimeUnit.MICROSECONDS.toNanos(presentationTimestampUs));
+ return true;
}
+ catch (RuntimeException e) {
+ Logging.e(TAG, "encodeTexture failed", e);
+ return false;
+ }
+ }
+
+ void release() {
+ Logging.d(TAG, "Java releaseEncoder");
+ checkOnMediaCodecThread();
+
+ // Run Mediacodec stop() and release() on separate thread since sometime
+ // Mediacodec.stop() may hang.
+ final CountDownLatch releaseDone = new CountDownLatch(1);
+
+ Runnable runMediaCodecRelease = new Runnable() {
+ @Override
+ public void run() {
+ try {
+ Logging.d(TAG, "Java releaseEncoder on release thread");
+ mediaCodec.stop();
+ mediaCodec.release();
+ Logging.d(TAG, "Java releaseEncoder on release thread done");
+ } catch (Exception e) {
+ Logging.e(TAG, "Media encoder release failed", e);
+ }
+ releaseDone.countDown();
+ }
+ };
+ new Thread(runMediaCodecRelease).start();
+
+ if (!ThreadUtils.awaitUninterruptibly(releaseDone, MEDIA_CODEC_RELEASE_TIMEOUT_MS)) {
+ Logging.e(TAG, "Media encoder release timeout");
+ codecErrors++;
+ if (errorCallback != null) {
+ Logging.e(TAG, "Invoke codec error callback. Errors: " + codecErrors);
+ errorCallback.onMediaCodecVideoEncoderCriticalError(codecErrors);
+ }
+ }
+
mediaCodec = null;
mediaCodecThread = null;
+ if (drawer != null) {
+ drawer.release();
+ drawer = null;
+ }
+ if (eglBase != null) {
+ eglBase.release();
+ eglBase = null;
+ }
+ if (inputSurface != null) {
+ inputSurface.release();
+ inputSurface = null;
+ }
runningInstance = null;
Logging.d(TAG, "Java releaseEncoder done");
}
@@ -336,7 +467,7 @@
// Dequeue an input buffer and return its index, -1 if no input buffer is
// available, or -2 if the codec is no longer operative.
- private int dequeueInputBuffer() {
+ int dequeueInputBuffer() {
checkOnMediaCodecThread();
try {
return mediaCodec.dequeueInputBuffer(DEQUEUE_TIMEOUT);
@@ -347,7 +478,7 @@
}
// Helper struct for dequeueOutputBuffer() below.
- private static class OutputBufferInfo {
+ static class OutputBufferInfo {
public OutputBufferInfo(
int index, ByteBuffer buffer,
boolean isKeyFrame, long presentationTimestampUs) {
@@ -357,15 +488,15 @@
this.presentationTimestampUs = presentationTimestampUs;
}
- private final int index;
- private final ByteBuffer buffer;
- private final boolean isKeyFrame;
- private final long presentationTimestampUs;
+ public final int index;
+ public final ByteBuffer buffer;
+ public final boolean isKeyFrame;
+ public final long presentationTimestampUs;
}
// Dequeue and return an output buffer, or null if no output is ready. Return
// a fake OutputBufferInfo with index -1 if the codec is no longer operable.
- private OutputBufferInfo dequeueOutputBuffer() {
+ OutputBufferInfo dequeueOutputBuffer() {
checkOnMediaCodecThread();
try {
MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
@@ -434,7 +565,7 @@
// Release a dequeued output buffer back to the codec for re-use. Return
// false if the codec is no longer operable.
- private boolean releaseOutputBuffer(int index) {
+ boolean releaseOutputBuffer(int index) {
checkOnMediaCodecThread();
try {
mediaCodec.releaseOutputBuffer(index, false);
diff --git a/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java b/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
index 5002300..36cd075 100644
--- a/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
+++ b/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java
@@ -28,7 +28,6 @@
package org.webrtc;
-import java.util.ArrayList;
import java.util.Collections;
import java.util.LinkedList;
import java.util.List;
@@ -151,6 +150,7 @@
public int audioJitterBufferMaxPackets;
public boolean audioJitterBufferFastAccelerate;
public int iceConnectionReceivingTimeout;
+ public int iceBackupCandidatePairPingInterval;
public KeyType keyType;
public ContinualGatheringPolicy continualGatheringPolicy;
@@ -163,6 +163,7 @@
audioJitterBufferMaxPackets = 50;
audioJitterBufferFastAccelerate = false;
iceConnectionReceivingTimeout = -1;
+ iceBackupCandidatePairPingInterval = -1;
keyType = KeyType.ECDSA;
continualGatheringPolicy = ContinualGatheringPolicy.GATHER_ONCE;
}
@@ -223,6 +224,14 @@
localStreams.remove(stream);
}
+ public RtpSender createSender(String kind, String stream_id) {
+ RtpSender new_sender = nativeCreateSender(kind, stream_id);
+ if (new_sender != null) {
+ senders.add(new_sender);
+ }
+ return new_sender;
+ }
+
// Note that calling getSenders will dispose of the senders previously
// returned (and same goes for getReceivers).
public List<RtpSender> getSenders() {
@@ -288,6 +297,8 @@
private native boolean nativeGetStats(
StatsObserver observer, long nativeTrack);
+ private native RtpSender nativeCreateSender(String kind, String stream_id);
+
private native List<RtpSender> nativeGetSenders();
private native List<RtpReceiver> nativeGetReceivers();
diff --git a/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java b/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
index 83999ec..d759c69 100644
--- a/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
+++ b/talk/app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java
@@ -73,6 +73,15 @@
// Field trial initialization. Must be called before PeerConnectionFactory
// is created.
public static native void initializeFieldTrials(String fieldTrialsInitString);
+ // Internal tracing initialization. Must be called before PeerConnectionFactory is created to
+ // prevent racing with tracing code.
+ public static native void initializeInternalTracer();
+ // Internal tracing shutdown, called to prevent resource leaks. Must be called after
+ // PeerConnectionFactory is gone to prevent races with code performing tracing.
+ public static native void shutdownInternalTracer();
+ // Start/stop internal capturing of internal tracing.
+ public static native boolean startInternalTracingCapture(String tracing_filename);
+ public static native void stopInternalTracingCapture();
public PeerConnectionFactory() {
nativeFactory = nativeCreatePeerConnectionFactory();
@@ -131,12 +140,52 @@
nativeFactory, id, source.nativeSource));
}
+ // Starts recording an AEC dump. Ownership of the file is transfered to the
+ // native code. If an AEC dump is already in progress, it will be stopped and
+ // a new one will start using the provided file.
+ public boolean startAecDump(int file_descriptor) {
+ return nativeStartAecDump(nativeFactory, file_descriptor);
+ }
+
+ // Stops recording an AEC dump. If no AEC dump is currently being recorded,
+ // this call will have no effect.
+ public void stopAecDump() {
+ nativeStopAecDump(nativeFactory);
+ }
+
+ // Starts recording an RTC event log. Ownership of the file is transfered to
+ // the native code. If an RTC event log is already being recorded, it will be
+ // stopped and a new one will start using the provided file.
+ public boolean startRtcEventLog(int file_descriptor) {
+ return nativeStartRtcEventLog(nativeFactory, file_descriptor);
+ }
+
+ // Stops recording an RTC event log. If no RTC event log is currently being
+ // recorded, this call will have no effect.
+ public void StopRtcEventLog() {
+ nativeStopRtcEventLog(nativeFactory);
+ }
+
public void setOptions(Options options) {
nativeSetOptions(nativeFactory, options);
}
+ @Deprecated
public void setVideoHwAccelerationOptions(Object renderEGLContext) {
- nativeSetVideoHwAccelerationOptions(nativeFactory, renderEGLContext);
+ nativeSetVideoHwAccelerationOptions(nativeFactory, renderEGLContext, renderEGLContext);
+ }
+
+ /** Set the EGL context used by HW Video encoding and decoding.
+ *
+ *
+ * @param localEGLContext An instance of javax.microedition.khronos.egl.EGLContext.
+ * Must be the same as used by VideoCapturerAndroid and any local
+ * video renderer.
+ * @param remoteEGLContext An instance of javax.microedition.khronos.egl.EGLContext.
+ * Must be the same as used by any remote video renderer.
+ */
+ public void setVideoHwAccelerationOptions(Object localEGLContext, Object remoteEGLContext) {
+ nativeSetVideoHwAccelerationOptions(nativeFactory, localEGLContext, remoteEGLContext);
}
public void dispose() {
@@ -201,10 +250,18 @@
private static native long nativeCreateAudioTrack(
long nativeFactory, String id, long nativeSource);
+ private static native boolean nativeStartAecDump(long nativeFactory, int file_descriptor);
+
+ private static native void nativeStopAecDump(long nativeFactory);
+
+ private static native boolean nativeStartRtcEventLog(long nativeFactory, int file_descriptor);
+
+ private static native void nativeStopRtcEventLog(long nativeFactory);
+
public native void nativeSetOptions(long nativeFactory, Options options);
private static native void nativeSetVideoHwAccelerationOptions(
- long nativeFactory, Object renderEGLContext);
+ long nativeFactory, Object localEGLContext, Object remoteEGLContext);
private static native void nativeThreadsCallbacks(long nativeFactory);
diff --git a/talk/app/webrtc/java/src/org/webrtc/RtpSender.java b/talk/app/webrtc/java/src/org/webrtc/RtpSender.java
index 37357c0..9ac2e70 100644
--- a/talk/app/webrtc/java/src/org/webrtc/RtpSender.java
+++ b/talk/app/webrtc/java/src/org/webrtc/RtpSender.java
@@ -32,6 +32,7 @@
final long nativeRtpSender;
private MediaStreamTrack cachedTrack;
+ private boolean ownsTrack = true;
public RtpSender(long nativeRtpSender) {
this.nativeRtpSender = nativeRtpSender;
@@ -40,14 +41,22 @@
cachedTrack = (track == 0) ? null : new MediaStreamTrack(track);
}
- // NOTE: This should not be called with a track that's already used by
- // another RtpSender, because then it would be double-disposed.
- public void setTrack(MediaStreamTrack track) {
- if (cachedTrack != null) {
+ // If |takeOwnership| is true, the RtpSender takes ownership of the track
+ // from the caller, and will auto-dispose of it when no longer needed.
+ // |takeOwnership| should only be used if the caller owns the track; it is
+ // not appropriate when the track is owned by, for example, another RtpSender
+ // or a MediaStream.
+ public boolean setTrack(MediaStreamTrack track, boolean takeOwnership) {
+ if (!nativeSetTrack(nativeRtpSender,
+ (track == null) ? 0 : track.nativeTrack)) {
+ return false;
+ }
+ if (cachedTrack != null && ownsTrack) {
cachedTrack.dispose();
}
cachedTrack = track;
- nativeSetTrack(nativeRtpSender, (track == null) ? 0 : track.nativeTrack);
+ ownsTrack = takeOwnership;
+ return true;
}
public MediaStreamTrack track() {
@@ -59,14 +68,14 @@
}
public void dispose() {
- if (cachedTrack != null) {
+ if (cachedTrack != null && ownsTrack) {
cachedTrack.dispose();
}
free(nativeRtpSender);
}
- private static native void nativeSetTrack(long nativeRtpSender,
- long nativeTrack);
+ private static native boolean nativeSetTrack(long nativeRtpSender,
+ long nativeTrack);
// This should increment the reference count of the track.
// Will be released in dispose() or setTrack().
diff --git a/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java b/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java
index 3c255dd..2e307fc 100644
--- a/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java
+++ b/talk/app/webrtc/java/src/org/webrtc/VideoRenderer.java
@@ -46,7 +46,11 @@
public final int[] yuvStrides;
public ByteBuffer[] yuvPlanes;
public final boolean yuvFrame;
- public Object textureObject;
+ // Matrix that transforms standard coordinates to their proper sampling locations in
+ // the texture. This transform compensates for any properties of the video source that
+ // cause it to appear different from a normalized texture. This matrix does not take
+ // |rotationDegree| into account.
+ public final float[] samplingMatrix;
public int textureId;
// Frame pointer in C++.
private long nativeFramePointer;
@@ -70,19 +74,27 @@
if (rotationDegree % 90 != 0) {
throw new IllegalArgumentException("Rotation degree not multiple of 90: " + rotationDegree);
}
+ // The convention in WebRTC is that the first element in a ByteBuffer corresponds to the
+ // top-left corner of the image, but in glTexImage2D() the first element corresponds to the
+ // bottom-left corner. This discrepancy is corrected by setting a vertical flip as sampling
+ // matrix.
+ samplingMatrix = new float[] {
+ 1, 0, 0, 0,
+ 0, -1, 0, 0,
+ 0, 0, 1, 0,
+ 0, 1, 0, 1};
}
/**
* Construct a texture frame of the given dimensions with data in SurfaceTexture
*/
- I420Frame(
- int width, int height, int rotationDegree,
- Object textureObject, int textureId, long nativeFramePointer) {
+ I420Frame(int width, int height, int rotationDegree, int textureId, float[] samplingMatrix,
+ long nativeFramePointer) {
this.width = width;
this.height = height;
this.yuvStrides = null;
this.yuvPlanes = null;
- this.textureObject = textureObject;
+ this.samplingMatrix = samplingMatrix;
this.textureId = textureId;
this.yuvFrame = false;
this.rotationDegree = rotationDegree;
@@ -125,7 +137,6 @@
*/
public static void renderFrameDone(I420Frame frame) {
frame.yuvPlanes = null;
- frame.textureObject = null;
frame.textureId = 0;
if (frame.nativeFramePointer != 0) {
releaseNativeFrame(frame.nativeFramePointer);
diff --git a/talk/app/webrtc/jsepsessiondescription.cc b/talk/app/webrtc/jsepsessiondescription.cc
index 24bd9d4..226432d 100644
--- a/talk/app/webrtc/jsepsessiondescription.cc
+++ b/talk/app/webrtc/jsepsessiondescription.cc
@@ -29,6 +29,7 @@
#include "talk/app/webrtc/webrtcsdp.h"
#include "talk/session/media/mediasession.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/stringencode.h"
using rtc::scoped_ptr;
@@ -44,7 +45,7 @@
static bool IsTypeSupported(const std::string& type) {
bool type_supported = false;
- for (size_t i = 0; i < ARRAY_SIZE(kSupportedTypes); ++i) {
+ for (size_t i = 0; i < arraysize(kSupportedTypes); ++i) {
if (kSupportedTypes[i] == type) {
type_supported = true;
break;
diff --git a/talk/app/webrtc/localaudiosource.cc b/talk/app/webrtc/localaudiosource.cc
index 63c6f13..591877a 100644
--- a/talk/app/webrtc/localaudiosource.cc
+++ b/talk/app/webrtc/localaudiosource.cc
@@ -49,7 +49,7 @@
// a different algorithm will be required.
struct {
const char* name;
- cricket::Settable<bool>& value;
+ rtc::Optional<bool>& value;
} key_to_value[] = {
{MediaConstraintsInterface::kGoogEchoCancellation,
options->echo_cancellation},
@@ -78,7 +78,7 @@
for (auto& entry : key_to_value) {
if (constraint.key.compare(entry.name) == 0)
- entry.value.Set(value);
+ entry.value = rtc::Optional<bool>(value);
}
}
}
diff --git a/talk/app/webrtc/localaudiosource.h b/talk/app/webrtc/localaudiosource.h
index 557745b..5158eb1 100644
--- a/talk/app/webrtc/localaudiosource.h
+++ b/talk/app/webrtc/localaudiosource.h
@@ -48,16 +48,17 @@
const PeerConnectionFactoryInterface::Options& options,
const MediaConstraintsInterface* constraints);
- virtual SourceState state() const { return source_state_; }
+ SourceState state() const override { return source_state_; }
+ bool remote() const override { return false; }
+
virtual const cricket::AudioOptions& options() const { return options_; }
- protected:
- LocalAudioSource()
- : source_state_(kInitializing) {
- }
+ void AddSink(AudioTrackSinkInterface* sink) override {}
+ void RemoveSink(AudioTrackSinkInterface* sink) override {}
- ~LocalAudioSource() {
- }
+ protected:
+ LocalAudioSource() : source_state_(kInitializing) {}
+ ~LocalAudioSource() override {}
private:
void Initialize(const PeerConnectionFactoryInterface::Options& options,
diff --git a/talk/app/webrtc/localaudiosource_unittest.cc b/talk/app/webrtc/localaudiosource_unittest.cc
index 8e05c18..75d0c35 100644
--- a/talk/app/webrtc/localaudiosource_unittest.cc
+++ b/talk/app/webrtc/localaudiosource_unittest.cc
@@ -58,23 +58,14 @@
LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
&constraints);
- bool value;
- EXPECT_TRUE(source->options().echo_cancellation.Get(&value));
- EXPECT_FALSE(value);
- EXPECT_TRUE(source->options().extended_filter_aec.Get(&value));
- EXPECT_TRUE(value);
- EXPECT_TRUE(source->options().delay_agnostic_aec.Get(&value));
- EXPECT_TRUE(value);
- EXPECT_TRUE(source->options().auto_gain_control.Get(&value));
- EXPECT_TRUE(value);
- EXPECT_TRUE(source->options().experimental_agc.Get(&value));
- EXPECT_TRUE(value);
- EXPECT_TRUE(source->options().noise_suppression.Get(&value));
- EXPECT_FALSE(value);
- EXPECT_TRUE(source->options().highpass_filter.Get(&value));
- EXPECT_TRUE(value);
- EXPECT_TRUE(source->options().aec_dump.Get(&value));
- EXPECT_TRUE(value);
+ EXPECT_EQ(rtc::Optional<bool>(false), source->options().echo_cancellation);
+ EXPECT_EQ(rtc::Optional<bool>(true), source->options().extended_filter_aec);
+ EXPECT_EQ(rtc::Optional<bool>(true), source->options().delay_agnostic_aec);
+ EXPECT_EQ(rtc::Optional<bool>(true), source->options().auto_gain_control);
+ EXPECT_EQ(rtc::Optional<bool>(true), source->options().experimental_agc);
+ EXPECT_EQ(rtc::Optional<bool>(false), source->options().noise_suppression);
+ EXPECT_EQ(rtc::Optional<bool>(true), source->options().highpass_filter);
+ EXPECT_EQ(rtc::Optional<bool>(true), source->options().aec_dump);
}
TEST(LocalAudioSourceTest, OptionNotSet) {
@@ -82,8 +73,7 @@
rtc::scoped_refptr<LocalAudioSource> source =
LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
&constraints);
- bool value;
- EXPECT_FALSE(source->options().highpass_filter.Get(&value));
+ EXPECT_EQ(rtc::Optional<bool>(), source->options().highpass_filter);
}
TEST(LocalAudioSourceTest, MandatoryOverridesOptional) {
@@ -97,9 +87,7 @@
LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
&constraints);
- bool value;
- EXPECT_TRUE(source->options().echo_cancellation.Get(&value));
- EXPECT_FALSE(value);
+ EXPECT_EQ(rtc::Optional<bool>(false), source->options().echo_cancellation);
}
TEST(LocalAudioSourceTest, InvalidOptional) {
@@ -112,9 +100,7 @@
&constraints);
EXPECT_EQ(MediaSourceInterface::kLive, source->state());
- bool value;
- EXPECT_TRUE(source->options().highpass_filter.Get(&value));
- EXPECT_FALSE(value);
+ EXPECT_EQ(rtc::Optional<bool>(false), source->options().highpass_filter);
}
TEST(LocalAudioSourceTest, InvalidMandatory) {
@@ -127,7 +113,5 @@
&constraints);
EXPECT_EQ(MediaSourceInterface::kLive, source->state());
- bool value;
- EXPECT_TRUE(source->options().highpass_filter.Get(&value));
- EXPECT_FALSE(value);
+ EXPECT_EQ(rtc::Optional<bool>(false), source->options().highpass_filter);
}
diff --git a/talk/app/webrtc/mediacontroller.cc b/talk/app/webrtc/mediacontroller.cc
index f7d8511..24f5877 100644
--- a/talk/app/webrtc/mediacontroller.cc
+++ b/talk/app/webrtc/mediacontroller.cc
@@ -47,11 +47,10 @@
RTC_DCHECK(nullptr != worker_thread);
worker_thread_->Invoke<void>(
rtc::Bind(&MediaController::Construct_w, this,
- channel_manager_->media_engine()->GetVoE()));
+ channel_manager_->media_engine()));
}
~MediaController() override {
- worker_thread_->Invoke<void>(
- rtc::Bind(&MediaController::Destruct_w, this));
+ worker_thread_->Invoke<void>(rtc::Bind(&MediaController::Destruct_w, this));
}
webrtc::Call* call_w() override {
@@ -64,10 +63,11 @@
}
private:
- void Construct_w(webrtc::VoiceEngine* voice_engine) {
+ void Construct_w(cricket::MediaEngineInterface* media_engine) {
RTC_DCHECK(worker_thread_->IsCurrent());
+ RTC_DCHECK(media_engine);
webrtc::Call::Config config;
- config.voice_engine = voice_engine;
+ config.audio_state = media_engine->GetAudioState();
config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
@@ -84,7 +84,7 @@
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaController);
};
-} // namespace {
+} // namespace {
namespace webrtc {
@@ -93,4 +93,4 @@
cricket::ChannelManager* channel_manager) {
return new MediaController(worker_thread, channel_manager);
}
-} // namespace webrtc
+} // namespace webrtc
diff --git a/talk/app/webrtc/mediastream_unittest.cc b/talk/app/webrtc/mediastream_unittest.cc
index 2cf930c..f19b945 100644
--- a/talk/app/webrtc/mediastream_unittest.cc
+++ b/talk/app/webrtc/mediastream_unittest.cc
@@ -48,9 +48,23 @@
// Helper class to test Observer.
class MockObserver : public ObserverInterface {
public:
- MockObserver() {}
+ explicit MockObserver(NotifierInterface* notifier) : notifier_(notifier) {
+ notifier_->RegisterObserver(this);
+ }
+
+ ~MockObserver() { Unregister(); }
+
+ void Unregister() {
+ if (notifier_) {
+ notifier_->UnregisterObserver(this);
+ notifier_ = nullptr;
+ }
+ }
MOCK_METHOD0(OnChanged, void());
+
+ private:
+ NotifierInterface* notifier_;
};
class MediaStreamTest: public testing::Test {
@@ -75,8 +89,7 @@
}
void ChangeTrack(MediaStreamTrackInterface* track) {
- MockObserver observer;
- track->RegisterObserver(&observer);
+ MockObserver observer(track);
EXPECT_CALL(observer, OnChanged())
.Times(Exactly(1));
@@ -127,8 +140,7 @@
}
TEST_F(MediaStreamTest, RemoveTrack) {
- MockObserver observer;
- stream_->RegisterObserver(&observer);
+ MockObserver observer(stream_);
EXPECT_CALL(observer, OnChanged())
.Times(Exactly(2));
diff --git a/talk/app/webrtc/mediastreaminterface.h b/talk/app/webrtc/mediastreaminterface.h
index 5911e85..9b137d9 100644
--- a/talk/app/webrtc/mediastreaminterface.h
+++ b/talk/app/webrtc/mediastreaminterface.h
@@ -71,8 +71,6 @@
// Base class for sources. A MediaStreamTrack have an underlying source that
// provide media. A source can be shared with multiple tracks.
-// TODO(perkj): Implement sources for local and remote audio tracks and
-// remote video tracks.
class MediaSourceInterface : public rtc::RefCountInterface,
public NotifierInterface {
public:
@@ -85,6 +83,8 @@
virtual SourceState state() const = 0;
+ virtual bool remote() const = 0;
+
protected:
virtual ~MediaSourceInterface() {}
};
@@ -100,6 +100,9 @@
kFailed = 3, // Track negotiation failed.
};
+ static const char kAudioKind[];
+ static const char kVideoKind[];
+
virtual std::string kind() const = 0;
virtual std::string id() const = 0;
virtual bool enabled() const = 0;
@@ -115,13 +118,6 @@
// Interface for rendering VideoFrames from a VideoTrack
class VideoRendererInterface {
public:
- // TODO(guoweis): Remove this function. Obsolete. The implementation of
- // VideoRendererInterface should be able to handle different frame size as
- // well as pending rotation. If it can't apply the frame rotation by itself,
- // it should call |frame|.GetCopyWithRotationApplied() to get a frame that has
- // the rotation applied.
- virtual void SetSize(int width, int height) {}
-
// |frame| may have pending rotation. For clients which can't apply rotation,
// |frame|->GetCopyWithRotationApplied() will return a frame that has the
// rotation applied.
@@ -149,6 +145,19 @@
virtual ~VideoTrackInterface() {}
};
+// Interface for receiving audio data from a AudioTrack.
+class AudioTrackSinkInterface {
+ public:
+ virtual void OnData(const void* audio_data,
+ int bits_per_sample,
+ int sample_rate,
+ size_t number_of_channels,
+ size_t number_of_frames) = 0;
+
+ protected:
+ virtual ~AudioTrackSinkInterface() {}
+};
+
// AudioSourceInterface is a reference counted source used for AudioTracks.
// The same source can be used in multiple AudioTracks.
class AudioSourceInterface : public MediaSourceInterface {
@@ -164,23 +173,17 @@
// TODO(xians): Makes all the interface pure virtual after Chrome has their
// implementations.
// Sets the volume to the source. |volume| is in the range of [0, 10].
+ // TODO(tommi): This method should be on the track and ideally volume should
+ // be applied in the track in a way that does not affect clones of the track.
virtual void SetVolume(double volume) {}
// Registers/unregisters observer to the audio source.
virtual void RegisterAudioObserver(AudioObserver* observer) {}
virtual void UnregisterAudioObserver(AudioObserver* observer) {}
-};
-// Interface for receiving audio data from a AudioTrack.
-class AudioTrackSinkInterface {
- public:
- virtual void OnData(const void* audio_data,
- int bits_per_sample,
- int sample_rate,
- int number_of_channels,
- size_t number_of_frames) = 0;
- protected:
- virtual ~AudioTrackSinkInterface() {}
+ // TODO(tommi): Make pure virtual.
+ virtual void AddSink(AudioTrackSinkInterface* sink) {}
+ virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
};
// Interface of the audio processor used by the audio track to collect
diff --git a/talk/app/webrtc/mediastreamobserver.cc b/talk/app/webrtc/mediastreamobserver.cc
new file mode 100644
index 0000000..2650b9a
--- /dev/null
+++ b/talk/app/webrtc/mediastreamobserver.cc
@@ -0,0 +1,101 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "talk/app/webrtc/mediastreamobserver.h"
+
+#include <algorithm>
+
+namespace webrtc {
+
+MediaStreamObserver::MediaStreamObserver(MediaStreamInterface* stream)
+ : stream_(stream),
+ cached_audio_tracks_(stream->GetAudioTracks()),
+ cached_video_tracks_(stream->GetVideoTracks()) {
+ stream_->RegisterObserver(this);
+}
+
+MediaStreamObserver::~MediaStreamObserver() {
+ stream_->UnregisterObserver(this);
+}
+
+void MediaStreamObserver::OnChanged() {
+ AudioTrackVector new_audio_tracks = stream_->GetAudioTracks();
+ VideoTrackVector new_video_tracks = stream_->GetVideoTracks();
+
+ // Find removed audio tracks.
+ for (const auto& cached_track : cached_audio_tracks_) {
+ auto it = std::find_if(
+ new_audio_tracks.begin(), new_audio_tracks.end(),
+ [cached_track](const AudioTrackVector::value_type& new_track) {
+ return new_track->id().compare(cached_track->id()) == 0;
+ });
+ if (it == new_audio_tracks.end()) {
+ SignalAudioTrackRemoved(cached_track.get(), stream_);
+ }
+ }
+
+ // Find added audio tracks.
+ for (const auto& new_track : new_audio_tracks) {
+ auto it = std::find_if(
+ cached_audio_tracks_.begin(), cached_audio_tracks_.end(),
+ [new_track](const AudioTrackVector::value_type& cached_track) {
+ return new_track->id().compare(cached_track->id()) == 0;
+ });
+ if (it == cached_audio_tracks_.end()) {
+ SignalAudioTrackAdded(new_track.get(), stream_);
+ }
+ }
+
+ // Find removed video tracks.
+ for (const auto& cached_track : cached_video_tracks_) {
+ auto it = std::find_if(
+ new_video_tracks.begin(), new_video_tracks.end(),
+ [cached_track](const VideoTrackVector::value_type& new_track) {
+ return new_track->id().compare(cached_track->id()) == 0;
+ });
+ if (it == new_video_tracks.end()) {
+ SignalVideoTrackRemoved(cached_track.get(), stream_);
+ }
+ }
+
+ // Find added video tracks.
+ for (const auto& new_track : new_video_tracks) {
+ auto it = std::find_if(
+ cached_video_tracks_.begin(), cached_video_tracks_.end(),
+ [new_track](const VideoTrackVector::value_type& cached_track) {
+ return new_track->id().compare(cached_track->id()) == 0;
+ });
+ if (it == cached_video_tracks_.end()) {
+ SignalVideoTrackAdded(new_track.get(), stream_);
+ }
+ }
+
+ cached_audio_tracks_ = new_audio_tracks;
+ cached_video_tracks_ = new_video_tracks;
+}
+
+} // namespace webrtc
diff --git a/talk/app/webrtc/mediastreamobserver.h b/talk/app/webrtc/mediastreamobserver.h
new file mode 100644
index 0000000..1dd6c4c
--- /dev/null
+++ b/talk/app/webrtc/mediastreamobserver.h
@@ -0,0 +1,65 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_
+#define TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_
+
+#include "talk/app/webrtc/mediastreaminterface.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/base/sigslot.h"
+
+namespace webrtc {
+
+// Helper class which will listen for changes to a stream and emit the
+// corresponding signals.
+class MediaStreamObserver : public ObserverInterface {
+ public:
+ explicit MediaStreamObserver(MediaStreamInterface* stream);
+ ~MediaStreamObserver();
+
+ const MediaStreamInterface* stream() const { return stream_; }
+
+ void OnChanged() override;
+
+ sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
+ SignalAudioTrackAdded;
+ sigslot::signal2<AudioTrackInterface*, MediaStreamInterface*>
+ SignalAudioTrackRemoved;
+ sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
+ SignalVideoTrackAdded;
+ sigslot::signal2<VideoTrackInterface*, MediaStreamInterface*>
+ SignalVideoTrackRemoved;
+
+ private:
+ rtc::scoped_refptr<MediaStreamInterface> stream_;
+ AudioTrackVector cached_audio_tracks_;
+ VideoTrackVector cached_video_tracks_;
+};
+
+} // namespace webrtc
+
+#endif // TALK_APP_WEBRTC_MEDIASTREAMOBSERVER_H_
diff --git a/talk/app/webrtc/mediastreamprovider.h b/talk/app/webrtc/mediastreamprovider.h
index 1c62daf..585d51b 100644
--- a/talk/app/webrtc/mediastreamprovider.h
+++ b/talk/app/webrtc/mediastreamprovider.h
@@ -29,6 +29,7 @@
#define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
#include "webrtc/base/basictypes.h"
+#include "webrtc/base/scoped_ptr.h"
namespace cricket {
@@ -42,6 +43,8 @@
namespace webrtc {
+class AudioSinkInterface;
+
// TODO(deadbeef): Change the key from an ssrc to a "sender_id" or
// "receiver_id" string, which will be the MSID in the short term and MID in
// the long term.
@@ -50,8 +53,8 @@
// RtpSenders/Receivers to get to the BaseChannels. These interfaces should be
// refactored away eventually, as the classes converge.
-// This interface is called by AudioTrackHandler classes in mediastreamhandler.h
-// to change the settings of an audio track connected to certain PeerConnection.
+// This interface is called by AudioRtpSender/Receivers to change the settings
+// of an audio track connected to certain PeerConnection.
class AudioProviderInterface {
public:
// Enable/disable the audio playout of a remote audio track with |ssrc|.
@@ -67,13 +70,19 @@
// |volume| is in the range of [0, 10].
virtual void SetAudioPlayoutVolume(uint32_t ssrc, double volume) = 0;
+ // Allows for setting a direct audio sink for an incoming audio source.
+ // Only one audio sink is supported per ssrc and ownership of the sink is
+ // passed to the provider.
+ virtual void SetRawAudioSink(
+ uint32_t ssrc,
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0;
+
protected:
virtual ~AudioProviderInterface() {}
};
-// This interface is called by VideoTrackHandler classes in mediastreamhandler.h
-// to change the settings of a video track connected to a certain
-// PeerConnection.
+// This interface is called by VideoRtpSender/Receivers to change the settings
+// of a video track connected to a certain PeerConnection.
class VideoProviderInterface {
public:
virtual bool SetCaptureDevice(uint32_t ssrc,
diff --git a/talk/app/webrtc/mediastreamsignaling.cc b/talk/app/webrtc/mediastreamsignaling.cc
deleted file mode 100644
index b405273..0000000
--- a/talk/app/webrtc/mediastreamsignaling.cc
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- * libjingle
- * Copyright 2012 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/app/webrtc/mediastreamsignaling.h"
-
-// TODO(deadbeef): Remove this file once Chrome build files don't reference it.
diff --git a/talk/app/webrtc/mediastreamsignaling.h b/talk/app/webrtc/mediastreamsignaling.h
deleted file mode 100644
index e8c5c11..0000000
--- a/talk/app/webrtc/mediastreamsignaling.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * libjingle
- * Copyright 2012 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-// TODO(deadbeef): Remove this file once Chrome build files don't reference it.
diff --git a/talk/app/webrtc/objc/README b/talk/app/webrtc/objc/README
index 692fbbc..c323e73 100644
--- a/talk/app/webrtc/objc/README
+++ b/talk/app/webrtc/objc/README
@@ -12,69 +12,59 @@
up for building for iOS-device, iOS-simulator, and Mac (resp) are:
function wrbase() {
cd /path/to/webrtc/trunk
- export GYP_DEFINES="build_with_libjingle=1 build_with_chromium=0 libjingle_objc=1"
+ export GYP_DEFINES="build_with_libjingle=1 build_with_chromium=0"
export GYP_GENERATORS="ninja"
}
function wrios() {
wrbase
- export GYP_DEFINES="$GYP_DEFINES OS=ios target_arch=armv7"
+ export GYP_DEFINES="$GYP_DEFINES OS=ios"
export GYP_GENERATOR_FLAGS="$GYP_GENERATOR_FLAGS output_dir=out_ios"
export GYP_CROSSCOMPILE=1
}
+function wrios32() {
+ wrios
+ export GYP_DEFINES="$GYP_DEFINES target_arch=arm"
+}
+
+function wrios64() {
+ wrios
+ export GYP_DEFINES="$GYP_DEFINES target_arch=arm64"
+}
+
function wrsim() {
wrbase
- export GYP_DEFINES="$GYP_DEFINES OS=ios target_arch=ia32"
+ export GYP_DEFINES="$GYP_DEFINES OS=ios target_subarch=arm32 target_arch=ia32"
export GYP_GENERATOR_FLAGS="$GYP_GENERATOR_FLAGS output_dir=out_sim"
export GYP_CROSSCOMPILE=1
}
function wrmac() {
wrbase
- export GYP_DEFINES="$GYP_DEFINES OS=mac target_arch=x64"
+ export GYP_DEFINES="$GYP_DEFINES OS=mac target_subarch=arm64 target_arch=x64"
export GYP_GENERATOR_FLAGS="$GYP_GENERATOR_FLAGS output_dir=out_mac"
}
-- Finally, run "gclient runhooks" to generate ninja files.
+- Finally, run "webrtc/build/gyp_webrtc" to generate ninja files.
Example of building & using the unittest & app:
- To build & run the unittest (must target mac):
- wrmac && gclient runhooks && \
+ wrmac && ./webrtc/build/gyp_webrtc && \
ninja -C out_mac/Debug libjingle_peerconnection_objc_test && \
./out_mac/Debug/libjingle_peerconnection_objc_test.app/Contents/MacOS/libjingle_peerconnection_objc_test
- To build & launch the sample app on OSX:
- wrmac && gclient runhooks && ninja -C out_mac/Debug AppRTCDemo && \
+ wrmac && ./webrtc/build/gyp_webrtc && ninja -C out_mac/Debug AppRTCDemo && \
./out_mac/Debug/AppRTCDemo.app/Contents/MacOS/AppRTCDemo
- To build & launch the sample app on the iOS simulator:
- wrsim && gclient runhooks && ninja -C out_sim/Debug iossim AppRTCDemo && \
+ wrsim && ./webrtc/build/gyp_webrtc && ninja -C out_sim/Debug iossim AppRTCDemo && \
./out_sim/Debug/iossim out_sim/Debug/AppRTCDemo.app
-- To build & sign the sample app for an iOS device:
- wrios && gclient runhooks && ninja -C out_ios/Debug-iphoneos AppRTCDemo
+- To build & sign the sample app for an iOS device (32 bit):
+ wrios32 && ./webrtc/build/gyp_webrtc && ninja -C out_ios/Debug-iphoneos AppRTCDemo
-- To install the sample app on an iOS device:
- ideviceinstaller -i out_ios/Debug-iphoneos/AppRTCDemo.app
- (if installing ideviceinstaller from brew, use --HEAD to get support
- for .app directories)
-- Alternatively, use iPhone Configuration Utility:
- - Open "iPhone Configuration Utility" (http://support.apple.com/kb/DL1465)
- - Click the "Add" icon (command-o)
- - Open the app under out_ios/Debug-iphoneos/AppRTCDemo (should be added to the Applications tab)
- - Click the device's name in the left-hand panel and select the Applications tab
- - Click Install on the AppRTCDemo line.
- (If you have any problems deploying for the first time, check
- the Info.plist file to ensure that the Bundle Identifier matches
- your phone provisioning profile, or use a development wildcard
- provisioning profile.)
-- Alternately, use ios-deploy:
- ios-deploy -d -b out_ios/Debug-iphoneos/AppRTCDemo.app
-
-- Once installed:
- - Tap AppRTCDemo on the iOS device's home screen (might have to scroll to find it).
- - In desktop chrome, navigate to http://apprtc.appspot.com and note
- the r=<NNN> room number in the resulting URL; enter that number
- into the text field on the phone.
+- To build & sign the sample app for an iOS device (64 bit):
+ wrios64 && ./webrtc/build/gyp_webrtc && ninja -C out_ios/Debug-iphoneos AppRTCDemo
diff --git a/talk/app/webrtc/objc/RTCFileLogger.mm b/talk/app/webrtc/objc/RTCFileLogger.mm
index c4e4696..44ada3e 100644
--- a/talk/app/webrtc/objc/RTCFileLogger.mm
+++ b/talk/app/webrtc/objc/RTCFileLogger.mm
@@ -35,15 +35,17 @@
NSString *const kDefaultLogDirName = @"webrtc_logs";
NSUInteger const kDefaultMaxFileSize = 10 * 1024 * 1024; // 10MB.
+const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
@implementation RTCFileLogger {
BOOL _hasStarted;
NSString *_dirPath;
NSUInteger _maxFileSize;
- rtc::scoped_ptr<rtc::CallSessionFileRotatingLogSink> _logSink;
+ rtc::scoped_ptr<rtc::FileRotatingLogSink> _logSink;
}
@synthesize severity = _severity;
+@synthesize rotationType = _rotationType;
- (instancetype)init {
NSArray *paths = NSSearchPathForDirectoriesInDomains(
@@ -57,6 +59,14 @@
- (instancetype)initWithDirPath:(NSString *)dirPath
maxFileSize:(NSUInteger)maxFileSize {
+ return [self initWithDirPath:dirPath
+ maxFileSize:maxFileSize
+ rotationType:kRTCFileLoggerTypeCall];
+}
+
+- (instancetype)initWithDirPath:(NSString *)dirPath
+ maxFileSize:(NSUInteger)maxFileSize
+ rotationType:(RTCFileLoggerRotationType)rotationType {
NSParameterAssert(dirPath.length);
NSParameterAssert(maxFileSize);
if (self = [super init]) {
@@ -91,8 +101,20 @@
if (_hasStarted) {
return;
}
- _logSink.reset(new rtc::CallSessionFileRotatingLogSink(_dirPath.UTF8String,
- _maxFileSize));
+ switch (_rotationType) {
+ case kRTCFileLoggerTypeApp:
+ _logSink.reset(
+ new rtc::FileRotatingLogSink(_dirPath.UTF8String,
+ kRTCFileLoggerRotatingLogPrefix,
+ _maxFileSize,
+ _maxFileSize / 10));
+ break;
+ case kRTCFileLoggerTypeCall:
+ _logSink.reset(
+ new rtc::CallSessionFileRotatingLogSink(_dirPath.UTF8String,
+ _maxFileSize));
+ break;
+ }
if (!_logSink->Init()) {
LOG(LS_ERROR) << "Failed to open log files at path: "
<< _dirPath.UTF8String;
@@ -120,8 +142,17 @@
return nil;
}
NSMutableData* logData = [NSMutableData data];
- rtc::scoped_ptr<rtc::CallSessionFileRotatingStream> stream(
- new rtc::CallSessionFileRotatingStream(_dirPath.UTF8String));
+ rtc::scoped_ptr<rtc::FileRotatingStream> stream;
+ switch(_rotationType) {
+ case kRTCFileLoggerTypeApp:
+ stream.reset(
+ new rtc::FileRotatingStream(_dirPath.UTF8String,
+ kRTCFileLoggerRotatingLogPrefix));
+ break;
+ case kRTCFileLoggerTypeCall:
+ stream.reset(new rtc::CallSessionFileRotatingStream(_dirPath.UTF8String));
+ break;
+ }
if (!stream->Open()) {
return logData;
}
diff --git a/talk/app/webrtc/objc/RTCPeerConnection.mm b/talk/app/webrtc/objc/RTCPeerConnection.mm
index 44d39cb..f814f06 100644
--- a/talk/app/webrtc/objc/RTCPeerConnection.mm
+++ b/talk/app/webrtc/objc/RTCPeerConnection.mm
@@ -271,11 +271,13 @@
- (instancetype)initWithFactory:(webrtc::PeerConnectionFactoryInterface*)factory
iceServers:(const webrtc::PeerConnectionInterface::IceServers&)iceServers
constraints:(const webrtc::MediaConstraintsInterface*)constraints {
- NSParameterAssert(factory != NULL);
+ NSParameterAssert(factory != nullptr);
if (self = [super init]) {
+ webrtc::PeerConnectionInterface::RTCConfiguration config;
+ config.servers = iceServers;
_observer.reset(new webrtc::RTCPeerConnectionObserver(self));
_peerConnection = factory->CreatePeerConnection(
- iceServers, constraints, NULL, NULL, _observer.get());
+ config, constraints, nullptr, nullptr, _observer.get());
_localStreams = [[NSMutableArray alloc] init];
}
return self;
diff --git a/talk/app/webrtc/objc/RTCPeerConnectionInterface.mm b/talk/app/webrtc/objc/RTCPeerConnectionInterface.mm
index 58d12ac..ff45bd2 100644
--- a/talk/app/webrtc/objc/RTCPeerConnectionInterface.mm
+++ b/talk/app/webrtc/objc/RTCPeerConnectionInterface.mm
@@ -39,6 +39,7 @@
@synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
@synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
@synthesize iceConnectionReceivingTimeout = _iceConnectionReceivingTimeout;
+@synthesize iceBackupCandidatePairPingInterval = _iceBackupCandidatePairPingInterval;
- (instancetype)init {
if (self = [super init]) {
@@ -51,6 +52,7 @@
[RTCEnumConverter tcpCandidatePolicyForNativeEnum:config.tcp_candidate_policy];
_audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
_iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
+ _iceBackupCandidatePairPingInterval = config.ice_backup_candidate_pair_ping_interval;
}
return self;
}
@@ -60,7 +62,8 @@
rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets
- iceConnectionReceivingTimeout:(int)iceConnectionReceivingTimeout {
+ iceConnectionReceivingTimeout:(int)iceConnectionReceivingTimeout
+ iceBackupCandidatePairPingInterval:(int)iceBackupCandidatePairPingInterval {
if (self = [super init]) {
_iceTransportsType = iceTransportsType;
_bundlePolicy = bundlePolicy;
@@ -68,6 +71,7 @@
_tcpCandidatePolicy = tcpCandidatePolicy;
_audioJitterBufferMaxPackets = audioJitterBufferMaxPackets;
_iceConnectionReceivingTimeout = iceConnectionReceivingTimeout;
+ _iceBackupCandidatePairPingInterval = iceBackupCandidatePairPingInterval;
}
return self;
}
@@ -85,8 +89,8 @@
nativeConfig.tcp_candidate_policy =
[RTCEnumConverter nativeEnumForTcpCandidatePolicy:_tcpCandidatePolicy];
nativeConfig.audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
- nativeConfig.ice_connection_receiving_timeout =
- _iceConnectionReceivingTimeout;
+ nativeConfig.ice_connection_receiving_timeout = _iceConnectionReceivingTimeout;
+ nativeConfig.ice_backup_candidate_pair_ping_interval = _iceBackupCandidatePairPingInterval;
return nativeConfig;
}
diff --git a/talk/app/webrtc/objc/avfoundationvideocapturer.h b/talk/app/webrtc/objc/avfoundationvideocapturer.h
index ded80f6..32de09a 100644
--- a/talk/app/webrtc/objc/avfoundationvideocapturer.h
+++ b/talk/app/webrtc/objc/avfoundationvideocapturer.h
@@ -71,7 +71,6 @@
RTCAVFoundationVideoCapturerInternal* _capturer;
rtc::Thread* _startThread; // Set in Start(), unset in Stop().
- uint64_t _startTime;
}; // AVFoundationVideoCapturer
} // namespace webrtc
diff --git a/talk/app/webrtc/objc/avfoundationvideocapturer.mm b/talk/app/webrtc/objc/avfoundationvideocapturer.mm
index e1b0f88..0f9dc68 100644
--- a/talk/app/webrtc/objc/avfoundationvideocapturer.mm
+++ b/talk/app/webrtc/objc/avfoundationvideocapturer.mm
@@ -33,6 +33,8 @@
#import <Foundation/Foundation.h>
#import <UIKit/UIKit.h>
+#import "webrtc/base/objc/RTCDispatcher.h"
+
// TODO(tkchin): support other formats.
static NSString* const kDefaultPreset = AVCaptureSessionPreset640x480;
static cricket::VideoFormat const kDefaultFormat =
@@ -41,11 +43,6 @@
cricket::VideoFormat::FpsToInterval(30),
cricket::FOURCC_NV12);
-// This queue is used to start and stop the capturer without blocking the
-// calling thread. -[AVCaptureSession startRunning] blocks until the camera is
-// running.
-static dispatch_queue_t kBackgroundQueue = nil;
-
// This class used to capture frames using AVFoundation APIs on iOS. It is meant
// to be owned by an instance of AVFoundationVideoCapturer. The reason for this
// because other webrtc objects own cricket::VideoCapturer, which is not
@@ -80,15 +77,6 @@
@synthesize useBackCamera = _useBackCamera;
@synthesize isRunning = _isRunning;
-+ (void)initialize {
- static dispatch_once_t onceToken;
- dispatch_once(&onceToken, ^{
- kBackgroundQueue = dispatch_queue_create(
- "com.google.webrtc.RTCAVFoundationCapturerBackground",
- DISPATCH_QUEUE_SERIAL);
- });
-}
-
- (instancetype)initWithCapturer:(webrtc::AVFoundationVideoCapturer*)capturer {
NSParameterAssert(capturer);
if (self = [super init]) {
@@ -132,9 +120,10 @@
_orientationHasChanged = NO;
[[UIDevice currentDevice] beginGeneratingDeviceOrientationNotifications];
AVCaptureSession* session = _captureSession;
- dispatch_async(kBackgroundQueue, ^{
+ [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+ block:^{
[session startRunning];
- });
+ }];
_isRunning = YES;
}
@@ -144,9 +133,10 @@
}
[_videoOutput setSampleBufferDelegate:nil queue:nullptr];
AVCaptureSession* session = _captureSession;
- dispatch_async(kBackgroundQueue, ^{
+ [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+ block:^{
[session stopRunning];
- });
+ }];
[[UIDevice currentDevice] endGeneratingDeviceOrientationNotifications];
_isRunning = NO;
}
diff --git a/talk/app/webrtc/objc/public/RTCFileLogger.h b/talk/app/webrtc/objc/public/RTCFileLogger.h
index 3900cb6..70b3825 100644
--- a/talk/app/webrtc/objc/public/RTCFileLogger.h
+++ b/talk/app/webrtc/objc/public/RTCFileLogger.h
@@ -39,21 +39,38 @@
kRTCFileLoggerSeverityError
};
+typedef NS_ENUM(NSUInteger, RTCFileLoggerRotationType) {
+ kRTCFileLoggerTypeCall,
+ kRTCFileLoggerTypeApp,
+};
+
// This class intercepts WebRTC logs and saves them to a file. The file size
// will not exceed the given maximum bytesize. When the maximum bytesize is
-// reached logs from the beginning and the end are preserved while the middle
-// section is overwritten instead.
+// reached, logs are rotated according to the rotationType specified.
+// For kRTCFileLoggerTypeCall, logs from the beginning and the end
+// are preserved while the middle section is overwritten instead.
+// For kRTCFileLoggerTypeApp, the oldest log is overwritten.
// This class is not threadsafe.
@interface RTCFileLogger : NSObject
// The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
@property(nonatomic, assign) RTCFileLoggerSeverity severity;
-// Default constructor provides default settings for dir path and file size.
+// The rotation type for this file logger. The default is
+// kRTCFileLoggerTypeCall.
+@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
+
+// Default constructor provides default settings for dir path, file size and
+// rotation type.
- (instancetype)init;
+// Create file logger with default rotation type.
+- (instancetype)initWithDirPath:(NSString *)dirPath
+ maxFileSize:(NSUInteger)maxFileSize;
+
- (instancetype)initWithDirPath:(NSString *)dirPath
maxFileSize:(NSUInteger)maxFileSize
+ rotationType:(RTCFileLoggerRotationType)rotationType
NS_DESIGNATED_INITIALIZER;
// Starts writing WebRTC logs to disk if not already started. Overwrites any
diff --git a/talk/app/webrtc/objc/public/RTCPeerConnectionInterface.h b/talk/app/webrtc/objc/public/RTCPeerConnectionInterface.h
index b0cc72b..44b971c 100644
--- a/talk/app/webrtc/objc/public/RTCPeerConnectionInterface.h
+++ b/talk/app/webrtc/objc/public/RTCPeerConnectionInterface.h
@@ -64,12 +64,14 @@
@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
@property(nonatomic, assign) int audioJitterBufferMaxPackets;
@property(nonatomic, assign) int iceConnectionReceivingTimeout;
+@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
- (instancetype)initWithIceTransportsType:(RTCIceTransportsType)iceTransportsType
bundlePolicy:(RTCBundlePolicy)bundlePolicy
rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets
- iceConnectionReceivingTimeout:(int)iceConnectionReceivingTimeout;
+ iceConnectionReceivingTimeout:(int)iceConnectionReceivingTimeout
+ iceBackupCandidatePairPingInterval:(int)iceBackupCandidatePairPingInterval;
@end
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 0d519b2..ccca18a 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -27,8 +27,10 @@
#include "talk/app/webrtc/peerconnection.h"
-#include <vector>
+#include <algorithm>
#include <cctype> // for isdigit
+#include <utility>
+#include <vector>
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/dtmfsender.h"
@@ -36,6 +38,7 @@
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/mediaconstraintsinterface.h"
#include "talk/app/webrtc/mediastream.h"
+#include "talk/app/webrtc/mediastreamobserver.h"
#include "talk/app/webrtc/mediastreamproxy.h"
#include "talk/app/webrtc/mediastreamtrackproxy.h"
#include "talk/app/webrtc/remoteaudiosource.h"
@@ -46,11 +49,13 @@
#include "talk/app/webrtc/videosource.h"
#include "talk/app/webrtc/videotrack.h"
#include "talk/media/sctp/sctpdataengine.h"
-#include "webrtc/p2p/client/basicportallocator.h"
#include "talk/session/media/channelmanager.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/p2p/client/basicportallocator.h"
#include "webrtc/system_wrappers/include/field_trial.h"
namespace {
@@ -59,13 +64,8 @@
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionInterface;
+using webrtc::RtpSenderInterface;
using webrtc::StreamCollection;
-using webrtc::StunConfigurations;
-using webrtc::TurnConfigurations;
-typedef webrtc::PortAllocatorFactoryInterface::StunConfiguration
- StunConfiguration;
-typedef webrtc::PortAllocatorFactoryInterface::TurnConfiguration
- TurnConfiguration;
static const char kDefaultStreamLabel[] = "default";
static const char kDefaultAudioTrackLabel[] = "defaulta0";
@@ -80,8 +80,6 @@
static const int kDefaultStunPort = 3478;
static const int kDefaultStunTlsPort = 5349;
static const char kTransport[] = "transport";
-static const char kUdpTransportType[] = "udp";
-static const char kTcpTransportType[] = "tcp";
// NOTE: Must be in the same order as the ServiceType enum.
static const char* kValidIceServiceTypes[] = {"stun", "stuns", "turn", "turns"};
@@ -95,7 +93,7 @@
TURNS, // Indicates a TURN server used with a TLS session.
INVALID, // Unknown.
};
-static_assert(INVALID == ARRAY_SIZE(kValidIceServiceTypes),
+static_assert(INVALID == arraysize(kValidIceServiceTypes),
"kValidIceServiceTypes must have as many strings as ServiceType "
"has values.");
@@ -104,6 +102,7 @@
MSG_SET_SESSIONDESCRIPTION_FAILED,
MSG_CREATE_SESSIONDESCRIPTION_FAILED,
MSG_GETSTATS,
+ MSG_FREE_DATACHANNELS,
};
struct SetSessionDescriptionMsg : public rtc::MessageData {
@@ -156,7 +155,7 @@
return false;
}
*service_type = INVALID;
- for (size_t i = 0; i < ARRAY_SIZE(kValidIceServiceTypes); ++i) {
+ for (size_t i = 0; i < arraysize(kValidIceServiceTypes); ++i) {
if (in_str.compare(0, colonpos, kValidIceServiceTypes[i]) == 0) {
*service_type = static_cast<ServiceType>(i);
break;
@@ -216,12 +215,12 @@
return !host->empty();
}
-// Adds a StunConfiguration or TurnConfiguration to the appropriate list,
+// Adds a STUN or TURN server to the appropriate list,
// by parsing |url| and using the username/password in |server|.
bool ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
const std::string& url,
- StunConfigurations* stun_config,
- TurnConfigurations* turn_config) {
+ cricket::ServerAddresses* stun_servers,
+ std::vector<cricket::RelayServerConfig>* turn_servers) {
// draft-nandakumar-rtcweb-stun-uri-01
// stunURI = scheme ":" stun-host [ ":" stun-port ]
// scheme = "stun" / "stuns"
@@ -236,10 +235,10 @@
// transport-ext = 1*unreserved
// turn-host = IP-literal / IPv4address / reg-name
// turn-port = *DIGIT
- RTC_DCHECK(stun_config != nullptr);
- RTC_DCHECK(turn_config != nullptr);
+ RTC_DCHECK(stun_servers != nullptr);
+ RTC_DCHECK(turn_servers != nullptr);
std::vector<std::string> tokens;
- std::string turn_transport_type = kUdpTransportType;
+ cricket::ProtocolType turn_transport_type = cricket::PROTO_UDP;
RTC_DCHECK(!url.empty());
rtc::tokenize(url, '?', &tokens);
std::string uri_without_transport = tokens[0];
@@ -250,11 +249,12 @@
if (tokens[0] == kTransport) {
// As per above grammar transport param will be consist of lower case
// letters.
- if (tokens[1] != kUdpTransportType && tokens[1] != kTcpTransportType) {
+ if (!cricket::StringToProto(tokens[1].c_str(), &turn_transport_type) ||
+ (turn_transport_type != cricket::PROTO_UDP &&
+ turn_transport_type != cricket::PROTO_TCP)) {
LOG(LS_WARNING) << "Transport param should always be udp or tcp.";
return false;
}
- turn_transport_type = tokens[1];
}
}
@@ -293,7 +293,7 @@
int port = kDefaultStunPort;
if (service_type == TURNS) {
port = kDefaultStunTlsPort;
- turn_transport_type = kTcpTransportType;
+ turn_transport_type = cricket::PROTO_TCP;
}
std::string address;
@@ -310,16 +310,14 @@
switch (service_type) {
case STUN:
case STUNS:
- stun_config->push_back(StunConfiguration(address, port));
+ stun_servers->insert(rtc::SocketAddress(address, port));
break;
case TURN:
case TURNS: {
bool secure = (service_type == TURNS);
- turn_config->push_back(TurnConfiguration(address, port,
- username,
- server.password,
- turn_transport_type,
- secure));
+ turn_servers->push_back(
+ cricket::RelayServerConfig(address, port, username, server.password,
+ turn_transport_type, secure));
break;
}
case INVALID:
@@ -365,25 +363,15 @@
}
// Add the stream and RTP data channel info to |session_options|.
-void SetStreams(cricket::MediaSessionOptions* session_options,
- rtc::scoped_refptr<StreamCollection> streams,
- const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
- rtp_data_channels) {
+void AddSendStreams(
+ cricket::MediaSessionOptions* session_options,
+ const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
+ const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
+ rtp_data_channels) {
session_options->streams.clear();
- if (streams != nullptr) {
- for (size_t i = 0; i < streams->count(); ++i) {
- MediaStreamInterface* stream = streams->at(i);
- // For each audio track in the stream, add it to the MediaSessionOptions.
- for (const auto& track : stream->GetAudioTracks()) {
- session_options->AddSendStream(cricket::MEDIA_TYPE_AUDIO, track->id(),
- stream->label());
- }
- // For each video track in the stream, add it to the MediaSessionOptions.
- for (const auto& track : stream->GetVideoTracks()) {
- session_options->AddSendStream(cricket::MEDIA_TYPE_VIDEO, track->id(),
- stream->label());
- }
- }
+ for (const auto& sender : senders) {
+ session_options->AddSendStream(sender->media_type(), sender->id(),
+ sender->stream_id());
}
// Check for data channels.
@@ -421,10 +409,12 @@
MediaStream::Create(stream_label));
}
- AudioTrackInterface* AddAudioTrack(webrtc::MediaStreamInterface* stream,
+ AudioTrackInterface* AddAudioTrack(uint32_t ssrc,
+ AudioProviderInterface* provider,
+ webrtc::MediaStreamInterface* stream,
const std::string& track_id) {
return AddTrack<AudioTrackInterface, AudioTrack, AudioTrackProxy>(
- stream, track_id, RemoteAudioSource::Create().get());
+ stream, track_id, RemoteAudioSource::Create(ssrc, provider));
}
VideoTrackInterface* AddVideoTrack(webrtc::MediaStreamInterface* stream,
@@ -432,7 +422,7 @@
return AddTrack<VideoTrackInterface, VideoTrack, VideoTrackProxy>(
stream, track_id,
VideoSource::Create(channel_manager_, new RemoteVideoCapturer(),
- nullptr)
+ nullptr, true)
.get());
}
@@ -440,7 +430,7 @@
template <typename TI, typename T, typename TP, typename S>
TI* AddTrack(MediaStreamInterface* stream,
const std::string& track_id,
- S* source) {
+ const S& source) {
rtc::scoped_refptr<TI> track(
TP::Create(signaling_thread_, T::Create(track_id, source)));
track->set_state(webrtc::MediaStreamTrackInterface::kLive);
@@ -471,7 +461,11 @@
}
session_options->vad_enabled = rtc_options.voice_activity_detection;
- session_options->transport_options.ice_restart = rtc_options.ice_restart;
+ session_options->audio_transport_options.ice_restart =
+ rtc_options.ice_restart;
+ session_options->video_transport_options.ice_restart =
+ rtc_options.ice_restart;
+ session_options->data_transport_options.ice_restart = rtc_options.ice_restart;
session_options->bundle_enabled = rtc_options.use_rtp_mux;
return true;
@@ -517,10 +511,14 @@
if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
&value, &mandatory_constraints_satisfied)) {
- session_options->transport_options.ice_restart = value;
+ session_options->audio_transport_options.ice_restart = value;
+ session_options->video_transport_options.ice_restart = value;
+ session_options->data_transport_options.ice_restart = value;
} else {
// kIceRestart defaults to false according to spec.
- session_options->transport_options.ice_restart = false;
+ session_options->audio_transport_options.ice_restart = false;
+ session_options->video_transport_options.ice_restart = false;
+ session_options->data_transport_options.ice_restart = false;
}
if (!constraints) {
@@ -530,8 +528,8 @@
}
bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
- StunConfigurations* stun_config,
- TurnConfigurations* turn_config) {
+ cricket::ServerAddresses* stun_servers,
+ std::vector<cricket::RelayServerConfig>* turn_servers) {
for (const webrtc::PeerConnectionInterface::IceServer& server : servers) {
if (!server.urls.empty()) {
for (const std::string& url : server.urls) {
@@ -539,13 +537,13 @@
LOG(LS_ERROR) << "Empty uri.";
return false;
}
- if (!ParseIceServerUrl(server, url, stun_config, turn_config)) {
+ if (!ParseIceServerUrl(server, url, stun_servers, turn_servers)) {
return false;
}
}
} else if (!server.uri.empty()) {
// Fallback to old .uri if new .urls isn't present.
- if (!ParseIceServerUrl(server, server.uri, stun_config, turn_config)) {
+ if (!ParseIceServerUrl(server, server.uri, stun_servers, turn_servers)) {
return false;
}
} else {
@@ -553,6 +551,13 @@
return false;
}
}
+ // Candidates must have unique priorities, so that connectivity checks
+ // are performed in a well-defined order.
+ int priority = static_cast<int>(turn_servers->size() - 1);
+ for (cricket::RelayServerConfig& turn_server : *turn_servers) {
+ // First in the list gets highest priority.
+ turn_server.priority = priority--;
+ }
return true;
}
@@ -568,6 +573,7 @@
remote_streams_(StreamCollection::Create()) {}
PeerConnection::~PeerConnection() {
+ TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
RTC_DCHECK(signaling_thread()->IsCurrent());
// Need to detach RTP senders/receivers from WebRtcSession,
// since it's about to be destroyed.
@@ -582,22 +588,24 @@
bool PeerConnection::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
- PortAllocatorFactoryInterface* allocator_factory,
+ rtc::scoped_ptr<cricket::PortAllocator> allocator,
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
PeerConnectionObserver* observer) {
+ TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
RTC_DCHECK(observer != nullptr);
if (!observer) {
return false;
}
observer_ = observer;
- std::vector<PortAllocatorFactoryInterface::StunConfiguration> stun_config;
- std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turn_config;
- if (!ParseIceServers(configuration.servers, &stun_config, &turn_config)) {
+ port_allocator_ = std::move(allocator);
+
+ cricket::ServerAddresses stun_servers;
+ std::vector<cricket::RelayServerConfig> turn_servers;
+ if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
return false;
}
- port_allocator_.reset(
- allocator_factory->CreatePortAllocator(stun_config, turn_config));
+ port_allocator_->SetIceServers(stun_servers, turn_servers);
// To handle both internal and externally created port allocator, we will
// enable BUNDLE here.
@@ -637,7 +645,7 @@
// Initialize the WebRtcSession. It creates transport channels etc.
if (!session_->Initialize(factory_->options(), constraints,
- dtls_identity_store.Pass(), configuration)) {
+ std::move(dtls_identity_store), configuration)) {
return false;
}
@@ -668,9 +676,8 @@
return remote_streams_;
}
-// TODO(deadbeef): Create RtpSenders immediately here, even if local
-// description hasn't yet been set.
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
+ TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
if (IsClosed()) {
return false;
}
@@ -679,25 +686,22 @@
}
local_streams_->AddStream(local_stream);
+ MediaStreamObserver* observer = new MediaStreamObserver(local_stream);
+ observer->SignalAudioTrackAdded.connect(this,
+ &PeerConnection::OnAudioTrackAdded);
+ observer->SignalAudioTrackRemoved.connect(
+ this, &PeerConnection::OnAudioTrackRemoved);
+ observer->SignalVideoTrackAdded.connect(this,
+ &PeerConnection::OnVideoTrackAdded);
+ observer->SignalVideoTrackRemoved.connect(
+ this, &PeerConnection::OnVideoTrackRemoved);
+ stream_observers_.push_back(rtc::scoped_ptr<MediaStreamObserver>(observer));
- // Find tracks that have already been configured in SDP. This can occur if a
- // local session description that contains the MSID of these tracks is set
- // before AddLocalStream is called. It can also occur if the local session
- // description is not changed and RemoveLocalStream is called and later
- // AddLocalStream is called again with the same stream.
for (const auto& track : local_stream->GetAudioTracks()) {
- const TrackInfo* track_info =
- FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
- if (track_info) {
- CreateAudioSender(local_stream, track.get(), track_info->ssrc);
- }
+ OnAudioTrackAdded(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
- const TrackInfo* track_info =
- FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
- if (track_info) {
- CreateVideoSender(local_stream, track.get(), track_info->ssrc);
- }
+ OnVideoTrackAdded(track.get(), local_stream);
}
stats_->AddStream(local_stream);
@@ -705,25 +709,24 @@
return true;
}
-// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
-// indefinitely.
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
+ TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
for (const auto& track : local_stream->GetAudioTracks()) {
- const TrackInfo* track_info =
- FindTrackInfo(local_audio_tracks_, local_stream->label(), track->id());
- if (track_info) {
- DestroyAudioSender(local_stream, track.get(), track_info->ssrc);
- }
+ OnAudioTrackRemoved(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
- const TrackInfo* track_info =
- FindTrackInfo(local_video_tracks_, local_stream->label(), track->id());
- if (track_info) {
- DestroyVideoSender(local_stream, track.get());
- }
+ OnVideoTrackRemoved(track.get(), local_stream);
}
local_streams_->RemoveStream(local_stream);
+ stream_observers_.erase(
+ std::remove_if(
+ stream_observers_.begin(), stream_observers_.end(),
+ [local_stream](const rtc::scoped_ptr<MediaStreamObserver>& observer) {
+ return observer->stream()->label().compare(local_stream->label()) ==
+ 0;
+ }),
+ stream_observers_.end());
if (IsClosed()) {
return;
@@ -733,6 +736,7 @@
rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
AudioTrackInterface* track) {
+ TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
if (!track) {
LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
return NULL;
@@ -751,6 +755,26 @@
return DtmfSenderProxy::Create(signaling_thread(), sender.get());
}
+rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
+ const std::string& kind,
+ const std::string& stream_id) {
+ TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
+ RtpSenderInterface* new_sender;
+ if (kind == MediaStreamTrackInterface::kAudioKind) {
+ new_sender = new AudioRtpSender(session_.get(), stats_.get());
+ } else if (kind == MediaStreamTrackInterface::kVideoKind) {
+ new_sender = new VideoRtpSender(session_.get());
+ } else {
+ LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
+ return rtc::scoped_refptr<RtpSenderInterface>();
+ }
+ if (!stream_id.empty()) {
+ new_sender->set_stream_id(stream_id);
+ }
+ senders_.push_back(new_sender);
+ return RtpSenderProxy::Create(signaling_thread(), new_sender);
+}
+
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
const {
std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders;
@@ -773,6 +797,7 @@
bool PeerConnection::GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track,
StatsOutputLevel level) {
+ TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK(signaling_thread()->IsCurrent());
if (!VERIFY(observer != NULL)) {
LOG(LS_ERROR) << "GetStats - observer is NULL.";
@@ -807,6 +832,7 @@
PeerConnection::CreateDataChannel(
const std::string& label,
const DataChannelInit* config) {
+ TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
bool first_datachannel = !HasDataChannels();
rtc::scoped_ptr<InternalDataChannelInit> internal_config;
@@ -830,6 +856,7 @@
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
+ TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
@@ -881,6 +908,7 @@
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
+ TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
@@ -900,6 +928,7 @@
void PeerConnection::CreateAnswer(
CreateSessionDescriptionObserver* observer,
const MediaConstraintsInterface* constraints) {
+ TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
@@ -919,6 +948,7 @@
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
+ TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
return;
@@ -940,7 +970,7 @@
// SCTP sids.
rtc::SSLRole role;
if (session_->data_channel_type() == cricket::DCT_SCTP &&
- session_->GetSslRole(&role)) {
+ session_->GetSslRole(session_->data_channel(), &role)) {
AllocateSctpSids(role);
}
@@ -949,19 +979,27 @@
const cricket::ContentInfo* audio_content =
GetFirstAudioContent(desc->description());
if (audio_content) {
- const cricket::AudioContentDescription* audio_desc =
- static_cast<const cricket::AudioContentDescription*>(
- audio_content->description);
- UpdateLocalTracks(audio_desc->streams(), audio_desc->type());
+ if (audio_content->rejected) {
+ RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
+ } else {
+ const cricket::AudioContentDescription* audio_desc =
+ static_cast<const cricket::AudioContentDescription*>(
+ audio_content->description);
+ UpdateLocalTracks(audio_desc->streams(), audio_desc->type());
+ }
}
const cricket::ContentInfo* video_content =
GetFirstVideoContent(desc->description());
if (video_content) {
- const cricket::VideoContentDescription* video_desc =
- static_cast<const cricket::VideoContentDescription*>(
- video_content->description);
- UpdateLocalTracks(video_desc->streams(), video_desc->type());
+ if (video_content->rejected) {
+ RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
+ } else {
+ const cricket::VideoContentDescription* video_desc =
+ static_cast<const cricket::VideoContentDescription*>(
+ video_content->description);
+ UpdateLocalTracks(video_desc->streams(), video_desc->type());
+ }
}
const cricket::ContentInfo* data_content =
@@ -988,6 +1026,7 @@
void PeerConnection::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
+ TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
if (!VERIFY(observer != nullptr)) {
LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
return;
@@ -1009,11 +1048,27 @@
// SCTP sids.
rtc::SSLRole role;
if (session_->data_channel_type() == cricket::DCT_SCTP &&
- session_->GetSslRole(&role)) {
+ session_->GetSslRole(session_->data_channel(), &role)) {
AllocateSctpSids(role);
}
const cricket::SessionDescription* remote_desc = desc->description();
+ const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
+ const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
+ const cricket::AudioContentDescription* audio_desc =
+ GetFirstAudioContentDescription(remote_desc);
+ const cricket::VideoContentDescription* video_desc =
+ GetFirstVideoContentDescription(remote_desc);
+ const cricket::DataContentDescription* data_desc =
+ GetFirstDataContentDescription(remote_desc);
+
+ // Check if the descriptions include streams, just in case the peer supports
+ // MSID, but doesn't indicate so with "a=msid-semantic".
+ if (remote_desc->msid_supported() ||
+ (audio_desc && !audio_desc->streams().empty()) ||
+ (video_desc && !video_desc->streams().empty())) {
+ remote_peer_supports_msid_ = true;
+ }
// We wait to signal new streams until we finish processing the description,
// since only at that point will new streams have all their tracks.
@@ -1021,39 +1076,39 @@
// Find all audio rtp streams and create corresponding remote AudioTracks
// and MediaStreams.
- const cricket::ContentInfo* audio_content = GetFirstAudioContent(remote_desc);
if (audio_content) {
- const cricket::AudioContentDescription* desc =
- static_cast<const cricket::AudioContentDescription*>(
- audio_content->description);
- UpdateRemoteStreamsList(GetActiveStreams(desc), desc->type(), new_streams);
- remote_info_.default_audio_track_needed =
- !remote_desc->msid_supported() && desc->streams().empty() &&
- MediaContentDirectionHasSend(desc->direction());
+ if (audio_content->rejected) {
+ RemoveTracks(cricket::MEDIA_TYPE_AUDIO);
+ } else {
+ bool default_audio_track_needed =
+ !remote_peer_supports_msid_ &&
+ MediaContentDirectionHasSend(audio_desc->direction());
+ UpdateRemoteStreamsList(GetActiveStreams(audio_desc),
+ default_audio_track_needed, audio_desc->type(),
+ new_streams);
+ }
}
// Find all video rtp streams and create corresponding remote VideoTracks
// and MediaStreams.
- const cricket::ContentInfo* video_content = GetFirstVideoContent(remote_desc);
if (video_content) {
- const cricket::VideoContentDescription* desc =
- static_cast<const cricket::VideoContentDescription*>(
- video_content->description);
- UpdateRemoteStreamsList(GetActiveStreams(desc), desc->type(), new_streams);
- remote_info_.default_video_track_needed =
- !remote_desc->msid_supported() && desc->streams().empty() &&
- MediaContentDirectionHasSend(desc->direction());
+ if (video_content->rejected) {
+ RemoveTracks(cricket::MEDIA_TYPE_VIDEO);
+ } else {
+ bool default_video_track_needed =
+ !remote_peer_supports_msid_ &&
+ MediaContentDirectionHasSend(video_desc->direction());
+ UpdateRemoteStreamsList(GetActiveStreams(video_desc),
+ default_video_track_needed, video_desc->type(),
+ new_streams);
+ }
}
// Update the DataChannels with the information from the remote peer.
- const cricket::ContentInfo* data_content = GetFirstDataContent(remote_desc);
- if (data_content) {
- const cricket::DataContentDescription* desc =
- static_cast<const cricket::DataContentDescription*>(
- data_content->description);
- if (rtc::starts_with(desc->protocol().data(),
+ if (data_desc) {
+ if (rtc::starts_with(data_desc->protocol().data(),
cricket::kMediaProtocolRtpPrefix)) {
- UpdateRemoteRtpDataChannels(GetActiveStreams(desc));
+ UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
}
}
@@ -1064,58 +1119,21 @@
observer_->OnAddStream(new_stream);
}
- // Find removed MediaStreams.
- if (remote_info_.IsDefaultMediaStreamNeeded() &&
- remote_streams_->find(kDefaultStreamLabel) != nullptr) {
- // The default media stream already exists. No need to do anything.
- } else {
- UpdateEndedRemoteMediaStreams();
- remote_info_.msid_supported |= remote_streams_->count() > 0;
- }
- MaybeCreateDefaultStream();
+ UpdateEndedRemoteMediaStreams();
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
signaling_thread()->Post(this, MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
}
bool PeerConnection::SetConfiguration(const RTCConfiguration& config) {
+ TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (port_allocator_) {
- std::vector<PortAllocatorFactoryInterface::StunConfiguration> stuns;
- std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turns;
- if (!ParseIceServers(config.servers, &stuns, &turns)) {
+ cricket::ServerAddresses stun_servers;
+ std::vector<cricket::RelayServerConfig> turn_servers;
+ if (!ParseIceServers(config.servers, &stun_servers, &turn_servers)) {
return false;
}
-
- std::vector<rtc::SocketAddress> stun_hosts;
- typedef std::vector<StunConfiguration>::const_iterator StunIt;
- for (StunIt stun_it = stuns.begin(); stun_it != stuns.end(); ++stun_it) {
- stun_hosts.push_back(stun_it->server);
- }
-
- rtc::SocketAddress stun_addr;
- if (!stun_hosts.empty()) {
- stun_addr = stun_hosts.front();
- LOG(LS_INFO) << "SetConfiguration: StunServer Address: "
- << stun_addr.ToString();
- }
-
- for (size_t i = 0; i < turns.size(); ++i) {
- cricket::RelayCredentials credentials(turns[i].username,
- turns[i].password);
- cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
- cricket::ProtocolType protocol;
- if (cricket::StringToProto(turns[i].transport_type.c_str(), &protocol)) {
- relay_server.ports.push_back(cricket::ProtocolAddress(
- turns[i].server, protocol, turns[i].secure));
- relay_server.credentials = credentials;
- LOG(LS_INFO) << "SetConfiguration: TurnServer Address: "
- << turns[i].server.ToString();
- } else {
- LOG(LS_WARNING) << "Ignoring TURN server " << turns[i].server << ". "
- << "Reason= Incorrect " << turns[i].transport_type
- << " transport parameter.";
- }
- }
+ port_allocator_->SetIceServers(stun_servers, turn_servers);
}
session_->SetIceConfig(session_->ParseIceConfig(config));
return session_->SetIceTransports(config.type);
@@ -1123,10 +1141,12 @@
bool PeerConnection::AddIceCandidate(
const IceCandidateInterface* ice_candidate) {
+ TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
return session_->ProcessIceMessage(ice_candidate);
}
void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
+ TRACE_EVENT0("webrtc", "PeerConnection::RegisterUmaObserver");
uma_observer_ = observer;
if (session_) {
@@ -1156,6 +1176,7 @@
}
void PeerConnection::Close() {
+ TRACE_EVENT0("webrtc", "PeerConnection::Close");
// Update stats here so that we have the most recent stats for tracks and
// streams before the channels are closed.
stats_->UpdateStats(kStatsOutputLevelStandard);
@@ -1223,6 +1244,10 @@
delete param;
break;
}
+ case MSG_FREE_DATACHANNELS: {
+ sctp_data_channels_to_free_.clear();
+ break;
+ }
default:
RTC_DCHECK(false && "Not implemented");
break;
@@ -1267,49 +1292,6 @@
}
}
-void PeerConnection::CreateAudioSender(MediaStreamInterface* stream,
- AudioTrackInterface* audio_track,
- uint32_t ssrc) {
- senders_.push_back(new AudioRtpSender(audio_track, ssrc, session_.get()));
- stats_->AddLocalAudioTrack(audio_track, ssrc);
-}
-
-void PeerConnection::CreateVideoSender(MediaStreamInterface* stream,
- VideoTrackInterface* video_track,
- uint32_t ssrc) {
- senders_.push_back(new VideoRtpSender(video_track, ssrc, session_.get()));
-}
-
-// TODO(deadbeef): Keep RtpSenders around even if track goes away in local
-// description.
-void PeerConnection::DestroyAudioSender(MediaStreamInterface* stream,
- AudioTrackInterface* audio_track,
- uint32_t ssrc) {
- auto it = FindSenderForTrack(audio_track);
- if (it == senders_.end()) {
- LOG(LS_WARNING) << "RtpSender for track with id " << audio_track->id()
- << " doesn't exist.";
- return;
- } else {
- (*it)->Stop();
- senders_.erase(it);
- }
- stats_->RemoveLocalAudioTrack(audio_track, ssrc);
-}
-
-void PeerConnection::DestroyVideoSender(MediaStreamInterface* stream,
- VideoTrackInterface* video_track) {
- auto it = FindSenderForTrack(video_track);
- if (it == senders_.end()) {
- LOG(LS_WARNING) << "RtpSender for track with id " << video_track->id()
- << " doesn't exist.";
- return;
- } else {
- (*it)->Stop();
- senders_.erase(it);
- }
-}
-
void PeerConnection::OnIceConnectionChange(
PeerConnectionInterface::IceConnectionState new_state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
@@ -1362,6 +1344,80 @@
observer_->OnStateChange(PeerConnectionObserver::kSignalingState);
}
+void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
+ MediaStreamInterface* stream) {
+ auto sender = FindSenderForTrack(track);
+ if (sender != senders_.end()) {
+ // We already have a sender for this track, so just change the stream_id
+ // so that it's correct in the next call to CreateOffer.
+ (*sender)->set_stream_id(stream->label());
+ return;
+ }
+
+ // Normal case; we've never seen this track before.
+ AudioRtpSender* new_sender =
+ new AudioRtpSender(track, stream->label(), session_.get(), stats_.get());
+ senders_.push_back(new_sender);
+ // If the sender has already been configured in SDP, we call SetSsrc,
+ // which will connect the sender to the underlying transport. This can
+ // occur if a local session description that contains the ID of the sender
+ // is set before AddStream is called. It can also occur if the local
+ // session description is not changed and RemoveStream is called, and
+ // later AddStream is called again with the same stream.
+ const TrackInfo* track_info =
+ FindTrackInfo(local_audio_tracks_, stream->label(), track->id());
+ if (track_info) {
+ new_sender->SetSsrc(track_info->ssrc);
+ }
+}
+
+// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
+// indefinitely, when we have unified plan SDP.
+void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
+ MediaStreamInterface* stream) {
+ auto sender = FindSenderForTrack(track);
+ if (sender == senders_.end()) {
+ LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
+ << " doesn't exist.";
+ return;
+ }
+ (*sender)->Stop();
+ senders_.erase(sender);
+}
+
+void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
+ MediaStreamInterface* stream) {
+ auto sender = FindSenderForTrack(track);
+ if (sender != senders_.end()) {
+ // We already have a sender for this track, so just change the stream_id
+ // so that it's correct in the next call to CreateOffer.
+ (*sender)->set_stream_id(stream->label());
+ return;
+ }
+
+ // Normal case; we've never seen this track before.
+ VideoRtpSender* new_sender =
+ new VideoRtpSender(track, stream->label(), session_.get());
+ senders_.push_back(new_sender);
+ const TrackInfo* track_info =
+ FindTrackInfo(local_video_tracks_, stream->label(), track->id());
+ if (track_info) {
+ new_sender->SetSsrc(track_info->ssrc);
+ }
+}
+
+void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
+ MediaStreamInterface* stream) {
+ auto sender = FindSenderForTrack(track);
+ if (sender == senders_.end()) {
+ LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
+ << " doesn't exist.";
+ return;
+ }
+ (*sender)->Stop();
+ senders_.erase(sender);
+}
+
void PeerConnection::PostSetSessionDescriptionFailure(
SetSessionDescriptionObserver* observer,
const std::string& error) {
@@ -1385,7 +1441,7 @@
return false;
}
- SetStreams(session_options, local_streams_, rtp_data_channels_);
+ AddSendStreams(session_options, senders_, rtp_data_channels_);
// Offer to receive audio/video if the constraint is not set and there are
// send streams, or we're currently receiving.
if (rtc_options.offer_to_receive_audio == RTCOfferAnswerOptions::kUndefined) {
@@ -1418,7 +1474,7 @@
return false;
}
- SetStreams(session_options, local_streams_, rtp_data_channels_);
+ AddSendStreams(session_options, senders_, rtp_data_channels_);
session_options->bundle_enabled =
session_options->bundle_enabled &&
(session_options->has_audio() || session_options->has_video() ||
@@ -1433,25 +1489,34 @@
return true;
}
+void PeerConnection::RemoveTracks(cricket::MediaType media_type) {
+ UpdateLocalTracks(std::vector<cricket::StreamParams>(), media_type);
+ UpdateRemoteStreamsList(std::vector<cricket::StreamParams>(), false,
+ media_type, nullptr);
+}
+
void PeerConnection::UpdateRemoteStreamsList(
const cricket::StreamParamsVec& streams,
+ bool default_track_needed,
cricket::MediaType media_type,
StreamCollection* new_streams) {
TrackInfos* current_tracks = GetRemoteTracks(media_type);
// Find removed tracks. I.e., tracks where the track id or ssrc don't match
- // the
- // new StreamParam.
+ // the new StreamParam.
auto track_it = current_tracks->begin();
while (track_it != current_tracks->end()) {
const TrackInfo& info = *track_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.ssrc);
- if (!params || params->id != info.track_id) {
+ bool track_exists = params && params->id == info.track_id;
+ // If this is a default track, and we still need it, don't remove it.
+ if ((info.stream_label == kDefaultStreamLabel && default_track_needed) ||
+ track_exists) {
+ ++track_it;
+ } else {
OnRemoteTrackRemoved(info.stream_label, info.track_id, media_type);
track_it = current_tracks->erase(track_it);
- } else {
- ++track_it;
}
}
@@ -1479,6 +1544,29 @@
OnRemoteTrackSeen(stream_label, track_id, ssrc, media_type);
}
}
+
+ // Add default track if necessary.
+ if (default_track_needed) {
+ rtc::scoped_refptr<MediaStreamInterface> default_stream =
+ remote_streams_->find(kDefaultStreamLabel);
+ if (!default_stream) {
+ // Create the new default MediaStream.
+ default_stream =
+ remote_stream_factory_->CreateMediaStream(kDefaultStreamLabel);
+ remote_streams_->AddStream(default_stream);
+ new_streams->AddStream(default_stream);
+ }
+ std::string default_track_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
+ ? kDefaultAudioTrackLabel
+ : kDefaultVideoTrackLabel;
+ const TrackInfo* default_track_info =
+ FindTrackInfo(*current_tracks, kDefaultStreamLabel, default_track_id);
+ if (!default_track_info) {
+ current_tracks->push_back(
+ TrackInfo(kDefaultStreamLabel, default_track_id, 0));
+ OnRemoteTrackSeen(kDefaultStreamLabel, default_track_id, 0, media_type);
+ }
+ }
}
void PeerConnection::OnRemoteTrackSeen(const std::string& stream_label,
@@ -1488,8 +1576,8 @@
MediaStreamInterface* stream = remote_streams_->find(stream_label);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
- AudioTrackInterface* audio_track =
- remote_stream_factory_->AddAudioTrack(stream, track_id);
+ AudioTrackInterface* audio_track = remote_stream_factory_->AddAudioTrack(
+ ssrc, session_.get(), stream, track_id);
CreateAudioReceiver(stream, audio_track, ssrc);
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
VideoTrackInterface* video_track =
@@ -1541,41 +1629,6 @@
}
}
-void PeerConnection::MaybeCreateDefaultStream() {
- if (!remote_info_.IsDefaultMediaStreamNeeded()) {
- return;
- }
-
- bool default_created = false;
-
- rtc::scoped_refptr<MediaStreamInterface> default_remote_stream =
- remote_streams_->find(kDefaultStreamLabel);
- if (default_remote_stream == nullptr) {
- default_created = true;
- default_remote_stream =
- remote_stream_factory_->CreateMediaStream(kDefaultStreamLabel);
- remote_streams_->AddStream(default_remote_stream);
- }
- if (remote_info_.default_audio_track_needed &&
- default_remote_stream->GetAudioTracks().size() == 0) {
- remote_audio_tracks_.push_back(
- TrackInfo(kDefaultStreamLabel, kDefaultAudioTrackLabel, 0));
- OnRemoteTrackSeen(kDefaultStreamLabel, kDefaultAudioTrackLabel, 0,
- cricket::MEDIA_TYPE_AUDIO);
- }
- if (remote_info_.default_video_track_needed &&
- default_remote_stream->GetVideoTracks().size() == 0) {
- remote_video_tracks_.push_back(
- TrackInfo(kDefaultStreamLabel, kDefaultVideoTrackLabel, 0));
- OnRemoteTrackSeen(kDefaultStreamLabel, kDefaultVideoTrackLabel, 0,
- cricket::MEDIA_TYPE_VIDEO);
- }
- if (default_created) {
- stats_->AddStream(default_remote_stream);
- observer_->OnAddStream(default_remote_stream);
- }
-}
-
void PeerConnection::EndRemoteTracks(cricket::MediaType media_type) {
TrackInfos* current_tracks = GetRemoteTracks(media_type);
for (TrackInfos::iterator track_it = current_tracks->begin();
@@ -1643,62 +1696,44 @@
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
- MediaStreamInterface* stream = local_streams_->find(stream_label);
- if (!stream) {
- LOG(LS_WARNING) << "An unknown local MediaStream with label "
- << stream_label << " has been configured.";
+ RtpSenderInterface* sender = FindSenderById(track_id);
+ if (!sender) {
+ LOG(LS_WARNING) << "An unknown RtpSender with id " << track_id
+ << " has been configured in the local description.";
return;
}
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
- AudioTrackInterface* audio_track = stream->FindAudioTrack(track_id);
- if (!audio_track) {
- LOG(LS_WARNING) << "An unknown local AudioTrack with id , " << track_id
- << " has been configured.";
- return;
- }
- CreateAudioSender(stream, audio_track, ssrc);
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
- VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
- if (!video_track) {
- LOG(LS_WARNING) << "An unknown local VideoTrack with id , " << track_id
- << " has been configured.";
- return;
- }
- CreateVideoSender(stream, video_track, ssrc);
- } else {
- RTC_DCHECK(false && "Invalid media type");
+ if (sender->media_type() != media_type) {
+ LOG(LS_WARNING) << "An RtpSender has been configured in the local"
+ << " description with an unexpected media type.";
+ return;
}
+
+ sender->set_stream_id(stream_label);
+ sender->SetSsrc(ssrc);
}
void PeerConnection::OnLocalTrackRemoved(const std::string& stream_label,
const std::string& track_id,
uint32_t ssrc,
cricket::MediaType media_type) {
- MediaStreamInterface* stream = local_streams_->find(stream_label);
- if (!stream) {
- // This is the normal case. I.e., RemoveLocalStream has been called and the
+ RtpSenderInterface* sender = FindSenderById(track_id);
+ if (!sender) {
+ // This is the normal case. I.e., RemoveStream has been called and the
// SessionDescriptions has been renegotiated.
return;
}
- // A track has been removed from the SessionDescription but the MediaStream
- // is still associated with PeerConnection. This only occurs if the SDP
- // doesn't match with the calls to AddLocalStream and RemoveLocalStream.
- if (media_type == cricket::MEDIA_TYPE_AUDIO) {
- AudioTrackInterface* audio_track = stream->FindAudioTrack(track_id);
- if (!audio_track) {
- return;
- }
- DestroyAudioSender(stream, audio_track, ssrc);
- } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
- VideoTrackInterface* video_track = stream->FindVideoTrack(track_id);
- if (!video_track) {
- return;
- }
- DestroyVideoSender(stream, video_track);
- } else {
- RTC_DCHECK(false && "Invalid media type.");
+
+ // A sender has been removed from the SessionDescription but it's still
+ // associated with the PeerConnection. This only occurs if the SDP doesn't
+ // match with the calls to CreateSender, AddStream and RemoveStream.
+ if (sender->media_type() != media_type) {
+ LOG(LS_WARNING) << "An RtpSender has been configured in the local"
+ << " description with an unexpected media type.";
+ return;
}
+
+ sender->SetSsrc(0);
}
void PeerConnection::UpdateLocalRtpDataChannels(
@@ -1806,7 +1841,7 @@
if (session_->data_channel_type() == cricket::DCT_SCTP) {
if (new_config.id < 0) {
rtc::SSLRole role;
- if (session_->GetSslRole(&role) &&
+ if ((session_->GetSslRole(session_->data_channel(), &role)) &&
!sid_allocator_.AllocateSid(role, &new_config.id)) {
LOG(LS_ERROR) << "No id can be allocated for the SCTP data channel.";
return nullptr;
@@ -1860,13 +1895,18 @@
}
void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
+ RTC_DCHECK(signaling_thread()->IsCurrent());
for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end();
++it) {
if (it->get() == channel) {
if (channel->id() >= 0) {
sid_allocator_.ReleaseSid(channel->id());
}
+ // Since this method is triggered by a signal from the DataChannel,
+ // we can't free it directly here; we need to free it asynchronously.
+ sctp_data_channels_to_free_.push_back(*it);
sctp_data_channels_.erase(it);
+ signaling_thread()->Post(this, MSG_FREE_DATACHANNELS, nullptr);
return;
}
}
@@ -1916,6 +1956,15 @@
DataChannelProxy::Create(signaling_thread(), channel));
}
+RtpSenderInterface* PeerConnection::FindSenderById(const std::string& id) {
+ auto it =
+ std::find_if(senders_.begin(), senders_.end(),
+ [id](const rtc::scoped_refptr<RtpSenderInterface>& sender) {
+ return sender->id() == id;
+ });
+ return it != senders_.end() ? it->get() : nullptr;
+}
+
std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) {
return std::find_if(
diff --git a/talk/app/webrtc/peerconnection.h b/talk/app/webrtc/peerconnection.h
index 2d388ae..6e2b967 100644
--- a/talk/app/webrtc/peerconnection.h
+++ b/talk/app/webrtc/peerconnection.h
@@ -42,13 +42,9 @@
namespace webrtc {
+class MediaStreamObserver;
class RemoteMediaStreamFactory;
-typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration>
- StunConfigurations;
-typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration>
- TurnConfigurations;
-
// Populates |session_options| from |rtc_options|, and returns true if options
// are valid.
bool ConvertRtcOptionsForOffer(
@@ -60,11 +56,11 @@
bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* session_options);
-// Parses the URLs for each server in |servers| to build |stun_config| and
-// |turn_config|.
+// Parses the URLs for each server in |servers| to build |stun_servers| and
+// |turn_servers|.
bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
- StunConfigurations* stun_config,
- TurnConfigurations* turn_config);
+ cricket::ServerAddresses* stun_servers,
+ std::vector<cricket::RelayServerConfig>* turn_servers);
// PeerConnection implements the PeerConnectionInterface interface.
// It uses WebRtcSession to implement the PeerConnection functionality.
@@ -78,9 +74,10 @@
bool Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
- PortAllocatorFactoryInterface* allocator_factory,
+ rtc::scoped_ptr<cricket::PortAllocator> allocator,
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
PeerConnectionObserver* observer);
+
rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
bool AddStream(MediaStreamInterface* local_stream) override;
@@ -91,6 +88,10 @@
rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
AudioTrackInterface* track) override;
+ rtc::scoped_refptr<RtpSenderInterface> CreateSender(
+ const std::string& kind,
+ const std::string& stream_id) override;
+
std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const override;
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
@@ -148,32 +149,16 @@
const std::string track_id,
uint32_t ssrc)
: stream_label(stream_label), track_id(track_id), ssrc(ssrc) {}
+ bool operator==(const TrackInfo& other) {
+ return this->stream_label == other.stream_label &&
+ this->track_id == other.track_id && this->ssrc == other.ssrc;
+ }
std::string stream_label;
std::string track_id;
uint32_t ssrc;
};
typedef std::vector<TrackInfo> TrackInfos;
- struct RemotePeerInfo {
- RemotePeerInfo()
- : msid_supported(false),
- default_audio_track_needed(false),
- default_video_track_needed(false) {}
- // True if it has been discovered that the remote peer support MSID.
- bool msid_supported;
- // The remote peer indicates in the session description that audio will be
- // sent but no MSID is given.
- bool default_audio_track_needed;
- // The remote peer indicates in the session description that video will be
- // sent but no MSID is given.
- bool default_video_track_needed;
-
- bool IsDefaultMediaStreamNeeded() {
- return !msid_supported &&
- (default_audio_track_needed || default_video_track_needed);
- }
- };
-
// Implements MessageHandler.
void OnMessage(rtc::Message* msg) override;
@@ -187,12 +172,6 @@
AudioTrackInterface* audio_track);
void DestroyVideoReceiver(MediaStreamInterface* stream,
VideoTrackInterface* video_track);
- void CreateAudioSender(MediaStreamInterface* stream,
- AudioTrackInterface* audio_track,
- uint32_t ssrc);
- void CreateVideoSender(MediaStreamInterface* stream,
- VideoTrackInterface* video_track,
- uint32_t ssrc);
void DestroyAudioSender(MediaStreamInterface* stream,
AudioTrackInterface* audio_track,
uint32_t ssrc);
@@ -210,6 +189,16 @@
void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
void ChangeSignalingState(SignalingState signaling_state);
+ // Signals from MediaStreamObserver.
+ void OnAudioTrackAdded(AudioTrackInterface* track,
+ MediaStreamInterface* stream);
+ void OnAudioTrackRemoved(AudioTrackInterface* track,
+ MediaStreamInterface* stream);
+ void OnVideoTrackAdded(VideoTrackInterface* track,
+ MediaStreamInterface* stream);
+ void OnVideoTrackRemoved(VideoTrackInterface* track,
+ MediaStreamInterface* stream);
+
rtc::Thread* signaling_thread() const {
return factory_->signaling_thread();
}
@@ -236,12 +225,19 @@
const MediaConstraintsInterface* constraints,
cricket::MediaSessionOptions* session_options);
- // Makes sure a MediaStream Track is created for each StreamParam in
- // |streams|. |media_type| is the type of the |streams| and can be either
- // audio or video.
+ // Remove all local and remote tracks of type |media_type|.
+ // Called when a media type is rejected (m-line set to port 0).
+ void RemoveTracks(cricket::MediaType media_type);
+
+ // Makes sure a MediaStreamTrack is created for each StreamParam in |streams|,
+ // and existing MediaStreamTracks are removed if there is no corresponding
+ // StreamParam. If |default_track_needed| is true, a default MediaStreamTrack
+ // is created if it doesn't exist; if false, it's removed if it exists.
+ // |media_type| is the type of the |streams| and can be either audio or video.
// If a new MediaStream is created it is added to |new_streams|.
void UpdateRemoteStreamsList(
const std::vector<cricket::StreamParams>& streams,
+ bool default_track_needed,
cricket::MediaType media_type,
StreamCollection* new_streams);
@@ -265,8 +261,6 @@
// exist.
void UpdateEndedRemoteMediaStreams();
- void MaybeCreateDefaultStream();
-
// Set the MediaStreamTrackInterface::TrackState to |kEnded| on all remote
// tracks of type |media_type|.
void EndRemoteTracks(cricket::MediaType media_type);
@@ -328,6 +322,8 @@
void OnDataChannelOpenMessage(const std::string& label,
const InternalDataChannelInit& config);
+ RtpSenderInterface* FindSenderById(const std::string& id);
+
std::vector<rtc::scoped_refptr<RtpSenderInterface>>::iterator
FindSenderForTrack(MediaStreamTrackInterface* track);
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>::iterator
@@ -366,6 +362,8 @@
// Streams created as a result of SetRemoteDescription.
rtc::scoped_refptr<StreamCollection> remote_streams_;
+ std::vector<rtc::scoped_ptr<MediaStreamObserver>> stream_observers_;
+
// These lists store track info seen in local/remote descriptions.
TrackInfos remote_audio_tracks_;
TrackInfos remote_video_tracks_;
@@ -376,8 +374,9 @@
// label -> DataChannel
std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_;
std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_;
+ std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_;
- RemotePeerInfo remote_info_;
+ bool remote_peer_supports_msid_ = false;
rtc::scoped_ptr<RemoteMediaStreamFactory> remote_stream_factory_;
std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders_;
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index 3cf66d6..8d0793e 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -30,11 +30,11 @@
#include <algorithm>
#include <list>
#include <map>
+#include <utility>
#include <vector>
#include "talk/app/webrtc/dtmfsender.h"
#include "talk/app/webrtc/fakemetricsobserver.h"
-#include "talk/app/webrtc/fakeportallocatorfactory.h"
#include "talk/app/webrtc/localaudiosource.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/peerconnection.h"
@@ -58,6 +58,7 @@
#include "webrtc/base/virtualsocketserver.h"
#include "webrtc/p2p/base/constants.h"
#include "webrtc/p2p/base/sessiondescription.h"
+#include "webrtc/p2p/client/fakeportallocator.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \
@@ -78,11 +79,13 @@
using webrtc::DtmfSenderObserverInterface;
using webrtc::FakeConstraints;
using webrtc::MediaConstraintsInterface;
+using webrtc::MediaStreamInterface;
using webrtc::MediaStreamTrackInterface;
using webrtc::MockCreateSessionDescriptionObserver;
using webrtc::MockDataChannelObserver;
using webrtc::MockSetSessionDescriptionObserver;
using webrtc::MockStatsObserver;
+using webrtc::ObserverInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionFactory;
using webrtc::SessionDescriptionInterface;
@@ -96,6 +99,7 @@
#if !defined(THREAD_SANITIZER)
static const int kMaxWaitForStatsMs = 3000;
#endif
+static const int kMaxWaitForActivationMs = 5000;
static const int kMaxWaitForFramesMs = 10000;
static const int kEndAudioFrameCount = 3;
static const int kEndVideoFrameCount = 3;
@@ -111,7 +115,7 @@
#if !defined(THREAD_SANITIZER)
// SRTP cipher name negotiated by the tests. This must be updated if the
// default changes.
-static const char kDefaultSrtpCipher[] = "AES_CM_128_HMAC_SHA1_32";
+static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32;
#endif
static void RemoveLinesFromSdp(const std::string& line_start,
@@ -139,26 +143,35 @@
};
class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
- public SignalingMessageReceiver {
+ public SignalingMessageReceiver,
+ public ObserverInterface {
public:
- static PeerConnectionTestClient* CreateClient(
+ static PeerConnectionTestClient* CreateClientWithDtlsIdentityStore(
const std::string& id,
const MediaConstraintsInterface* constraints,
- const PeerConnectionFactory::Options* options) {
+ const PeerConnectionFactory::Options* options,
+ rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
PeerConnectionTestClient* client(new PeerConnectionTestClient(id));
- if (!client->Init(constraints, options)) {
+ if (!client->Init(constraints, options, std::move(dtls_identity_store))) {
delete client;
return nullptr;
}
return client;
}
+ static PeerConnectionTestClient* CreateClient(
+ const std::string& id,
+ const MediaConstraintsInterface* constraints,
+ const PeerConnectionFactory::Options* options) {
+ rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
+ rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
+ : nullptr);
+
+ return CreateClientWithDtlsIdentityStore(id, constraints, options,
+ std::move(dtls_identity_store));
+ }
+
~PeerConnectionTestClient() {
- while (!fake_video_renderers_.empty()) {
- RenderMap::iterator it = fake_video_renderers_.begin();
- delete it->second;
- fake_video_renderers_.erase(it);
- }
}
void Negotiate() { Negotiate(true, true); }
@@ -206,16 +219,17 @@
webrtc::PeerConnectionInterface::SignalingState new_state) override {
EXPECT_EQ(pc()->signaling_state(), new_state);
}
- void OnAddStream(webrtc::MediaStreamInterface* media_stream) override {
+ void OnAddStream(MediaStreamInterface* media_stream) override {
+ media_stream->RegisterObserver(this);
for (size_t i = 0; i < media_stream->GetVideoTracks().size(); ++i) {
const std::string id = media_stream->GetVideoTracks()[i]->id();
ASSERT_TRUE(fake_video_renderers_.find(id) ==
fake_video_renderers_.end());
- fake_video_renderers_[id] =
- new webrtc::FakeVideoTrackRenderer(media_stream->GetVideoTracks()[i]);
+ fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
+ media_stream->GetVideoTracks()[i]));
}
}
- void OnRemoveStream(webrtc::MediaStreamInterface* media_stream) override {}
+ void OnRemoveStream(MediaStreamInterface* media_stream) override {}
void OnRenegotiationNeeded() override {}
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
@@ -238,6 +252,40 @@
candidate->sdp_mid(), candidate->sdp_mline_index(), ice_sdp);
}
+ // MediaStreamInterface callback
+ void OnChanged() override {
+ // Track added or removed from MediaStream, so update our renderers.
+ rtc::scoped_refptr<StreamCollectionInterface> remote_streams =
+ pc()->remote_streams();
+ // Remove renderers for tracks that were removed.
+ for (auto it = fake_video_renderers_.begin();
+ it != fake_video_renderers_.end();) {
+ if (remote_streams->FindVideoTrack(it->first) == nullptr) {
+ auto to_remove = it++;
+ removed_fake_video_renderers_.push_back(std::move(to_remove->second));
+ fake_video_renderers_.erase(to_remove);
+ } else {
+ ++it;
+ }
+ }
+ // Create renderers for new video tracks.
+ for (size_t stream_index = 0; stream_index < remote_streams->count();
+ ++stream_index) {
+ MediaStreamInterface* remote_stream = remote_streams->at(stream_index);
+ for (size_t track_index = 0;
+ track_index < remote_stream->GetVideoTracks().size();
+ ++track_index) {
+ const std::string id =
+ remote_stream->GetVideoTracks()[track_index]->id();
+ if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) {
+ continue;
+ }
+ fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
+ remote_stream->GetVideoTracks()[track_index]));
+ }
+ }
+ }
+
void SetVideoConstraints(const webrtc::FakeConstraints& video_constraint) {
video_constraints_ = video_constraint;
}
@@ -246,22 +294,11 @@
std::string stream_label =
kStreamLabelBase +
rtc::ToString<int>(static_cast<int>(pc()->local_streams()->count()));
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+ rtc::scoped_refptr<MediaStreamInterface> stream =
peer_connection_factory_->CreateLocalMediaStream(stream_label);
if (audio && can_receive_audio()) {
- FakeConstraints constraints;
- // Disable highpass filter so that we can get all the test audio frames.
- constraints.AddMandatory(
- MediaConstraintsInterface::kHighpassFilter, false);
- rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
- peer_connection_factory_->CreateAudioSource(&constraints);
- // TODO(perkj): Test audio source when it is implemented. Currently audio
- // always use the default input.
- std::string label = stream_label + kAudioTrackLabelBase;
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
- peer_connection_factory_->CreateAudioTrack(label, source));
- stream->AddTrack(audio_track);
+ stream->AddTrack(CreateLocalAudioTrack(stream_label));
}
if (video && can_receive_video()) {
stream->AddTrack(CreateLocalVideoTrack(stream_label));
@@ -276,6 +313,12 @@
return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
}
+ // Automatically add a stream when receiving an offer, if we don't have one.
+ // Defaults to true.
+ void set_auto_add_stream(bool auto_add_stream) {
+ auto_add_stream_ = auto_add_stream;
+ }
+
void set_signaling_message_receiver(
SignalingMessageReceiver* signaling_message_receiver) {
signaling_message_receiver_ = signaling_message_receiver;
@@ -357,6 +400,35 @@
data_observer_.reset(new MockDataChannelObserver(data_channel_));
}
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
+ const std::string& stream_label) {
+ FakeConstraints constraints;
+ // Disable highpass filter so that we can get all the test audio frames.
+ constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
+ rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+ peer_connection_factory_->CreateAudioSource(&constraints);
+ // TODO(perkj): Test audio source when it is implemented. Currently audio
+ // always use the default input.
+ std::string label = stream_label + kAudioTrackLabelBase;
+ return peer_connection_factory_->CreateAudioTrack(label, source);
+ }
+
+ rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
+ const std::string& stream_label) {
+ // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
+ FakeConstraints source_constraints = video_constraints_;
+ source_constraints.SetMandatoryMaxFrameRate(10);
+
+ cricket::FakeVideoCapturer* fake_capturer =
+ new webrtc::FakePeriodicVideoCapturer();
+ video_capturers_.push_back(fake_capturer);
+ rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
+ peer_connection_factory_->CreateVideoSource(fake_capturer,
+ &source_constraints);
+ std::string label = stream_label + kVideoTrackLabelBase;
+ return peer_connection_factory_->CreateVideoTrack(label, source);
+ }
+
DataChannelInterface* data_channel() { return data_channel_; }
const MockDataChannelObserver* data_observer() const {
return data_observer_.get();
@@ -376,6 +448,10 @@
return number_of_frames <= fake_audio_capture_module_->frames_received();
}
+ int audio_frames_received() const {
+ return fake_audio_capture_module_->frames_received();
+ }
+
bool VideoFramesReceivedCheck(int number_of_frames) {
if (video_decoder_factory_enabled_) {
const std::vector<FakeWebRtcVideoDecoder*>& decoders
@@ -384,9 +460,8 @@
return number_of_frames <= 0;
}
- for (std::vector<FakeWebRtcVideoDecoder*>::const_iterator
- it = decoders.begin(); it != decoders.end(); ++it) {
- if (number_of_frames > (*it)->GetNumFramesReceived()) {
+ for (FakeWebRtcVideoDecoder* decoder : decoders) {
+ if (number_of_frames > decoder->GetNumFramesReceived()) {
return false;
}
}
@@ -396,9 +471,8 @@
return number_of_frames <= 0;
}
- for (RenderMap::const_iterator it = fake_video_renderers_.begin();
- it != fake_video_renderers_.end(); ++it) {
- if (number_of_frames > it->second->num_rendered_frames()) {
+ for (const auto& pair : fake_video_renderers_) {
+ if (number_of_frames > pair.second->num_rendered_frames()) {
return false;
}
}
@@ -406,6 +480,25 @@
}
}
+ int video_frames_received() const {
+ int total = 0;
+ if (video_decoder_factory_enabled_) {
+ const std::vector<FakeWebRtcVideoDecoder*>& decoders =
+ fake_video_decoder_factory_->decoders();
+ for (const FakeWebRtcVideoDecoder* decoder : decoders) {
+ total += decoder->GetNumFramesReceived();
+ }
+ } else {
+ for (const auto& pair : fake_video_renderers_) {
+ total += pair.second->num_rendered_frames();
+ }
+ for (const auto& renderer : removed_fake_video_renderers_) {
+ total += renderer->num_rendered_frames();
+ }
+ }
+ return total;
+ }
+
// Verify the CreateDtmfSender interface
void VerifyDtmf() {
rtc::scoped_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver());
@@ -641,14 +734,14 @@
explicit PeerConnectionTestClient(const std::string& id) : id_(id) {}
- bool Init(const MediaConstraintsInterface* constraints,
- const PeerConnectionFactory::Options* options) {
+ bool Init(
+ const MediaConstraintsInterface* constraints,
+ const PeerConnectionFactory::Options* options,
+ rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
EXPECT_TRUE(!peer_connection_);
EXPECT_TRUE(!peer_connection_factory_);
- allocator_factory_ = webrtc::FakePortAllocatorFactory::Create();
- if (!allocator_factory_) {
- return false;
- }
+ rtc::scoped_ptr<cricket::PortAllocator> port_allocator(
+ new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
if (fake_audio_capture_module_ == nullptr) {
@@ -666,46 +759,29 @@
if (options) {
peer_connection_factory_->SetOptions(*options);
}
- peer_connection_ = CreatePeerConnection(allocator_factory_.get(),
- constraints);
+ peer_connection_ = CreatePeerConnection(
+ std::move(port_allocator), constraints, std::move(dtls_identity_store));
return peer_connection_.get() != nullptr;
}
- rtc::scoped_refptr<webrtc::VideoTrackInterface>
- CreateLocalVideoTrack(const std::string stream_label) {
- // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
- FakeConstraints source_constraints = video_constraints_;
- source_constraints.SetMandatoryMaxFrameRate(10);
-
- cricket::FakeVideoCapturer* fake_capturer =
- new webrtc::FakePeriodicVideoCapturer();
- video_capturers_.push_back(fake_capturer);
- rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
- peer_connection_factory_->CreateVideoSource(
- fake_capturer, &source_constraints);
- std::string label = stream_label + kVideoTrackLabelBase;
- return peer_connection_factory_->CreateVideoTrack(label, source);
- }
-
rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
- webrtc::PortAllocatorFactoryInterface* factory,
- const MediaConstraintsInterface* constraints) {
- // CreatePeerConnection with IceServers.
- webrtc::PeerConnectionInterface::IceServers ice_servers;
+ rtc::scoped_ptr<cricket::PortAllocator> port_allocator,
+ const MediaConstraintsInterface* constraints,
+ rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
+ // CreatePeerConnection with RTCConfiguration.
+ webrtc::PeerConnectionInterface::RTCConfiguration config;
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = "stun:stun.l.google.com:19302";
- ice_servers.push_back(ice_server);
+ config.servers.push_back(ice_server);
- rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store(
- rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
- : nullptr);
return peer_connection_factory_->CreatePeerConnection(
- ice_servers, constraints, factory, dtls_identity_store.Pass(), this);
+ config, constraints, std::move(port_allocator),
+ std::move(dtls_identity_store), this);
}
void HandleIncomingOffer(const std::string& msg) {
LOG(INFO) << id_ << "HandleIncomingOffer ";
- if (NumberOfLocalMediaStreams() == 0) {
+ if (NumberOfLocalMediaStreams() == 0 && auto_add_stream_) {
// If we are not sending any streams ourselves it is time to add some.
AddMediaStream(true, true);
}
@@ -807,20 +883,24 @@
std::string id_;
- rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface> allocator_factory_;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
+ bool auto_add_stream_ = true;
+
typedef std::pair<std::string, std::string> IceUfragPwdPair;
std::map<int, IceUfragPwdPair> ice_ufrag_pwd_;
bool expect_ice_restart_ = false;
- // Needed to keep track of number of frames send.
+ // Needed to keep track of number of frames sent.
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
// Needed to keep track of number of frames received.
- typedef std::map<std::string, webrtc::FakeVideoTrackRenderer*> RenderMap;
- RenderMap fake_video_renderers_;
+ std::map<std::string, rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer>>
+ fake_video_renderers_;
+ // Needed to ensure frames aren't received for removed tracks.
+ std::vector<rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer>>
+ removed_fake_video_renderers_;
// Needed to keep track of number of frames received when external decoder
// used.
FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_ = nullptr;
@@ -846,11 +926,9 @@
rtc::scoped_ptr<MockDataChannelObserver> data_observer_;
};
-// TODO(deadbeef): Rename this to P2PTestConductor once the Linux memcheck and
-// Windows DrMemory Full bots' blacklists are updated.
-class JsepPeerConnectionP2PTestClient : public testing::Test {
+class P2PTestConductor : public testing::Test {
public:
- JsepPeerConnectionP2PTestClient()
+ P2PTestConductor()
: pss_(new rtc::PhysicalSocketServer),
ss_(new rtc::VirtualSocketServer(pss_.get())),
ss_scope_(ss_.get()) {}
@@ -882,13 +960,26 @@
}
void TestUpdateOfferWithRejectedContent() {
+ // Renegotiate, rejecting the video m-line.
initiating_client_->Negotiate(true, false);
- EXPECT_TRUE_WAIT(
- FramesNotPending(kEndAudioFrameCount * 2, kEndVideoFrameCount),
- kMaxWaitForFramesMs);
- // There shouldn't be any more video frame after the new offer is
- // negotiated.
- EXPECT_FALSE(VideoFramesReceivedCheck(kEndVideoFrameCount + 1));
+ ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
+
+ int pc1_audio_received = initiating_client_->audio_frames_received();
+ int pc1_video_received = initiating_client_->video_frames_received();
+ int pc2_audio_received = receiving_client_->audio_frames_received();
+ int pc2_video_received = receiving_client_->video_frames_received();
+
+ // Wait for some additional audio frames to be received.
+ EXPECT_TRUE_WAIT(initiating_client_->AudioFramesReceivedCheck(
+ pc1_audio_received + kEndAudioFrameCount) &&
+ receiving_client_->AudioFramesReceivedCheck(
+ pc2_audio_received + kEndAudioFrameCount),
+ kMaxWaitForFramesMs);
+
+ // During this time, we shouldn't have received any additional video frames
+ // for the rejected video tracks.
+ EXPECT_EQ(pc1_video_received, initiating_client_->video_frames_received());
+ EXPECT_EQ(pc2_video_received, receiving_client_->video_frames_received());
}
void VerifyRenderedSize(int width, int height) {
@@ -905,7 +996,7 @@
receiving_client_->VerifyLocalIceUfragAndPassword();
}
- ~JsepPeerConnectionP2PTestClient() {
+ ~P2PTestConductor() {
if (initiating_client_) {
initiating_client_->set_signaling_message_receiver(nullptr);
}
@@ -922,6 +1013,11 @@
nullptr);
}
+ void SetSignalingReceivers() {
+ initiating_client_->set_signaling_message_receiver(receiving_client_.get());
+ receiving_client_->set_signaling_message_receiver(initiating_client_.get());
+ }
+
bool CreateTestClients(MediaConstraintsInterface* init_constraints,
PeerConnectionFactory::Options* init_options,
MediaConstraintsInterface* recv_constraints,
@@ -933,8 +1029,7 @@
if (!initiating_client_ || !receiving_client_) {
return false;
}
- initiating_client_->set_signaling_message_receiver(receiving_client_.get());
- receiving_client_->set_signaling_message_receiver(initiating_client_.get());
+ SetSignalingReceivers();
return true;
}
@@ -957,13 +1052,11 @@
initiating_client_->AddMediaStream(true, true);
}
initiating_client_->Negotiate();
- const int kMaxWaitForActivationMs = 5000;
// Assert true is used here since next tests are guaranteed to fail and
// would eat up 5 seconds.
ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
VerifySessionDescriptions();
-
int audio_frame_count = kEndAudioFrameCount;
// TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.
if (!initiating_client_->can_receive_audio() ||
@@ -1013,6 +1106,32 @@
kMaxWaitForFramesMs);
}
+ void SetupAndVerifyDtlsCall() {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+ FakeConstraints setup_constraints;
+ setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
+ true);
+ ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
+ LocalP2PTest();
+ VerifyRenderedSize(640, 480);
+ }
+
+ PeerConnectionTestClient* CreateDtlsClientWithAlternateKey() {
+ FakeConstraints setup_constraints;
+ setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
+ true);
+
+ rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
+ rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore()
+ : nullptr);
+ dtls_identity_store->use_alternate_key();
+
+ // Make sure the new client is using a different certificate.
+ return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore(
+ "New Peer: ", &setup_constraints, nullptr,
+ std::move(dtls_identity_store));
+ }
+
void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) {
// Messages may get lost on the unreliable DataChannel, so we send multiple
// times to avoid test flakiness.
@@ -1026,10 +1145,29 @@
PeerConnectionTestClient* initializing_client() {
return initiating_client_.get();
}
+
+ // Set the |initiating_client_| to the |client| passed in and return the
+ // original |initiating_client_|.
+ PeerConnectionTestClient* set_initializing_client(
+ PeerConnectionTestClient* client) {
+ PeerConnectionTestClient* old = initiating_client_.release();
+ initiating_client_.reset(client);
+ return old;
+ }
+
PeerConnectionTestClient* receiving_client() {
return receiving_client_.get();
}
+ // Set the |receiving_client_| to the |client| passed in and return the
+ // original |receiving_client_|.
+ PeerConnectionTestClient* set_receiving_client(
+ PeerConnectionTestClient* client) {
+ PeerConnectionTestClient* old = receiving_client_.release();
+ receiving_client_.reset(client);
+ return old;
+ }
+
private:
rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_;
rtc::scoped_ptr<rtc::VirtualSocketServer> ss_;
@@ -1045,7 +1183,7 @@
// This test sets up a Jsep call between two parties and test Dtmf.
// TODO(holmer): Disabled due to sometimes crashing on buildbots.
// See issue webrtc/2378.
-TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestDtmf) {
+TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
VerifyDtmf();
@@ -1053,7 +1191,7 @@
// This test sets up a Jsep call between two parties and test that we can get a
// video aspect ratio of 16:9.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTest16To9) {
+TEST_F(P2PTestConductor, LocalP2PTest16To9) {
ASSERT_TRUE(CreateTestClients());
FakeConstraints constraint;
double requested_ratio = 640.0/360;
@@ -1078,7 +1216,7 @@
// received video has a resolution of 1280*720.
// TODO(mallinath): Enable when
// http://code.google.com/p/webrtc/issues/detail?id=981 is fixed.
-TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTest1280By720) {
+TEST_F(P2PTestConductor, DISABLED_LocalP2PTest1280By720) {
ASSERT_TRUE(CreateTestClients());
FakeConstraints constraint;
constraint.SetMandatoryMinWidth(1280);
@@ -1090,19 +1228,13 @@
// This test sets up a call between two endpoints that are configured to use
// DTLS key agreement. As a result, DTLS is negotiated and used for transport.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtls) {
- MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
- FakeConstraints setup_constraints;
- setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
- true);
- ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
- LocalP2PTest();
- VerifyRenderedSize(640, 480);
+TEST_F(P2PTestConductor, LocalP2PTestDtls) {
+ SetupAndVerifyDtlsCall();
}
// This test sets up a audio call initially and then upgrades to audio/video,
// using DTLS.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsRenegotiate) {
+TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
@@ -1114,10 +1246,66 @@
receiving_client()->Negotiate();
}
+// This test sets up a call transfer to a new caller with a different DTLS
+// fingerprint.
+TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+ SetupAndVerifyDtlsCall();
+
+ // Keeping the original peer around which will still send packets to the
+ // receiving client. These SRTP packets will be dropped.
+ rtc::scoped_ptr<PeerConnectionTestClient> original_peer(
+ set_initializing_client(CreateDtlsClientWithAlternateKey()));
+ original_peer->pc()->Close();
+
+ SetSignalingReceivers();
+ receiving_client()->SetExpectIceRestart(true);
+ LocalP2PTest();
+ VerifyRenderedSize(640, 480);
+}
+
+// This test sets up a non-bundle call and apply bundle during ICE restart. When
+// bundle is in effect in the restart, the channel can successfully reset its
+// DTLS-SRTP context.
+TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+ FakeConstraints setup_constraints;
+ setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
+ true);
+ ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
+ receiving_client()->RemoveBundleFromReceivedSdp(true);
+ LocalP2PTest();
+ VerifyRenderedSize(640, 480);
+
+ initializing_client()->IceRestart();
+ receiving_client()->SetExpectIceRestart(true);
+ receiving_client()->RemoveBundleFromReceivedSdp(false);
+ LocalP2PTest();
+ VerifyRenderedSize(640, 480);
+}
+
+// This test sets up a call transfer to a new callee with a different DTLS
+// fingerprint.
+TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+ SetupAndVerifyDtlsCall();
+
+ // Keeping the original peer around which will still send packets to the
+ // receiving client. These SRTP packets will be dropped.
+ rtc::scoped_ptr<PeerConnectionTestClient> original_peer(
+ set_receiving_client(CreateDtlsClientWithAlternateKey()));
+ original_peer->pc()->Close();
+
+ SetSignalingReceivers();
+ initializing_client()->IceRestart();
+ LocalP2PTest();
+ VerifyRenderedSize(640, 480);
+}
+
// This test sets up a call between two endpoints that are configured to use
// DTLS key agreement. The offerer don't support SDES. As a result, DTLS is
// negotiated and used for transport.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestOfferDtlsButNotSdes) {
+TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints setup_constraints;
setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
@@ -1130,7 +1318,7 @@
// This test sets up a Jsep call between two parties, and the callee only
// accept to receive video.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerVideo) {
+TEST_F(P2PTestConductor, LocalP2PTestAnswerVideo) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->SetReceiveAudioVideo(false, true);
LocalP2PTest();
@@ -1138,7 +1326,7 @@
// This test sets up a Jsep call between two parties, and the callee only
// accept to receive audio.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerAudio) {
+TEST_F(P2PTestConductor, LocalP2PTestAnswerAudio) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->SetReceiveAudioVideo(true, false);
LocalP2PTest();
@@ -1146,7 +1334,7 @@
// This test sets up a Jsep call between two parties, and the callee reject both
// audio and video.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestAnswerNone) {
+TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->SetReceiveAudioVideo(false, false);
LocalP2PTest();
@@ -1156,9 +1344,7 @@
// runs for a while (10 frames), the caller sends an update offer with video
// being rejected. Once the re-negotiation is done, the video flow should stop
// and the audio flow should continue.
-// Disabled due to b/14955157.
-TEST_F(JsepPeerConnectionP2PTestClient,
- DISABLED_UpdateOfferWithRejectedContent) {
+TEST_F(P2PTestConductor, UpdateOfferWithRejectedContent) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
TestUpdateOfferWithRejectedContent();
@@ -1166,8 +1352,7 @@
// This test sets up a Jsep call between two parties. The MSID is removed from
// the SDP strings from the caller.
-// Disabled due to b/14955157.
-TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestWithoutMsid) {
+TEST_F(P2PTestConductor, LocalP2PTestWithoutMsid) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->RemoveMsidFromReceivedSdp(true);
// TODO(perkj): Currently there is a bug that cause audio to stop playing if
@@ -1182,7 +1367,7 @@
// sends two steams.
// TODO(perkj): Disabled due to
// https://code.google.com/p/webrtc/issues/detail?id=1454
-TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_LocalP2PTestTwoStreams) {
+TEST_F(P2PTestConductor, DISABLED_LocalP2PTestTwoStreams) {
ASSERT_TRUE(CreateTestClients());
// Set optional video constraint to max 320pixels to decrease CPU usage.
FakeConstraints constraint;
@@ -1196,7 +1381,7 @@
}
// Test that we can receive the audio output level from a remote audio track.
-TEST_F(JsepPeerConnectionP2PTestClient, GetAudioOutputLevelStats) {
+TEST_F(P2PTestConductor, GetAudioOutputLevelStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
@@ -1215,7 +1400,7 @@
}
// Test that an audio input level is reported.
-TEST_F(JsepPeerConnectionP2PTestClient, GetAudioInputLevelStats) {
+TEST_F(P2PTestConductor, GetAudioInputLevelStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
@@ -1226,7 +1411,7 @@
}
// Test that we can get incoming byte counts from both audio and video tracks.
-TEST_F(JsepPeerConnectionP2PTestClient, GetBytesReceivedStats) {
+TEST_F(P2PTestConductor, GetBytesReceivedStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
@@ -1248,7 +1433,7 @@
}
// Test that we can get outgoing byte counts from both audio and video tracks.
-TEST_F(JsepPeerConnectionP2PTestClient, GetBytesSentStats) {
+TEST_F(P2PTestConductor, GetBytesSentStats) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
@@ -1270,7 +1455,7 @@
}
// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
-TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
+TEST_F(P2PTestConductor, GetDtls12None) {
PeerConnectionFactory::Options init_options;
init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
PeerConnectionFactory::Options recv_options;
@@ -1282,7 +1467,7 @@
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
initializing_client()->GetDtlsCipherStats(),
@@ -1292,16 +1477,23 @@
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
- EXPECT_EQ_WAIT(kDefaultSrtpCipher,
+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
- EXPECT_EQ(1, init_observer->GetEnumCounter(
- webrtc::kEnumCounterAudioSrtpCipher,
- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+ EXPECT_EQ(1,
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
+ kDefaultSrtpCryptoSuite));
}
+#if defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
+#define MAYBE_GetDtls12Both DISABLED_GetDtls12Both
+#else
+#define MAYBE_GetDtls12Both GetDtls12Both
+#endif
// Test that DTLS 1.2 is used if both ends support it.
-TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) {
+TEST_F(P2PTestConductor, MAYBE_GetDtls12Both) {
PeerConnectionFactory::Options init_options;
init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
PeerConnectionFactory::Options recv_options;
@@ -1313,7 +1505,7 @@
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)),
initializing_client()->GetDtlsCipherStats(),
@@ -1323,17 +1515,17 @@
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)));
- EXPECT_EQ_WAIT(kDefaultSrtpCipher,
+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
- EXPECT_EQ(1, init_observer->GetEnumCounter(
- webrtc::kEnumCounterAudioSrtpCipher,
- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+ EXPECT_EQ(1,
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
+ kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
// received supports 1.0.
-TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {
+TEST_F(P2PTestConductor, GetDtls12Init) {
PeerConnectionFactory::Options init_options;
init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
PeerConnectionFactory::Options recv_options;
@@ -1345,7 +1537,7 @@
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
initializing_client()->GetDtlsCipherStats(),
@@ -1355,17 +1547,17 @@
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
- EXPECT_EQ_WAIT(kDefaultSrtpCipher,
+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
- EXPECT_EQ(1, init_observer->GetEnumCounter(
- webrtc::kEnumCounterAudioSrtpCipher,
- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+ EXPECT_EQ(1,
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
+ kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
// received supports 1.2.
-TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
+TEST_F(P2PTestConductor, GetDtls12Recv) {
PeerConnectionFactory::Options init_options;
init_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
PeerConnectionFactory::Options recv_options;
@@ -1377,7 +1569,7 @@
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
initializing_client()->GetDtlsCipherStats(),
@@ -1387,16 +1579,17 @@
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
- EXPECT_EQ_WAIT(kDefaultSrtpCipher,
+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
- EXPECT_EQ(1, init_observer->GetEnumCounter(
- webrtc::kEnumCounterAudioSrtpCipher,
- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+ EXPECT_EQ(1,
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
+ kDefaultSrtpCryptoSuite));
}
-// This test sets up a call between two parties with audio, video and data.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDataChannel) {
+// This test sets up a call between two parties with audio, video and an RTP
+// data channel.
+TEST_F(P2PTestConductor, LocalP2PTestRtpDataChannel) {
FakeConstraints setup_constraints;
setup_constraints.SetAllowRtpDataChannels();
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
@@ -1426,6 +1619,34 @@
EXPECT_FALSE(receiving_client()->data_observer()->IsOpen());
}
+// This test sets up a call between two parties with audio, video and an SCTP
+// data channel.
+TEST_F(P2PTestConductor, LocalP2PTestSctpDataChannel) {
+ ASSERT_TRUE(CreateTestClients());
+ initializing_client()->CreateDataChannel();
+ LocalP2PTest();
+ ASSERT_TRUE(initializing_client()->data_channel() != nullptr);
+ EXPECT_TRUE_WAIT(receiving_client()->data_channel() != nullptr, kMaxWaitMs);
+ EXPECT_TRUE_WAIT(initializing_client()->data_observer()->IsOpen(),
+ kMaxWaitMs);
+ EXPECT_TRUE_WAIT(receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
+
+ std::string data = "hello world";
+
+ initializing_client()->data_channel()->Send(DataBuffer(data));
+ EXPECT_EQ_WAIT(data, receiving_client()->data_observer()->last_message(),
+ kMaxWaitMs);
+
+ receiving_client()->data_channel()->Send(DataBuffer(data));
+ EXPECT_EQ_WAIT(data, initializing_client()->data_observer()->last_message(),
+ kMaxWaitMs);
+
+ receiving_client()->data_channel()->Close();
+ EXPECT_TRUE_WAIT(!initializing_client()->data_observer()->IsOpen(),
+ kMaxWaitMs);
+ EXPECT_TRUE_WAIT(!receiving_client()->data_observer()->IsOpen(), kMaxWaitMs);
+}
+
// This test sets up a call between two parties and creates a data channel.
// The test tests that received data is buffered unless an observer has been
// registered.
@@ -1433,7 +1654,7 @@
// transport has detected that a channel is writable and thus data can be
// received before the data channel state changes to open. That is hard to test
// but the same buffering is used in that case.
-TEST_F(JsepPeerConnectionP2PTestClient, RegisterDataChannelObserver) {
+TEST_F(P2PTestConductor, RegisterDataChannelObserver) {
FakeConstraints setup_constraints;
setup_constraints.SetAllowRtpDataChannels();
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
@@ -1463,7 +1684,7 @@
// This test sets up a call between two parties with audio, video and but only
// the initiating client support data.
-TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestReceiverDoesntSupportData) {
+TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportData) {
FakeConstraints setup_constraints_1;
setup_constraints_1.SetAllowRtpDataChannels();
// Must disable DTLS to make negotiation succeed.
@@ -1482,7 +1703,7 @@
// This test sets up a call between two parties with audio, video. When audio
// and video is setup and flowing and data channel is negotiated.
-TEST_F(JsepPeerConnectionP2PTestClient, AddDataChannelAfterRenegotiation) {
+TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) {
FakeConstraints setup_constraints;
setup_constraints.SetAllowRtpDataChannels();
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
@@ -1501,7 +1722,7 @@
// This test sets up a Jsep call with SCTP DataChannel and verifies the
// negotiation is completed without error.
#ifdef HAVE_SCTP
-TEST_F(JsepPeerConnectionP2PTestClient, CreateOfferWithSctpDataChannel) {
+TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) {
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
FakeConstraints constraints;
constraints.SetMandatory(
@@ -1515,7 +1736,7 @@
// This test sets up a call between two parties with audio, and video.
// During the call, the initializing side restart ice and the test verifies that
// new ice candidates are generated and audio and video still can flow.
-TEST_F(JsepPeerConnectionP2PTestClient, IceRestart) {
+TEST_F(P2PTestConductor, IceRestart) {
ASSERT_TRUE(CreateTestClients());
// Negotiate and wait for ice completion and make sure audio and video plays.
@@ -1562,17 +1783,69 @@
EXPECT_NE(receiver_candidate, receiver_candidate_restart);
}
+// This test sets up a call between two parties with audio, and video.
+// It then renegotiates setting the video m-line to "port 0", then later
+// renegotiates again, enabling video.
+TEST_F(P2PTestConductor, LocalP2PTestVideoDisableEnable) {
+ ASSERT_TRUE(CreateTestClients());
+
+ // Do initial negotiation. Will result in video and audio sendonly m-lines.
+ receiving_client()->set_auto_add_stream(false);
+ initializing_client()->AddMediaStream(true, true);
+ initializing_client()->Negotiate();
+
+ // Negotiate again, disabling the video m-line (receiving client will
+ // set port to 0 due to mandatory "OfferToReceiveVideo: false" constraint).
+ receiving_client()->SetReceiveVideo(false);
+ initializing_client()->Negotiate();
+
+ // Enable video and do negotiation again, making sure video is received
+ // end-to-end.
+ receiving_client()->SetReceiveVideo(true);
+ receiving_client()->AddMediaStream(true, true);
+ LocalP2PTest();
+}
+
// This test sets up a Jsep call between two parties with external
// VideoDecoderFactory.
// TODO(holmer): Disabled due to sometimes crashing on buildbots.
// See issue webrtc/2378.
-TEST_F(JsepPeerConnectionP2PTestClient,
- DISABLED_LocalP2PTestWithVideoDecoderFactory) {
+TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithVideoDecoderFactory) {
ASSERT_TRUE(CreateTestClients());
EnableVideoDecoderFactory();
LocalP2PTest();
}
+// This tests that if we negotiate after calling CreateSender but before we
+// have a track, then set a track later, frames from the newly-set track are
+// received end-to-end.
+TEST_F(P2PTestConductor, EarlyWarmupTest) {
+ ASSERT_TRUE(CreateTestClients());
+ auto audio_sender =
+ initializing_client()->pc()->CreateSender("audio", "stream_id");
+ auto video_sender =
+ initializing_client()->pc()->CreateSender("video", "stream_id");
+ initializing_client()->Negotiate();
+ // Wait for ICE connection to complete, without any tracks.
+ // Note that the receiving client WILL (in HandleIncomingOffer) create
+ // tracks, so it's only the initiator here that's doing early warmup.
+ ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
+ VerifySessionDescriptions();
+ EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+ initializing_client()->ice_connection_state(),
+ kMaxWaitForFramesMs);
+ EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+ receiving_client()->ice_connection_state(),
+ kMaxWaitForFramesMs);
+ // Now set the tracks, and expect frames to immediately start flowing.
+ EXPECT_TRUE(
+ audio_sender->SetTrack(initializing_client()->CreateLocalAudioTrack("")));
+ EXPECT_TRUE(
+ video_sender->SetTrack(initializing_client()->CreateLocalVideoTrack("")));
+ EXPECT_TRUE_WAIT(FramesNotPending(kEndAudioFrameCount, kEndVideoFrameCount),
+ kMaxWaitForFramesMs);
+}
+
class IceServerParsingTest : public testing::Test {
public:
// Convenience for parsing a single URL.
@@ -1589,38 +1862,37 @@
server.username = username;
server.password = password;
servers.push_back(server);
- return webrtc::ParseIceServers(servers, &stun_configurations_,
- &turn_configurations_);
+ return webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_);
}
protected:
- webrtc::StunConfigurations stun_configurations_;
- webrtc::TurnConfigurations turn_configurations_;
+ cricket::ServerAddresses stun_servers_;
+ std::vector<cricket::RelayServerConfig> turn_servers_;
};
// Make sure all STUN/TURN prefixes are parsed correctly.
TEST_F(IceServerParsingTest, ParseStunPrefixes) {
EXPECT_TRUE(ParseUrl("stun:hostname"));
- EXPECT_EQ(1U, stun_configurations_.size());
- EXPECT_EQ(0U, turn_configurations_.size());
- stun_configurations_.clear();
+ EXPECT_EQ(1U, stun_servers_.size());
+ EXPECT_EQ(0U, turn_servers_.size());
+ stun_servers_.clear();
EXPECT_TRUE(ParseUrl("stuns:hostname"));
- EXPECT_EQ(1U, stun_configurations_.size());
- EXPECT_EQ(0U, turn_configurations_.size());
- stun_configurations_.clear();
+ EXPECT_EQ(1U, stun_servers_.size());
+ EXPECT_EQ(0U, turn_servers_.size());
+ stun_servers_.clear();
EXPECT_TRUE(ParseUrl("turn:hostname"));
- EXPECT_EQ(0U, stun_configurations_.size());
- EXPECT_EQ(1U, turn_configurations_.size());
- EXPECT_FALSE(turn_configurations_[0].secure);
- turn_configurations_.clear();
+ EXPECT_EQ(0U, stun_servers_.size());
+ EXPECT_EQ(1U, turn_servers_.size());
+ EXPECT_FALSE(turn_servers_[0].ports[0].secure);
+ turn_servers_.clear();
EXPECT_TRUE(ParseUrl("turns:hostname"));
- EXPECT_EQ(0U, stun_configurations_.size());
- EXPECT_EQ(1U, turn_configurations_.size());
- EXPECT_TRUE(turn_configurations_[0].secure);
- turn_configurations_.clear();
+ EXPECT_EQ(0U, stun_servers_.size());
+ EXPECT_EQ(1U, turn_servers_.size());
+ EXPECT_TRUE(turn_servers_[0].ports[0].secure);
+ turn_servers_.clear();
// invalid prefixes
EXPECT_FALSE(ParseUrl("stunn:hostname"));
@@ -1632,67 +1904,69 @@
TEST_F(IceServerParsingTest, VerifyDefaults) {
// TURNS defaults
EXPECT_TRUE(ParseUrl("turns:hostname"));
- EXPECT_EQ(1U, turn_configurations_.size());
- EXPECT_EQ(5349, turn_configurations_[0].server.port());
- EXPECT_EQ("tcp", turn_configurations_[0].transport_type);
- turn_configurations_.clear();
+ EXPECT_EQ(1U, turn_servers_.size());
+ EXPECT_EQ(5349, turn_servers_[0].ports[0].address.port());
+ EXPECT_EQ(cricket::PROTO_TCP, turn_servers_[0].ports[0].proto);
+ turn_servers_.clear();
// TURN defaults
EXPECT_TRUE(ParseUrl("turn:hostname"));
- EXPECT_EQ(1U, turn_configurations_.size());
- EXPECT_EQ(3478, turn_configurations_[0].server.port());
- EXPECT_EQ("udp", turn_configurations_[0].transport_type);
- turn_configurations_.clear();
+ EXPECT_EQ(1U, turn_servers_.size());
+ EXPECT_EQ(3478, turn_servers_[0].ports[0].address.port());
+ EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
+ turn_servers_.clear();
// STUN defaults
EXPECT_TRUE(ParseUrl("stun:hostname"));
- EXPECT_EQ(1U, stun_configurations_.size());
- EXPECT_EQ(3478, stun_configurations_[0].server.port());
- stun_configurations_.clear();
+ EXPECT_EQ(1U, stun_servers_.size());
+ EXPECT_EQ(3478, stun_servers_.begin()->port());
+ stun_servers_.clear();
}
// Check that the 6 combinations of IPv4/IPv6/hostname and with/without port
// can be parsed correctly.
TEST_F(IceServerParsingTest, ParseHostnameAndPort) {
EXPECT_TRUE(ParseUrl("stun:1.2.3.4:1234"));
- EXPECT_EQ(1U, stun_configurations_.size());
- EXPECT_EQ("1.2.3.4", stun_configurations_[0].server.hostname());
- EXPECT_EQ(1234, stun_configurations_[0].server.port());
- stun_configurations_.clear();
+ EXPECT_EQ(1U, stun_servers_.size());
+ EXPECT_EQ("1.2.3.4", stun_servers_.begin()->hostname());
+ EXPECT_EQ(1234, stun_servers_.begin()->port());
+ stun_servers_.clear();
EXPECT_TRUE(ParseUrl("stun:[1:2:3:4:5:6:7:8]:4321"));
- EXPECT_EQ(1U, stun_configurations_.size());
- EXPECT_EQ("1:2:3:4:5:6:7:8", stun_configurations_[0].server.hostname());
- EXPECT_EQ(4321, stun_configurations_[0].server.port());
- stun_configurations_.clear();
+ EXPECT_EQ(1U, stun_servers_.size());
+ EXPECT_EQ("1:2:3:4:5:6:7:8", stun_servers_.begin()->hostname());
+ EXPECT_EQ(4321, stun_servers_.begin()->port());
+ stun_servers_.clear();
EXPECT_TRUE(ParseUrl("stun:hostname:9999"));
- EXPECT_EQ(1U, stun_configurations_.size());
- EXPECT_EQ("hostname", stun_configurations_[0].server.hostname());
- EXPECT_EQ(9999, stun_configurations_[0].server.port());
- stun_configurations_.clear();
+ EXPECT_EQ(1U, stun_servers_.size());
+ EXPECT_EQ("hostname", stun_servers_.begin()->hostname());
+ EXPECT_EQ(9999, stun_servers_.begin()->port());
+ stun_servers_.clear();
EXPECT_TRUE(ParseUrl("stun:1.2.3.4"));
- EXPECT_EQ(1U, stun_configurations_.size());
- EXPECT_EQ("1.2.3.4", stun_configurations_[0].server.hostname());
- EXPECT_EQ(3478, stun_configurations_[0].server.port());
- stun_configurations_.clear();
+ EXPECT_EQ(1U, stun_servers_.size());
+ EXPECT_EQ("1.2.3.4", stun_servers_.begin()->hostname());
+ EXPECT_EQ(3478, stun_servers_.begin()->port());
+ stun_servers_.clear();
EXPECT_TRUE(ParseUrl("stun:[1:2:3:4:5:6:7:8]"));
- EXPECT_EQ(1U, stun_configurations_.size());
- EXPECT_EQ("1:2:3:4:5:6:7:8", stun_configurations_[0].server.hostname());
- EXPECT_EQ(3478, stun_configurations_[0].server.port());
- stun_configurations_.clear();
+ EXPECT_EQ(1U, stun_servers_.size());
+ EXPECT_EQ("1:2:3:4:5:6:7:8", stun_servers_.begin()->hostname());
+ EXPECT_EQ(3478, stun_servers_.begin()->port());
+ stun_servers_.clear();
EXPECT_TRUE(ParseUrl("stun:hostname"));
- EXPECT_EQ(1U, stun_configurations_.size());
- EXPECT_EQ("hostname", stun_configurations_[0].server.hostname());
- EXPECT_EQ(3478, stun_configurations_[0].server.port());
- stun_configurations_.clear();
+ EXPECT_EQ(1U, stun_servers_.size());
+ EXPECT_EQ("hostname", stun_servers_.begin()->hostname());
+ EXPECT_EQ(3478, stun_servers_.begin()->port());
+ stun_servers_.clear();
// Try some invalid hostname:port strings.
EXPECT_FALSE(ParseUrl("stun:hostname:99a99"));
EXPECT_FALSE(ParseUrl("stun:hostname:-1"));
+ EXPECT_FALSE(ParseUrl("stun:hostname:port:more"));
+ EXPECT_FALSE(ParseUrl("stun:hostname:port more"));
EXPECT_FALSE(ParseUrl("stun:hostname:"));
EXPECT_FALSE(ParseUrl("stun:[1:2:3:4:5:6:7:8]junk:1000"));
EXPECT_FALSE(ParseUrl("stun::5555"));
@@ -1702,14 +1976,14 @@
// Test parsing the "?transport=xxx" part of the URL.
TEST_F(IceServerParsingTest, ParseTransport) {
EXPECT_TRUE(ParseUrl("turn:hostname:1234?transport=tcp"));
- EXPECT_EQ(1U, turn_configurations_.size());
- EXPECT_EQ("tcp", turn_configurations_[0].transport_type);
- turn_configurations_.clear();
+ EXPECT_EQ(1U, turn_servers_.size());
+ EXPECT_EQ(cricket::PROTO_TCP, turn_servers_[0].ports[0].proto);
+ turn_servers_.clear();
EXPECT_TRUE(ParseUrl("turn:hostname?transport=udp"));
- EXPECT_EQ(1U, turn_configurations_.size());
- EXPECT_EQ("udp", turn_configurations_[0].transport_type);
- turn_configurations_.clear();
+ EXPECT_EQ(1U, turn_servers_.size());
+ EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
+ turn_servers_.clear();
EXPECT_FALSE(ParseUrl("turn:hostname?transport=invalid"));
}
@@ -1717,9 +1991,9 @@
// Test parsing ICE username contained in URL.
TEST_F(IceServerParsingTest, ParseUsername) {
EXPECT_TRUE(ParseUrl("turn:user@hostname"));
- EXPECT_EQ(1U, turn_configurations_.size());
- EXPECT_EQ("user", turn_configurations_[0].username);
- turn_configurations_.clear();
+ EXPECT_EQ(1U, turn_servers_.size());
+ EXPECT_EQ("user", turn_servers_[0].credentials.username);
+ turn_servers_.clear();
EXPECT_FALSE(ParseUrl("turn:@hostname"));
EXPECT_FALSE(ParseUrl("turn:username@"));
@@ -1728,12 +2002,12 @@
}
// Test that username and password from IceServer is copied into the resulting
-// TurnConfiguration.
+// RelayServerConfig.
TEST_F(IceServerParsingTest, CopyUsernameAndPasswordFromIceServer) {
EXPECT_TRUE(ParseUrl("turn:hostname", "username", "password"));
- EXPECT_EQ(1U, turn_configurations_.size());
- EXPECT_EQ("username", turn_configurations_[0].username);
- EXPECT_EQ("password", turn_configurations_[0].password);
+ EXPECT_EQ(1U, turn_servers_.size());
+ EXPECT_EQ("username", turn_servers_[0].credentials.username);
+ EXPECT_EQ("password", turn_servers_[0].credentials.password);
}
// Ensure that if a server has multiple URLs, each one is parsed.
@@ -1743,10 +2017,22 @@
server.urls.push_back("stun:hostname");
server.urls.push_back("turn:hostname");
servers.push_back(server);
- EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_configurations_,
- &turn_configurations_));
- EXPECT_EQ(1U, stun_configurations_.size());
- EXPECT_EQ(1U, turn_configurations_.size());
+ EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
+ EXPECT_EQ(1U, stun_servers_.size());
+ EXPECT_EQ(1U, turn_servers_.size());
+}
+
+// Ensure that TURN servers are given unique priorities,
+// so that their resulting candidates have unique priorities.
+TEST_F(IceServerParsingTest, TurnServerPrioritiesUnique) {
+ PeerConnectionInterface::IceServers servers;
+ PeerConnectionInterface::IceServer server;
+ server.urls.push_back("turn:hostname");
+ server.urls.push_back("turn:hostname2");
+ servers.push_back(server);
+ EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
+ EXPECT_EQ(2U, turn_servers_.size());
+ EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
}
#endif // if !defined(THREAD_SANITIZER)
diff --git a/talk/app/webrtc/peerconnectionendtoend_unittest.cc b/talk/app/webrtc/peerconnectionendtoend_unittest.cc
index eacedd4..1a18031 100644
--- a/talk/app/webrtc/peerconnectionendtoend_unittest.cc
+++ b/talk/app/webrtc/peerconnectionendtoend_unittest.cc
@@ -27,6 +27,9 @@
#include "talk/app/webrtc/test/peerconnectiontestwrapper.h"
#include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
+#ifdef WEBRTC_ANDROID
+#include "talk/app/webrtc/test/androidtestinitializer.h"
+#endif
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/ssladapter.h"
@@ -50,56 +53,6 @@
const size_t kMaxWait = 10000;
-void RemoveLinesFromSdp(const std::string& line_start,
- std::string* sdp) {
- const char kSdpLineEnd[] = "\r\n";
- size_t ssrc_pos = 0;
- while ((ssrc_pos = sdp->find(line_start, ssrc_pos)) !=
- std::string::npos) {
- size_t end_ssrc = sdp->find(kSdpLineEnd, ssrc_pos);
- sdp->erase(ssrc_pos, end_ssrc - ssrc_pos + strlen(kSdpLineEnd));
- }
-}
-
-// Add |newlines| to the |message| after |line|.
-void InjectAfter(const std::string& line,
- const std::string& newlines,
- std::string* message) {
- const std::string tmp = line + newlines;
- rtc::replace_substrs(line.c_str(), line.length(),
- tmp.c_str(), tmp.length(), message);
-}
-
-void Replace(const std::string& line,
- const std::string& newlines,
- std::string* message) {
- rtc::replace_substrs(line.c_str(), line.length(),
- newlines.c_str(), newlines.length(), message);
-}
-
-void UseExternalSdes(std::string* sdp) {
- // Remove current crypto specification.
- RemoveLinesFromSdp("a=crypto", sdp);
- RemoveLinesFromSdp("a=fingerprint", sdp);
- // Add external crypto.
- const char kAudioSdes[] =
- "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
- "inline:PS1uQCVeeCFCanVmcjkpPywjNWhcYD0mXXtxaVBR\r\n";
- const char kVideoSdes[] =
- "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
- "inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj\r\n";
- const char kDataSdes[] =
- "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
- "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj\r\n";
- InjectAfter("a=mid:audio\r\n", kAudioSdes, sdp);
- InjectAfter("a=mid:video\r\n", kVideoSdes, sdp);
- InjectAfter("a=mid:data\r\n", kDataSdes, sdp);
-}
-
-void RemoveBundle(std::string* sdp) {
- RemoveLinesFromSdp("a=group:BUNDLE", sdp);
-}
-
} // namespace
class PeerConnectionEndToEndTest
@@ -114,6 +67,9 @@
"caller")),
callee_(new rtc::RefCountedObject<PeerConnectionTestWrapper>(
"callee")) {
+#ifdef WEBRTC_ANDROID
+ webrtc::InitializeAndroidObjects();
+#endif
}
void CreatePcs() {
@@ -217,15 +173,20 @@
DataChannelList callee_signaled_data_channels_;
};
+// Disabled for TSan v2, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=4719 for details.
+// Disabled for Mac, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5231 for details.
+#if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
TEST_F(PeerConnectionEndToEndTest, Call) {
CreatePcs();
GetAndAddUserMedia();
Negotiate();
WaitForCallEstablished();
}
+#endif // if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
-// Disabled per b/14899892
-TEST_F(PeerConnectionEndToEndTest, DISABLED_CallWithLegacySdp) {
+TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
FakeConstraints pc_constraints;
pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
false);
@@ -396,3 +357,30 @@
CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
}
+
+// This tests that if a data channel is closed remotely while not referenced
+// by the application (meaning only the PeerConnection contributes to its
+// reference count), no memory access violation will occur.
+// See: https://code.google.com/p/chromium/issues/detail?id=565048
+TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
+
+ CreatePcs();
+
+ webrtc::DataChannelInit init;
+ rtc::scoped_refptr<DataChannelInterface> caller_dc(
+ caller_->CreateDataChannel("data", init));
+
+ Negotiate();
+ WaitForConnection();
+
+ WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
+ // This removes the reference to the remote data channel that we hold.
+ callee_signaled_data_channels_.clear();
+ caller_dc->Close();
+ EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
+
+ // Wait for a bit longer so the remote data channel will receive the
+ // close message and be destroyed.
+ rtc::Thread::Current()->ProcessMessages(100);
+}
diff --git a/talk/app/webrtc/peerconnectionfactory.cc b/talk/app/webrtc/peerconnectionfactory.cc
index b46b4b6..c58f88c 100644
--- a/talk/app/webrtc/peerconnectionfactory.cc
+++ b/talk/app/webrtc/peerconnectionfactory.cc
@@ -27,6 +27,8 @@
#include "talk/app/webrtc/peerconnectionfactory.h"
+#include <utility>
+
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/localaudiosource.h"
#include "talk/app/webrtc/mediastream.h"
@@ -35,7 +37,6 @@
#include "talk/app/webrtc/peerconnection.h"
#include "talk/app/webrtc/peerconnectionfactoryproxy.h"
#include "talk/app/webrtc/peerconnectionproxy.h"
-#include "talk/app/webrtc/portallocatorfactory.h"
#include "talk/app/webrtc/videosource.h"
#include "talk/app/webrtc/videosourceproxy.h"
#include "talk/app/webrtc/videotrack.h"
@@ -44,6 +45,8 @@
#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
#include "webrtc/base/bind.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/p2p/base/basicpacketsocketfactory.h"
+#include "webrtc/p2p/client/basicportallocator.h"
namespace webrtc {
@@ -153,11 +156,13 @@
PeerConnectionFactory::~PeerConnectionFactory() {
RTC_DCHECK(signaling_thread_->IsCurrent());
channel_manager_.reset(nullptr);
- default_allocator_factory_ = nullptr;
// Make sure |worker_thread_| and |signaling_thread_| outlive
- // |dtls_identity_store_|.
+ // |dtls_identity_store_|, |default_socket_factory_| and
+ // |default_network_manager_|.
dtls_identity_store_ = nullptr;
+ default_socket_factory_ = nullptr;
+ default_network_manager_ = nullptr;
if (owns_ptrs_) {
if (wraps_current_thread_)
@@ -170,9 +175,16 @@
RTC_DCHECK(signaling_thread_->IsCurrent());
rtc::InitRandom(rtc::Time());
- default_allocator_factory_ = PortAllocatorFactory::Create(worker_thread_);
- if (!default_allocator_factory_)
+ default_network_manager_.reset(new rtc::BasicNetworkManager());
+ if (!default_network_manager_) {
return false;
+ }
+
+ default_socket_factory_.reset(
+ new rtc::BasicPacketSocketFactory(worker_thread_));
+ if (!default_socket_factory_) {
+ return false;
+ }
// TODO: Need to make sure only one VoE is created inside
// WebRtcMediaEngine.
@@ -208,8 +220,8 @@
cricket::VideoCapturer* capturer,
const MediaConstraintsInterface* constraints) {
RTC_DCHECK(signaling_thread_->IsCurrent());
- rtc::scoped_refptr<VideoSource> source(
- VideoSource::Create(channel_manager_.get(), capturer, constraints));
+ rtc::scoped_refptr<VideoSource> source(VideoSource::Create(
+ channel_manager_.get(), capturer, constraints, false));
return VideoSourceProxy::Create(signaling_thread_, source);
}
@@ -237,11 +249,10 @@
PeerConnectionFactory::CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
- PortAllocatorFactoryInterface* allocator_factory,
+ rtc::scoped_ptr<cricket::PortAllocator> allocator,
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
PeerConnectionObserver* observer) {
RTC_DCHECK(signaling_thread_->IsCurrent());
- RTC_DCHECK(allocator_factory || default_allocator_factory_);
if (!dtls_identity_store.get()) {
// Because |pc|->Initialize takes ownership of the store we need a new
@@ -251,19 +262,17 @@
new DtlsIdentityStoreWrapper(dtls_identity_store_));
}
- PortAllocatorFactoryInterface* chosen_allocator_factory =
- allocator_factory ? allocator_factory : default_allocator_factory_.get();
- chosen_allocator_factory->SetNetworkIgnoreMask(options_.network_ignore_mask);
+ if (!allocator) {
+ allocator.reset(new cricket::BasicPortAllocator(
+ default_network_manager_.get(), default_socket_factory_.get()));
+ }
+ allocator->SetNetworkIgnoreMask(options_.network_ignore_mask);
rtc::scoped_refptr<PeerConnection> pc(
new rtc::RefCountedObject<PeerConnection>(this));
- if (!pc->Initialize(
- configuration,
- constraints,
- chosen_allocator_factory,
- dtls_identity_store.Pass(),
- observer)) {
- return NULL;
+ if (!pc->Initialize(configuration, constraints, std::move(allocator),
+ std::move(dtls_identity_store), observer)) {
+ return nullptr;
}
return PeerConnectionProxy::Create(signaling_thread(), pc);
}
@@ -289,8 +298,7 @@
PeerConnectionFactory::CreateAudioTrack(const std::string& id,
AudioSourceInterface* source) {
RTC_DCHECK(signaling_thread_->IsCurrent());
- rtc::scoped_refptr<AudioTrackInterface> track(
- AudioTrack::Create(id, source));
+ rtc::scoped_refptr<AudioTrackInterface> track(AudioTrack::Create(id, source));
return AudioTrackProxy::Create(signaling_thread_, track);
}
diff --git a/talk/app/webrtc/peerconnectionfactory.h b/talk/app/webrtc/peerconnectionfactory.h
index af4117a..8b274e1 100644
--- a/talk/app/webrtc/peerconnectionfactory.h
+++ b/talk/app/webrtc/peerconnectionfactory.h
@@ -39,6 +39,11 @@
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread.h"
+namespace rtc {
+class BasicNetworkManager;
+class BasicPacketSocketFactory;
+}
+
namespace webrtc {
typedef rtc::RefCountedObject<DtlsIdentityStoreImpl>
@@ -50,14 +55,12 @@
options_ = options;
}
- // webrtc::PeerConnectionFactoryInterface override;
- rtc::scoped_refptr<PeerConnectionInterface>
- CreatePeerConnection(
- const PeerConnectionInterface::RTCConfiguration& configuration,
- const MediaConstraintsInterface* constraints,
- PortAllocatorFactoryInterface* allocator_factory,
- rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
- PeerConnectionObserver* observer) override;
+ rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
+ const PeerConnectionInterface::RTCConfiguration& configuration,
+ const MediaConstraintsInterface* constraints,
+ rtc::scoped_ptr<cricket::PortAllocator> allocator,
+ rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+ PeerConnectionObserver* observer) override;
bool Initialize();
@@ -107,7 +110,6 @@
rtc::Thread* signaling_thread_;
rtc::Thread* worker_thread_;
Options options_;
- rtc::scoped_refptr<PortAllocatorFactoryInterface> default_allocator_factory_;
// External Audio device used for audio playback.
rtc::scoped_refptr<AudioDeviceModule> default_adm_;
rtc::scoped_ptr<cricket::ChannelManager> channel_manager_;
@@ -119,6 +121,8 @@
// injected any. In that case, video engine will use the internal SW decoder.
rtc::scoped_ptr<cricket::WebRtcVideoDecoderFactory>
video_decoder_factory_;
+ rtc::scoped_ptr<rtc::BasicNetworkManager> default_network_manager_;
+ rtc::scoped_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;
};
diff --git a/talk/app/webrtc/peerconnectionfactory_unittest.cc b/talk/app/webrtc/peerconnectionfactory_unittest.cc
index f1d5353..9fb013b 100644
--- a/talk/app/webrtc/peerconnectionfactory_unittest.cc
+++ b/talk/app/webrtc/peerconnectionfactory_unittest.cc
@@ -26,10 +26,13 @@
*/
#include <string>
+#include <utility>
-#include "talk/app/webrtc/fakeportallocatorfactory.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/peerconnectionfactory.h"
+#ifdef WEBRTC_ANDROID
+#include "talk/app/webrtc/test/androidtestinitializer.h"
+#endif
#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
#include "talk/app/webrtc/test/fakevideotrackrenderer.h"
#include "talk/app/webrtc/videosourceinterface.h"
@@ -39,6 +42,7 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread.h"
+#include "webrtc/p2p/client/fakeportallocator.h"
using webrtc::DataChannelInterface;
using webrtc::DtlsIdentityStoreInterface;
@@ -47,17 +51,11 @@
using webrtc::PeerConnectionFactoryInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
-using webrtc::PortAllocatorFactoryInterface;
using webrtc::VideoSourceInterface;
using webrtc::VideoTrackInterface;
namespace {
-typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration>
- StunConfigurations;
-typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration>
- TurnConfigurations;
-
static const char kStunIceServer[] = "stun:stun.l.google.com:19302";
static const char kTurnIceServer[] = "turn:test%40hello.com@test.com:1234";
static const char kTurnIceServerWithTransport[] =
@@ -103,6 +101,9 @@
class PeerConnectionFactoryTest : public testing::Test {
void SetUp() {
+#ifdef WEBRTC_ANDROID
+ webrtc::InitializeAndroidObjects();
+#endif
factory_ = webrtc::CreatePeerConnectionFactory(rtc::Thread::Current(),
rtc::Thread::Current(),
NULL,
@@ -110,57 +111,58 @@
NULL);
ASSERT_TRUE(factory_.get() != NULL);
- allocator_factory_ = webrtc::FakePortAllocatorFactory::Create();
+ port_allocator_.reset(
+ new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
+ raw_port_allocator_ = port_allocator_.get();
}
protected:
- void VerifyStunConfigurations(StunConfigurations stun_config) {
- webrtc::FakePortAllocatorFactory* allocator =
- static_cast<webrtc::FakePortAllocatorFactory*>(
- allocator_factory_.get());
- ASSERT_TRUE(allocator != NULL);
- EXPECT_EQ(stun_config.size(), allocator->stun_configs().size());
- for (size_t i = 0; i < stun_config.size(); ++i) {
- EXPECT_EQ(stun_config[i].server.ToString(),
- allocator->stun_configs()[i].server.ToString());
- }
+ void VerifyStunServers(cricket::ServerAddresses stun_servers) {
+ EXPECT_EQ(stun_servers, raw_port_allocator_->stun_servers());
}
- void VerifyTurnConfigurations(TurnConfigurations turn_config) {
- webrtc::FakePortAllocatorFactory* allocator =
- static_cast<webrtc::FakePortAllocatorFactory*>(
- allocator_factory_.get());
- ASSERT_TRUE(allocator != NULL);
- EXPECT_EQ(turn_config.size(), allocator->turn_configs().size());
- for (size_t i = 0; i < turn_config.size(); ++i) {
- EXPECT_EQ(turn_config[i].server.ToString(),
- allocator->turn_configs()[i].server.ToString());
- EXPECT_EQ(turn_config[i].username, allocator->turn_configs()[i].username);
- EXPECT_EQ(turn_config[i].password, allocator->turn_configs()[i].password);
- EXPECT_EQ(turn_config[i].transport_type,
- allocator->turn_configs()[i].transport_type);
+ void VerifyTurnServers(std::vector<cricket::RelayServerConfig> turn_servers) {
+ EXPECT_EQ(turn_servers.size(), raw_port_allocator_->turn_servers().size());
+ for (size_t i = 0; i < turn_servers.size(); ++i) {
+ ASSERT_EQ(1u, turn_servers[i].ports.size());
+ EXPECT_EQ(1u, raw_port_allocator_->turn_servers()[i].ports.size());
+ EXPECT_EQ(
+ turn_servers[i].ports[0].address.ToString(),
+ raw_port_allocator_->turn_servers()[i].ports[0].address.ToString());
+ EXPECT_EQ(turn_servers[i].ports[0].proto,
+ raw_port_allocator_->turn_servers()[i].ports[0].proto);
+ EXPECT_EQ(turn_servers[i].credentials.username,
+ raw_port_allocator_->turn_servers()[i].credentials.username);
+ EXPECT_EQ(turn_servers[i].credentials.password,
+ raw_port_allocator_->turn_servers()[i].credentials.password);
}
}
rtc::scoped_refptr<PeerConnectionFactoryInterface> factory_;
NullPeerConnectionObserver observer_;
- rtc::scoped_refptr<PortAllocatorFactoryInterface> allocator_factory_;
+ rtc::scoped_ptr<cricket::FakePortAllocator> port_allocator_;
+ // Since the PC owns the port allocator after it's been initialized,
+ // this should only be used when known to be safe.
+ cricket::FakePortAllocator* raw_port_allocator_;
};
// Verify creation of PeerConnection using internal ADM, video factory and
// internal libjingle threads.
TEST(PeerConnectionFactoryTestInternal, CreatePCUsingInternalModules) {
+#ifdef WEBRTC_ANDROID
+ webrtc::InitializeAndroidObjects();
+#endif
+
rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
webrtc::CreatePeerConnectionFactory());
NullPeerConnectionObserver observer;
- webrtc::PeerConnectionInterface::IceServers servers;
+ webrtc::PeerConnectionInterface::RTCConfiguration config;
rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory->CreatePeerConnection(
- servers, nullptr, nullptr, dtls_identity_store.Pass(), &observer));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory->CreatePeerConnection(
+ config, nullptr, nullptr, std::move(dtls_identity_store), &observer));
EXPECT_TRUE(pc.get() != nullptr);
}
@@ -180,25 +182,22 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
- EXPECT_TRUE(pc.get() != NULL);
- StunConfigurations stun_configs;
- webrtc::PortAllocatorFactoryInterface::StunConfiguration stun1(
- "stun.l.google.com", 19302);
- stun_configs.push_back(stun1);
- VerifyStunConfigurations(stun_configs);
- TurnConfigurations turn_configs;
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn1(
- "test.com", 1234, "test@hello.com", kTurnPassword, "udp", false);
- turn_configs.push_back(turn1);
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn2(
- "hello.com", kDefaultStunPort, "test", kTurnPassword, "tcp", false);
- turn_configs.push_back(turn2);
- VerifyTurnConfigurations(turn_configs);
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, std::move(port_allocator_),
+ std::move(dtls_identity_store), &observer_));
+ ASSERT_TRUE(pc.get() != NULL);
+ cricket::ServerAddresses stun_servers;
+ rtc::SocketAddress stun1("stun.l.google.com", 19302);
+ stun_servers.insert(stun1);
+ VerifyStunServers(stun_servers);
+ std::vector<cricket::RelayServerConfig> turn_servers;
+ cricket::RelayServerConfig turn1("test.com", 1234, "test@hello.com",
+ kTurnPassword, cricket::PROTO_UDP, false);
+ turn_servers.push_back(turn1);
+ cricket::RelayServerConfig turn2("hello.com", kDefaultStunPort, "test",
+ kTurnPassword, cricket::PROTO_TCP, false);
+ turn_servers.push_back(turn2);
+ VerifyTurnServers(turn_servers);
}
// This test verifies creation of PeerConnection with valid STUN and TURN
@@ -213,63 +212,22 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
- EXPECT_TRUE(pc.get() != NULL);
- StunConfigurations stun_configs;
- webrtc::PortAllocatorFactoryInterface::StunConfiguration stun1(
- "stun.l.google.com", 19302);
- stun_configs.push_back(stun1);
- VerifyStunConfigurations(stun_configs);
- TurnConfigurations turn_configs;
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn1(
- "test.com", 1234, "test@hello.com", kTurnPassword, "udp", false);
- turn_configs.push_back(turn1);
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn2(
- "hello.com", kDefaultStunPort, "test", kTurnPassword, "tcp", false);
- turn_configs.push_back(turn2);
- VerifyTurnConfigurations(turn_configs);
-}
-
-// This test verifies creation of PeerConnection with valid STUN and TURN
-// configuration. Also verifies the URL's parsed correctly as expected.
-// This version doesn't use RTCConfiguration.
-// TODO(mallinath) - Remove this method after clients start using RTCConfig.
-TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersOldSignature) {
- webrtc::PeerConnectionInterface::IceServers ice_servers;
- webrtc::PeerConnectionInterface::IceServer ice_server;
- ice_server.uri = kStunIceServer;
- ice_servers.push_back(ice_server);
- ice_server.uri = kTurnIceServer;
- ice_server.password = kTurnPassword;
- ice_servers.push_back(ice_server);
- ice_server.uri = kTurnIceServerWithTransport;
- ice_server.password = kTurnPassword;
- ice_servers.push_back(ice_server);
- rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
- new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(ice_servers, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
- EXPECT_TRUE(pc.get() != NULL);
- StunConfigurations stun_configs;
- webrtc::PortAllocatorFactoryInterface::StunConfiguration stun1(
- "stun.l.google.com", 19302);
- stun_configs.push_back(stun1);
- VerifyStunConfigurations(stun_configs);
- TurnConfigurations turn_configs;
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn1(
- "test.com", 1234, "test@hello.com", kTurnPassword, "udp", false);
- turn_configs.push_back(turn1);
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn2(
- "hello.com", kDefaultStunPort, "test", kTurnPassword, "tcp", false);
- turn_configs.push_back(turn2);
- VerifyTurnConfigurations(turn_configs);
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, std::move(port_allocator_),
+ std::move(dtls_identity_store), &observer_));
+ ASSERT_TRUE(pc.get() != NULL);
+ cricket::ServerAddresses stun_servers;
+ rtc::SocketAddress stun1("stun.l.google.com", 19302);
+ stun_servers.insert(stun1);
+ VerifyStunServers(stun_servers);
+ std::vector<cricket::RelayServerConfig> turn_servers;
+ cricket::RelayServerConfig turn1("test.com", 1234, "test@hello.com",
+ kTurnPassword, cricket::PROTO_UDP, false);
+ turn_servers.push_back(turn1);
+ cricket::RelayServerConfig turn2("hello.com", kDefaultStunPort, "test",
+ kTurnPassword, cricket::PROTO_TCP, false);
+ turn_servers.push_back(turn2);
+ VerifyTurnServers(turn_servers);
}
TEST_F(PeerConnectionFactoryTest, CreatePCUsingNoUsernameInUri) {
@@ -283,17 +241,15 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
- EXPECT_TRUE(pc.get() != NULL);
- TurnConfigurations turn_configs;
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn(
- "test.com", 1234, kTurnUsername, kTurnPassword, "udp", false);
- turn_configs.push_back(turn);
- VerifyTurnConfigurations(turn_configs);
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, std::move(port_allocator_),
+ std::move(dtls_identity_store), &observer_));
+ ASSERT_TRUE(pc.get() != NULL);
+ std::vector<cricket::RelayServerConfig> turn_servers;
+ cricket::RelayServerConfig turn("test.com", 1234, kTurnUsername,
+ kTurnPassword, cricket::PROTO_UDP, false);
+ turn_servers.push_back(turn);
+ VerifyTurnServers(turn_servers);
}
// This test verifies the PeerConnection created properly with TURN url which
@@ -306,17 +262,15 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
- EXPECT_TRUE(pc.get() != NULL);
- TurnConfigurations turn_configs;
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn(
- "hello.com", kDefaultStunPort, "test", kTurnPassword, "tcp", false);
- turn_configs.push_back(turn);
- VerifyTurnConfigurations(turn_configs);
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, std::move(port_allocator_),
+ std::move(dtls_identity_store), &observer_));
+ ASSERT_TRUE(pc.get() != NULL);
+ std::vector<cricket::RelayServerConfig> turn_servers;
+ cricket::RelayServerConfig turn("hello.com", kDefaultStunPort, "test",
+ kTurnPassword, cricket::PROTO_TCP, false);
+ turn_servers.push_back(turn);
+ VerifyTurnServers(turn_servers);
}
TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) {
@@ -333,25 +287,23 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
- EXPECT_TRUE(pc.get() != NULL);
- TurnConfigurations turn_configs;
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn1(
- "hello.com", kDefaultStunTlsPort, "test", kTurnPassword, "tcp", true);
- turn_configs.push_back(turn1);
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, std::move(port_allocator_),
+ std::move(dtls_identity_store), &observer_));
+ ASSERT_TRUE(pc.get() != NULL);
+ std::vector<cricket::RelayServerConfig> turn_servers;
+ cricket::RelayServerConfig turn1("hello.com", kDefaultStunTlsPort, "test",
+ kTurnPassword, cricket::PROTO_TCP, true);
+ turn_servers.push_back(turn1);
// TURNS with transport param should be default to tcp.
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn2(
- "hello.com", 443, "test_no_transport", kTurnPassword, "tcp", true);
- turn_configs.push_back(turn2);
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn3(
- "hello.com", kDefaultStunTlsPort, "test_no_transport",
- kTurnPassword, "tcp", true);
- turn_configs.push_back(turn3);
- VerifyTurnConfigurations(turn_configs);
+ cricket::RelayServerConfig turn2("hello.com", 443, "test_no_transport",
+ kTurnPassword, cricket::PROTO_TCP, true);
+ turn_servers.push_back(turn2);
+ cricket::RelayServerConfig turn3("hello.com", kDefaultStunTlsPort,
+ "test_no_transport", kTurnPassword,
+ cricket::PROTO_TCP, true);
+ turn_servers.push_back(turn3);
+ VerifyTurnServers(turn_servers);
}
TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) {
@@ -370,32 +322,26 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
- EXPECT_TRUE(pc.get() != NULL);
- StunConfigurations stun_configs;
- webrtc::PortAllocatorFactoryInterface::StunConfiguration stun1(
- "1.2.3.4", 1234);
- stun_configs.push_back(stun1);
- webrtc::PortAllocatorFactoryInterface::StunConfiguration stun2(
- "1.2.3.4", 3478);
- stun_configs.push_back(stun2); // Default port
- webrtc::PortAllocatorFactoryInterface::StunConfiguration stun3(
- "2401:fa00:4::", 1234);
- stun_configs.push_back(stun3);
- webrtc::PortAllocatorFactoryInterface::StunConfiguration stun4(
- "2401:fa00:4::", 3478);
- stun_configs.push_back(stun4); // Default port
- VerifyStunConfigurations(stun_configs);
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, std::move(port_allocator_),
+ std::move(dtls_identity_store), &observer_));
+ ASSERT_TRUE(pc.get() != NULL);
+ cricket::ServerAddresses stun_servers;
+ rtc::SocketAddress stun1("1.2.3.4", 1234);
+ stun_servers.insert(stun1);
+ rtc::SocketAddress stun2("1.2.3.4", 3478);
+ stun_servers.insert(stun2); // Default port
+ rtc::SocketAddress stun3("2401:fa00:4::", 1234);
+ stun_servers.insert(stun3);
+ rtc::SocketAddress stun4("2401:fa00:4::", 3478);
+ stun_servers.insert(stun4); // Default port
+ VerifyStunServers(stun_servers);
- TurnConfigurations turn_configs;
- webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn1(
- "2401:fa00:4::", 1234, "test", kTurnPassword, "udp", false);
- turn_configs.push_back(turn1);
- VerifyTurnConfigurations(turn_configs);
+ std::vector<cricket::RelayServerConfig> turn_servers;
+ cricket::RelayServerConfig turn1("2401:fa00:4::", 1234, "test", kTurnPassword,
+ cricket::PROTO_UDP, false);
+ turn_servers.push_back(turn1);
+ VerifyTurnServers(turn_servers);
}
// This test verifies the captured stream is rendered locally using a
diff --git a/talk/app/webrtc/peerconnectionfactoryproxy.h b/talk/app/webrtc/peerconnectionfactoryproxy.h
index 5e924df..714ce6b 100644
--- a/talk/app/webrtc/peerconnectionfactoryproxy.h
+++ b/talk/app/webrtc/peerconnectionfactoryproxy.h
@@ -29,6 +29,7 @@
#define TALK_APP_WEBRTC_PEERCONNECTIONFACTORYPROXY_H_
#include <string>
+#include <utility>
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/proxy.h"
@@ -38,17 +39,17 @@
BEGIN_PROXY_MAP(PeerConnectionFactory)
PROXY_METHOD1(void, SetOptions, const Options&)
- // Can't use PROXY_METHOD5 because scoped_ptr must be Pass()ed.
+ // Can't use PROXY_METHOD5 because scoped_ptr must be moved.
// TODO(tommi,hbos): Use of templates to support scoped_ptr?
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& a1,
const MediaConstraintsInterface* a2,
- PortAllocatorFactoryInterface* a3,
+ rtc::scoped_ptr<cricket::PortAllocator> a3,
rtc::scoped_ptr<DtlsIdentityStoreInterface> a4,
PeerConnectionObserver* a5) override {
return owner_thread_->Invoke<rtc::scoped_refptr<PeerConnectionInterface>>(
rtc::Bind(&PeerConnectionFactoryProxy::CreatePeerConnection_ot, this,
- a1, a2, a3, a4.release(), a5));
+ a1, a2, a3.release(), a4.release(), a5));
}
PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
CreateLocalMediaStream, const std::string&)
@@ -70,11 +71,13 @@
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection_ot(
const PeerConnectionInterface::RTCConfiguration& a1,
const MediaConstraintsInterface* a2,
- PortAllocatorFactoryInterface* a3,
+ cricket::PortAllocator* a3,
DtlsIdentityStoreInterface* a4,
PeerConnectionObserver* a5) {
+ rtc::scoped_ptr<cricket::PortAllocator> ptr_a3(a3);
rtc::scoped_ptr<DtlsIdentityStoreInterface> ptr_a4(a4);
- return c_->CreatePeerConnection(a1, a2, a3, ptr_a4.Pass(), a5);
+ return c_->CreatePeerConnection(a1, a2, std::move(ptr_a3),
+ std::move(ptr_a4), a5);
}
END_PROXY()
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index 77caa9d..b9afbad 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -69,6 +69,7 @@
#define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
#include <string>
+#include <utility>
#include <vector>
#include "talk/app/webrtc/datachannelinterface.h"
@@ -86,6 +87,7 @@
#include "webrtc/base/rtccertificate.h"
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/socketaddress.h"
+#include "webrtc/p2p/base/portallocator.h"
namespace rtc {
class SSLIdentity;
@@ -93,7 +95,6 @@
}
namespace cricket {
-class PortAllocator;
class WebRtcVideoDecoderFactory;
class WebRtcVideoEncoderFactory;
}
@@ -248,28 +249,27 @@
// TODO(pthatcher): Rename this ice_servers, but update Chromium
// at the same time.
IceServers servers;
- // A localhost candidate is signaled whenever a candidate with the any
- // address is allocated.
- bool enable_localhost_ice_candidate;
BundlePolicy bundle_policy;
RtcpMuxPolicy rtcp_mux_policy;
TcpCandidatePolicy tcp_candidate_policy;
int audio_jitter_buffer_max_packets;
bool audio_jitter_buffer_fast_accelerate;
- int ice_connection_receiving_timeout;
+ int ice_connection_receiving_timeout; // ms
+ int ice_backup_candidate_pair_ping_interval; // ms
ContinualGatheringPolicy continual_gathering_policy;
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
-
+ bool disable_prerenderer_smoothing;
RTCConfiguration()
: type(kAll),
- enable_localhost_ice_candidate(false),
bundle_policy(kBundlePolicyBalanced),
rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
tcp_candidate_policy(kTcpCandidatePolicyEnabled),
audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets),
audio_jitter_buffer_fast_accelerate(false),
ice_connection_receiving_timeout(kUndefined),
- continual_gathering_policy(GATHER_ONCE) {}
+ ice_backup_candidate_pair_ping_interval(kUndefined),
+ continual_gathering_policy(GATHER_ONCE),
+ disable_prerenderer_smoothing(false) {}
};
struct RTCOfferAnswerOptions {
@@ -337,6 +337,15 @@
AudioTrackInterface* track) = 0;
// TODO(deadbeef): Make these pure virtual once all subclasses implement them.
+ // |kind| must be "audio" or "video".
+ // |stream_id| is used to populate the msid attribute; if empty, one will
+ // be generated automatically.
+ virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
+ const std::string& kind,
+ const std::string& stream_id) {
+ return rtc::scoped_refptr<RtpSenderInterface>();
+ }
+
virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
const {
return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
@@ -480,51 +489,6 @@
~PeerConnectionObserver() {}
};
-// Factory class used for creating cricket::PortAllocator that is used
-// for ICE negotiation.
-class PortAllocatorFactoryInterface : public rtc::RefCountInterface {
- public:
- struct StunConfiguration {
- StunConfiguration(const std::string& address, int port)
- : server(address, port) {}
- // STUN server address and port.
- rtc::SocketAddress server;
- };
-
- struct TurnConfiguration {
- TurnConfiguration(const std::string& address,
- int port,
- const std::string& username,
- const std::string& password,
- const std::string& transport_type,
- bool secure)
- : server(address, port),
- username(username),
- password(password),
- transport_type(transport_type),
- secure(secure) {}
- rtc::SocketAddress server;
- std::string username;
- std::string password;
- std::string transport_type;
- bool secure;
- };
-
- virtual cricket::PortAllocator* CreatePortAllocator(
- const std::vector<StunConfiguration>& stun_servers,
- const std::vector<TurnConfiguration>& turn_configurations) = 0;
-
- // TODO(phoglund): Make pure virtual when Chrome's factory implements this.
- // After this method is called, the port allocator should consider loopback
- // network interfaces as well.
- virtual void SetNetworkIgnoreMask(int network_ignore_mask) {
- }
-
- protected:
- PortAllocatorFactoryInterface() {}
- ~PortAllocatorFactoryInterface() {}
-};
-
// PeerConnectionFactoryInterface is the factory interface use for creating
// PeerConnection, MediaStream and media tracks.
// PeerConnectionFactoryInterface will create required libjingle threads,
@@ -532,19 +496,18 @@
// If an application decides to provide its own threads and network
// implementation of these classes it should use the alternate
// CreatePeerConnectionFactory method which accepts threads as input and use the
-// CreatePeerConnection version that takes a PortAllocatorFactoryInterface as
+// CreatePeerConnection version that takes a PortAllocator as an
// argument.
class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
public:
class Options {
public:
- Options() :
- disable_encryption(false),
- disable_sctp_data_channels(false),
- disable_network_monitor(false),
- network_ignore_mask(rtc::kDefaultNetworkIgnoreMask),
- ssl_max_version(rtc::SSL_PROTOCOL_DTLS_10) {
- }
+ Options()
+ : disable_encryption(false),
+ disable_sctp_data_channels(false),
+ disable_network_monitor(false),
+ network_ignore_mask(rtc::kDefaultNetworkIgnoreMask),
+ ssl_max_version(rtc::SSL_PROTOCOL_DTLS_12) {}
bool disable_encryption;
bool disable_sctp_data_channels;
bool disable_network_monitor;
@@ -562,31 +525,12 @@
virtual void SetOptions(const Options& options) = 0;
- virtual rtc::scoped_refptr<PeerConnectionInterface>
- CreatePeerConnection(
- const PeerConnectionInterface::RTCConfiguration& configuration,
- const MediaConstraintsInterface* constraints,
- PortAllocatorFactoryInterface* allocator_factory,
- rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
- PeerConnectionObserver* observer) = 0;
-
- // TODO(hbos): Remove below version after clients are updated to above method.
- // In latest W3C WebRTC draft, PC constructor will take RTCConfiguration,
- // and not IceServers. RTCConfiguration is made up of ice servers and
- // ice transport type.
- // http://dev.w3.org/2011/webrtc/editor/webrtc.html
- inline rtc::scoped_refptr<PeerConnectionInterface>
- CreatePeerConnection(
- const PeerConnectionInterface::IceServers& servers,
- const MediaConstraintsInterface* constraints,
- PortAllocatorFactoryInterface* allocator_factory,
- rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
- PeerConnectionObserver* observer) {
- PeerConnectionInterface::RTCConfiguration rtc_config;
- rtc_config.servers = servers;
- return CreatePeerConnection(rtc_config, constraints, allocator_factory,
- dtls_identity_store.Pass(), observer);
- }
+ virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
+ const PeerConnectionInterface::RTCConfiguration& configuration,
+ const MediaConstraintsInterface* constraints,
+ rtc::scoped_ptr<cricket::PortAllocator> allocator,
+ rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+ PeerConnectionObserver* observer) = 0;
virtual rtc::scoped_refptr<MediaStreamInterface>
CreateLocalMediaStream(const std::string& label) = 0;
diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc
index 63163fd..c3789b7 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -26,9 +26,9 @@
*/
#include <string>
+#include <utility>
#include "talk/app/webrtc/audiotrack.h"
-#include "talk/app/webrtc/fakeportallocatorfactory.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/mediastream.h"
#include "talk/app/webrtc/mediastreaminterface.h"
@@ -37,6 +37,9 @@
#include "talk/app/webrtc/rtpreceiverinterface.h"
#include "talk/app/webrtc/rtpsenderinterface.h"
#include "talk/app/webrtc/streamcollection.h"
+#ifdef WEBRTC_ANDROID
+#include "talk/app/webrtc/test/androidtestinitializer.h"
+#endif
#include "talk/app/webrtc/test/fakeconstraints.h"
#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
#include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
@@ -52,6 +55,7 @@
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/thread.h"
+#include "webrtc/p2p/client/fakeportallocator.h"
static const char kStreamLabel1[] = "local_stream_1";
static const char kStreamLabel2[] = "local_stream_2";
@@ -258,7 +262,6 @@
using webrtc::DataBuffer;
using webrtc::DataChannelInterface;
using webrtc::FakeConstraints;
-using webrtc::FakePortAllocatorFactory;
using webrtc::IceCandidateInterface;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStream;
@@ -270,7 +273,6 @@
using webrtc::MockStatsObserver;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
-using webrtc::PortAllocatorFactoryInterface;
using webrtc::RtpReceiverInterface;
using webrtc::RtpSenderInterface;
using webrtc::SdpParseError;
@@ -515,6 +517,12 @@
class PeerConnectionInterfaceTest : public testing::Test {
protected:
+ PeerConnectionInterfaceTest() {
+#ifdef WEBRTC_ANDROID
+ webrtc::InitializeAndroidObjects();
+#endif
+ }
+
virtual void SetUp() {
pc_factory_ = webrtc::CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(), NULL, NULL,
@@ -533,15 +541,17 @@
void CreatePeerConnection(const std::string& uri,
const std::string& password,
webrtc::MediaConstraintsInterface* constraints) {
+ PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
- PeerConnectionInterface::IceServers servers;
if (!uri.empty()) {
server.uri = uri;
server.password = password;
- servers.push_back(server);
+ config.servers.push_back(server);
}
- port_allocator_factory_ = FakePortAllocatorFactory::Create();
+ rtc::scoped_ptr<cricket::FakePortAllocator> port_allocator(
+ new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
+ port_allocator_ = port_allocator.get();
// DTLS does not work in a loopback call, so is disabled for most of the
// tests in this file. We only create a FakeIdentityService if the test
@@ -562,52 +572,47 @@
nullptr) && dtls) {
dtls_identity_store.reset(new FakeDtlsIdentityStore());
}
- pc_ = pc_factory_->CreatePeerConnection(servers, constraints,
- port_allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_);
+ pc_ = pc_factory_->CreatePeerConnection(
+ config, constraints, std::move(port_allocator),
+ std::move(dtls_identity_store), &observer_);
ASSERT_TRUE(pc_.get() != NULL);
observer_.SetPeerConnectionInterface(pc_.get());
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePeerConnectionExpectFail(const std::string& uri) {
+ PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
- PeerConnectionInterface::IceServers servers;
server.uri = uri;
- servers.push_back(server);
+ config.servers.push_back(server);
- scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store;
- port_allocator_factory_ = FakePortAllocatorFactory::Create();
scoped_refptr<PeerConnectionInterface> pc;
- pc = pc_factory_->CreatePeerConnection(
- servers, nullptr, port_allocator_factory_.get(),
- dtls_identity_store.Pass(), &observer_);
- ASSERT_EQ(nullptr, pc);
+ pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr, nullptr,
+ &observer_);
+ EXPECT_EQ(nullptr, pc);
}
void CreatePeerConnectionWithDifferentConfigurations() {
CreatePeerConnection(kStunAddressOnly, "", NULL);
- EXPECT_EQ(1u, port_allocator_factory_->stun_configs().size());
- EXPECT_EQ(0u, port_allocator_factory_->turn_configs().size());
- EXPECT_EQ("address",
- port_allocator_factory_->stun_configs()[0].server.hostname());
+ EXPECT_EQ(1u, port_allocator_->stun_servers().size());
+ EXPECT_EQ(0u, port_allocator_->turn_servers().size());
+ EXPECT_EQ("address", port_allocator_->stun_servers().begin()->hostname());
EXPECT_EQ(kDefaultStunPort,
- port_allocator_factory_->stun_configs()[0].server.port());
+ port_allocator_->stun_servers().begin()->port());
CreatePeerConnectionExpectFail(kStunInvalidPort);
CreatePeerConnectionExpectFail(kStunAddressPortAndMore1);
CreatePeerConnectionExpectFail(kStunAddressPortAndMore2);
CreatePeerConnection(kTurnIceServerUri, kTurnPassword, NULL);
- EXPECT_EQ(0u, port_allocator_factory_->stun_configs().size());
- EXPECT_EQ(1u, port_allocator_factory_->turn_configs().size());
+ EXPECT_EQ(0u, port_allocator_->stun_servers().size());
+ EXPECT_EQ(1u, port_allocator_->turn_servers().size());
EXPECT_EQ(kTurnUsername,
- port_allocator_factory_->turn_configs()[0].username);
+ port_allocator_->turn_servers()[0].credentials.username);
EXPECT_EQ(kTurnPassword,
- port_allocator_factory_->turn_configs()[0].password);
+ port_allocator_->turn_servers()[0].credentials.password);
EXPECT_EQ(kTurnHostname,
- port_allocator_factory_->turn_configs()[0].server.hostname());
+ port_allocator_->turn_servers()[0].ports[0].address.hostname());
}
void ReleasePeerConnection() {
@@ -926,7 +931,7 @@
ASSERT_TRUE(stream->AddTrack(video_track));
}
- scoped_refptr<FakePortAllocatorFactory> port_allocator_factory_;
+ cricket::FakePortAllocator* port_allocator_ = nullptr;
scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
scoped_refptr<PeerConnectionInterface> pc_;
MockPeerConnectionObserver observer_;
@@ -1156,6 +1161,64 @@
EXPECT_NE(audio_ssrc, video_ssrc);
}
+// Test that it's possible to call AddTrack on a MediaStream after adding
+// the stream to a PeerConnection.
+// TODO(deadbeef): Remove this test once this behavior is no longer supported.
+TEST_F(PeerConnectionInterfaceTest, AddTrackAfterAddStream) {
+ CreatePeerConnection();
+ // Create audio stream and add to PeerConnection.
+ AddVoiceStream(kStreamLabel1);
+ MediaStreamInterface* stream = pc_->local_streams()->at(0);
+
+ // Add video track to the audio-only stream.
+ scoped_refptr<VideoTrackInterface> video_track(
+ pc_factory_->CreateVideoTrack("video_label", nullptr));
+ stream->AddTrack(video_track.get());
+
+ scoped_ptr<SessionDescriptionInterface> offer;
+ ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
+
+ const cricket::MediaContentDescription* video_desc =
+ cricket::GetFirstVideoContentDescription(offer->description());
+ EXPECT_TRUE(video_desc != nullptr);
+}
+
+// Test that it's possible to call RemoveTrack on a MediaStream after adding
+// the stream to a PeerConnection.
+// TODO(deadbeef): Remove this test once this behavior is no longer supported.
+TEST_F(PeerConnectionInterfaceTest, RemoveTrackAfterAddStream) {
+ CreatePeerConnection();
+ // Create audio/video stream and add to PeerConnection.
+ AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
+ MediaStreamInterface* stream = pc_->local_streams()->at(0);
+
+ // Remove the video track.
+ stream->RemoveTrack(stream->GetVideoTracks()[0]);
+
+ scoped_ptr<SessionDescriptionInterface> offer;
+ ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
+
+ const cricket::MediaContentDescription* video_desc =
+ cricket::GetFirstVideoContentDescription(offer->description());
+ EXPECT_TRUE(video_desc == nullptr);
+}
+
+// Test creating a sender with a stream ID, and ensure the ID is populated
+// in the offer.
+TEST_F(PeerConnectionInterfaceTest, CreateSenderWithStream) {
+ CreatePeerConnection();
+ pc_->CreateSender("video", kStreamLabel1);
+
+ scoped_ptr<SessionDescriptionInterface> offer;
+ ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
+
+ const cricket::MediaContentDescription* video_desc =
+ cricket::GetFirstVideoContentDescription(offer->description());
+ ASSERT_TRUE(video_desc != nullptr);
+ ASSERT_EQ(1u, video_desc->streams().size());
+ EXPECT_EQ(kStreamLabel1, video_desc->streams()[0].sync_label);
+}
+
// Test that we can specify a certain track that we want statistics about.
TEST_F(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
InitiateCall();
@@ -1660,6 +1723,22 @@
ASSERT_EQ(cricket::MD_INACTIVE, audio_desc->direction());
}
+// Test that we can use SetConfiguration to change the ICE servers of the
+// PortAllocator.
+TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
+ CreatePeerConnection();
+
+ PeerConnectionInterface::RTCConfiguration config;
+ PeerConnectionInterface::IceServer server;
+ server.uri = "stun:test_hostname";
+ config.servers.push_back(server);
+ EXPECT_TRUE(pc_->SetConfiguration(config));
+
+ EXPECT_EQ(1u, port_allocator_->stun_servers().size());
+ EXPECT_EQ("test_hostname",
+ port_allocator_->stun_servers().begin()->hostname());
+}
+
// Test that PeerConnection::Close changes the states to closed and all remote
// tracks change state to ended.
TEST_F(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
@@ -1977,6 +2056,28 @@
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
+// This tests that when setting a new description, the old default tracks are
+// not destroyed and recreated.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
+TEST_F(PeerConnectionInterfaceTest, DefaultTracksNotDestroyedAndRecreated) {
+ FakeConstraints constraints;
+ constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
+ true);
+ CreatePeerConnection(&constraints);
+ CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
+
+ ASSERT_EQ(1u, observer_.remote_streams()->count());
+ MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
+ ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
+
+ // Set the track to "disabled", then set a new description and ensure the
+ // track is still disabled, which ensures it hasn't been recreated.
+ remote_stream->GetAudioTracks()[0]->set_enabled(false);
+ CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
+ ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
+ EXPECT_FALSE(remote_stream->GetAudioTracks()[0]->enabled());
+}
+
// This tests that a default MediaStream is not created if a remote session
// description is updated to not have any MediaStreams.
TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
@@ -2020,8 +2121,10 @@
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
// Remove an audio and video track.
+ pc_->RemoveStream(reference_collection_->at(0));
rtc::scoped_ptr<SessionDescriptionInterface> desc_2;
CreateSessionDescriptionAndReference(1, 1, desc_2.accept());
+ pc_->AddStream(reference_collection_->at(0));
EXPECT_TRUE(DoSetLocalDescription(desc_2.release()));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
@@ -2220,7 +2323,9 @@
EXPECT_FALSE(options.has_video());
EXPECT_TRUE(options.bundle_enabled);
EXPECT_TRUE(options.vad_enabled);
- EXPECT_FALSE(options.transport_options.ice_restart);
+ EXPECT_FALSE(options.audio_transport_options.ice_restart);
+ EXPECT_FALSE(options.video_transport_options.ice_restart);
+ EXPECT_FALSE(options.data_transport_options.ice_restart);
}
// Test that a correct MediaSessionOptions is created for an offer if
@@ -2255,18 +2360,22 @@
// Test that a correct MediaSessionOptions is created to restart ice if
// IceRestart is set. It also tests that subsequent MediaSessionOptions don't
-// have |transport_options.ice_restart| set.
+// have |audio_transport_options.ice_restart| etc. set.
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithIceRestart) {
RTCOfferAnswerOptions rtc_options;
rtc_options.ice_restart = true;
cricket::MediaSessionOptions options;
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
- EXPECT_TRUE(options.transport_options.ice_restart);
+ EXPECT_TRUE(options.audio_transport_options.ice_restart);
+ EXPECT_TRUE(options.video_transport_options.ice_restart);
+ EXPECT_TRUE(options.data_transport_options.ice_restart);
rtc_options = RTCOfferAnswerOptions();
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
- EXPECT_FALSE(options.transport_options.ice_restart);
+ EXPECT_FALSE(options.audio_transport_options.ice_restart);
+ EXPECT_FALSE(options.video_transport_options.ice_restart);
+ EXPECT_FALSE(options.data_transport_options.ice_restart);
}
// Test that the MediaConstraints in an answer don't affect if audio and video
diff --git a/talk/app/webrtc/peerconnectionproxy.h b/talk/app/webrtc/peerconnectionproxy.h
index d207fbb..3c983d7 100644
--- a/talk/app/webrtc/peerconnectionproxy.h
+++ b/talk/app/webrtc/peerconnectionproxy.h
@@ -43,6 +43,10 @@
PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>,
CreateDtmfSender, AudioTrackInterface*)
+ PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
+ CreateSender,
+ const std::string&,
+ const std::string&)
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>,
GetSenders)
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpReceiverInterface>>,
diff --git a/talk/app/webrtc/portallocatorfactory.cc b/talk/app/webrtc/portallocatorfactory.cc
index bd6cacc..64d714c 100644
--- a/talk/app/webrtc/portallocatorfactory.cc
+++ b/talk/app/webrtc/portallocatorfactory.cc
@@ -1,6 +1,6 @@
/*
* libjingle
- * Copyright 2004--2011 Google Inc.
+ * Copyright 2011 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
@@ -24,69 +24,7 @@
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
+// TODO(deadbeef): Remove this file once chromium build files no longer
+// reference it.
#include "talk/app/webrtc/portallocatorfactory.h"
-
-#include "webrtc/p2p/base/basicpacketsocketfactory.h"
-#include "webrtc/p2p/client/basicportallocator.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/network.h"
-#include "webrtc/base/thread.h"
-
-namespace webrtc {
-
-using rtc::scoped_ptr;
-
-rtc::scoped_refptr<PortAllocatorFactoryInterface>
-PortAllocatorFactory::Create(
- rtc::Thread* worker_thread) {
- rtc::RefCountedObject<PortAllocatorFactory>* allocator =
- new rtc::RefCountedObject<PortAllocatorFactory>(worker_thread);
- return allocator;
-}
-
-PortAllocatorFactory::PortAllocatorFactory(rtc::Thread* worker_thread)
- : network_manager_(new rtc::BasicNetworkManager()),
- socket_factory_(new rtc::BasicPacketSocketFactory(worker_thread)) {
-}
-
-PortAllocatorFactory::~PortAllocatorFactory() {}
-
-void PortAllocatorFactory::SetNetworkIgnoreMask(int network_ignore_mask) {
- network_manager_->set_network_ignore_mask(network_ignore_mask);
-}
-
-cricket::PortAllocator* PortAllocatorFactory::CreatePortAllocator(
- const std::vector<StunConfiguration>& stun,
- const std::vector<TurnConfiguration>& turn) {
- cricket::ServerAddresses stun_hosts;
- typedef std::vector<StunConfiguration>::const_iterator StunIt;
- for (StunIt stun_it = stun.begin(); stun_it != stun.end(); ++stun_it) {
- stun_hosts.insert(stun_it->server);
- }
-
- scoped_ptr<cricket::BasicPortAllocator> allocator(
- new cricket::BasicPortAllocator(
- network_manager_.get(), socket_factory_.get(), stun_hosts));
-
- for (size_t i = 0; i < turn.size(); ++i) {
- cricket::RelayCredentials credentials(turn[i].username, turn[i].password);
- cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
- cricket::ProtocolType protocol;
- if (cricket::StringToProto(turn[i].transport_type.c_str(), &protocol)) {
- relay_server.ports.push_back(cricket::ProtocolAddress(
- turn[i].server, protocol, turn[i].secure));
- relay_server.credentials = credentials;
- // First in the list gets highest priority.
- relay_server.priority = static_cast<int>(turn.size() - i - 1);
- allocator->AddRelay(relay_server);
- } else {
- LOG(LS_WARNING) << "Ignoring TURN server " << turn[i].server << ". "
- << "Reason= Incorrect " << turn[i].transport_type
- << " transport parameter.";
- }
- }
- return allocator.release();
-}
-
-} // namespace webrtc
diff --git a/talk/app/webrtc/portallocatorfactory.h b/talk/app/webrtc/portallocatorfactory.h
index 83376d0..bb6cf47 100644
--- a/talk/app/webrtc/portallocatorfactory.h
+++ b/talk/app/webrtc/portallocatorfactory.h
@@ -24,49 +24,10 @@
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-
-// This file defines the default implementation of
-// PortAllocatorFactoryInterface.
-// This implementation creates instances of cricket::HTTPPortAllocator and uses
-// the BasicNetworkManager and BasicPacketSocketFactory.
+// TODO(deadbeef): Remove this file once chromium build files no longer
+// reference it.
#ifndef TALK_APP_WEBRTC_PORTALLOCATORFACTORY_H_
#define TALK_APP_WEBRTC_PORTALLOCATORFACTORY_H_
-#include "talk/app/webrtc/peerconnectioninterface.h"
-#include "webrtc/base/scoped_ptr.h"
-
-namespace cricket {
-class PortAllocator;
-}
-
-namespace rtc {
-class BasicNetworkManager;
-class BasicPacketSocketFactory;
-}
-
-namespace webrtc {
-
-class PortAllocatorFactory : public PortAllocatorFactoryInterface {
- public:
- static rtc::scoped_refptr<PortAllocatorFactoryInterface> Create(
- rtc::Thread* worker_thread);
-
- virtual cricket::PortAllocator* CreatePortAllocator(
- const std::vector<StunConfiguration>& stun,
- const std::vector<TurnConfiguration>& turn);
-
- virtual void SetNetworkIgnoreMask(int network_ignore_mask);
-
- protected:
- explicit PortAllocatorFactory(rtc::Thread* worker_thread);
- ~PortAllocatorFactory();
-
- private:
- rtc::scoped_ptr<rtc::BasicNetworkManager> network_manager_;
- rtc::scoped_ptr<rtc::BasicPacketSocketFactory> socket_factory_;
-};
-
-} // namespace webrtc
-
#endif // TALK_APP_WEBRTC_PORTALLOCATORFACTORY_H_
diff --git a/talk/app/webrtc/remoteaudiosource.cc b/talk/app/webrtc/remoteaudiosource.cc
index 41f3d87..e904dd9 100644
--- a/talk/app/webrtc/remoteaudiosource.cc
+++ b/talk/app/webrtc/remoteaudiosource.cc
@@ -29,44 +29,148 @@
#include <algorithm>
#include <functional>
+#include <utility>
+#include "talk/app/webrtc/mediastreamprovider.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
+#include "webrtc/base/thread.h"
namespace webrtc {
-rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create() {
- return new rtc::RefCountedObject<RemoteAudioSource>();
+class RemoteAudioSource::MessageHandler : public rtc::MessageHandler {
+ public:
+ explicit MessageHandler(RemoteAudioSource* source) : source_(source) {}
+
+ private:
+ ~MessageHandler() override {}
+
+ void OnMessage(rtc::Message* msg) override {
+ source_->OnMessage(msg);
+ delete this;
+ }
+
+ const rtc::scoped_refptr<RemoteAudioSource> source_;
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MessageHandler);
+};
+
+class RemoteAudioSource::Sink : public AudioSinkInterface {
+ public:
+ explicit Sink(RemoteAudioSource* source) : source_(source) {}
+ ~Sink() override { source_->OnAudioProviderGone(); }
+
+ private:
+ void OnData(const AudioSinkInterface::Data& audio) override {
+ if (source_)
+ source_->OnData(audio);
+ }
+
+ const rtc::scoped_refptr<RemoteAudioSource> source_;
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sink);
+};
+
+rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create(
+ uint32_t ssrc,
+ AudioProviderInterface* provider) {
+ rtc::scoped_refptr<RemoteAudioSource> ret(
+ new rtc::RefCountedObject<RemoteAudioSource>());
+ ret->Initialize(ssrc, provider);
+ return ret;
}
-RemoteAudioSource::RemoteAudioSource() {
+RemoteAudioSource::RemoteAudioSource()
+ : main_thread_(rtc::Thread::Current()),
+ state_(MediaSourceInterface::kLive) {
+ RTC_DCHECK(main_thread_);
}
RemoteAudioSource::~RemoteAudioSource() {
- ASSERT(audio_observers_.empty());
+ RTC_DCHECK(main_thread_->IsCurrent());
+ RTC_DCHECK(audio_observers_.empty());
+ RTC_DCHECK(sinks_.empty());
}
-MediaSourceInterface::SourceState RemoteAudioSource::state() const {
- return MediaSourceInterface::kLive;
-}
-
-void RemoteAudioSource::SetVolume(double volume) {
- ASSERT(volume >= 0 && volume <= 10);
- for (AudioObserverList::iterator it = audio_observers_.begin();
- it != audio_observers_.end(); ++it) {
- (*it)->OnSetVolume(volume);
+void RemoteAudioSource::Initialize(uint32_t ssrc,
+ AudioProviderInterface* provider) {
+ RTC_DCHECK(main_thread_->IsCurrent());
+ // To make sure we always get notified when the provider goes out of scope,
+ // we register for callbacks here and not on demand in AddSink.
+ if (provider) { // May be null in tests.
+ provider->SetRawAudioSink(
+ ssrc, rtc::scoped_ptr<AudioSinkInterface>(new Sink(this)));
}
}
+MediaSourceInterface::SourceState RemoteAudioSource::state() const {
+ RTC_DCHECK(main_thread_->IsCurrent());
+ return state_;
+}
+
+bool RemoteAudioSource::remote() const {
+ RTC_DCHECK(main_thread_->IsCurrent());
+ return true;
+}
+
+void RemoteAudioSource::SetVolume(double volume) {
+ RTC_DCHECK(volume >= 0 && volume <= 10);
+ for (auto* observer : audio_observers_)
+ observer->OnSetVolume(volume);
+}
+
void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
- ASSERT(observer != NULL);
- ASSERT(std::find(audio_observers_.begin(), audio_observers_.end(),
- observer) == audio_observers_.end());
+ RTC_DCHECK(observer != NULL);
+ RTC_DCHECK(std::find(audio_observers_.begin(), audio_observers_.end(),
+ observer) == audio_observers_.end());
audio_observers_.push_back(observer);
}
void RemoteAudioSource::UnregisterAudioObserver(AudioObserver* observer) {
- ASSERT(observer != NULL);
+ RTC_DCHECK(observer != NULL);
audio_observers_.remove(observer);
}
+void RemoteAudioSource::AddSink(AudioTrackSinkInterface* sink) {
+ RTC_DCHECK(main_thread_->IsCurrent());
+ RTC_DCHECK(sink);
+
+ if (state_ != MediaSourceInterface::kLive) {
+ LOG(LS_ERROR) << "Can't register sink as the source isn't live.";
+ return;
+ }
+
+ rtc::CritScope lock(&sink_lock_);
+ RTC_DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end());
+ sinks_.push_back(sink);
+}
+
+void RemoteAudioSource::RemoveSink(AudioTrackSinkInterface* sink) {
+ RTC_DCHECK(main_thread_->IsCurrent());
+ RTC_DCHECK(sink);
+
+ rtc::CritScope lock(&sink_lock_);
+ sinks_.remove(sink);
+}
+
+void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) {
+ // Called on the externally-owned audio callback thread, via/from webrtc.
+ rtc::CritScope lock(&sink_lock_);
+ for (auto* sink : sinks_) {
+ sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
+ audio.samples_per_channel);
+ }
+}
+
+void RemoteAudioSource::OnAudioProviderGone() {
+ // Called when the data provider is deleted. It may be the worker thread
+ // in libjingle or may be a different worker thread.
+ main_thread_->Post(new MessageHandler(this));
+}
+
+void RemoteAudioSource::OnMessage(rtc::Message* msg) {
+ RTC_DCHECK(main_thread_->IsCurrent());
+ sinks_.clear();
+ state_ = MediaSourceInterface::kEnded;
+ FireOnChanged();
+}
+
} // namespace webrtc
diff --git a/talk/app/webrtc/remoteaudiosource.h b/talk/app/webrtc/remoteaudiosource.h
index e49aca5..d648ba4 100644
--- a/talk/app/webrtc/remoteaudiosource.h
+++ b/talk/app/webrtc/remoteaudiosource.h
@@ -29,36 +29,66 @@
#define TALK_APP_WEBRTC_REMOTEAUDIOSOURCE_H_
#include <list>
+#include <string>
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/notifier.h"
+#include "talk/media/base/audiorenderer.h"
+#include "webrtc/audio/audio_sink.h"
+#include "webrtc/base/criticalsection.h"
+
+namespace rtc {
+struct Message;
+class Thread;
+} // namespace rtc
namespace webrtc {
-using webrtc::AudioSourceInterface;
+class AudioProviderInterface;
// This class implements the audio source used by the remote audio track.
class RemoteAudioSource : public Notifier<AudioSourceInterface> {
public:
// Creates an instance of RemoteAudioSource.
- static rtc::scoped_refptr<RemoteAudioSource> Create();
-
- protected:
- RemoteAudioSource();
- virtual ~RemoteAudioSource();
-
- private:
- typedef std::list<AudioObserver*> AudioObserverList;
+ static rtc::scoped_refptr<RemoteAudioSource> Create(
+ uint32_t ssrc,
+ AudioProviderInterface* provider);
// MediaSourceInterface implementation.
MediaSourceInterface::SourceState state() const override;
+ bool remote() const override;
+
+ void AddSink(AudioTrackSinkInterface* sink) override;
+ void RemoveSink(AudioTrackSinkInterface* sink) override;
+
+ protected:
+ RemoteAudioSource();
+ ~RemoteAudioSource() override;
+
+ // Post construction initialize where we can do things like save a reference
+ // to ourselves (need to be fully constructed).
+ void Initialize(uint32_t ssrc, AudioProviderInterface* provider);
+
+ private:
+ typedef std::list<AudioObserver*> AudioObserverList;
// AudioSourceInterface implementation.
void SetVolume(double volume) override;
void RegisterAudioObserver(AudioObserver* observer) override;
void UnregisterAudioObserver(AudioObserver* observer) override;
+ class Sink;
+ void OnData(const AudioSinkInterface::Data& audio);
+ void OnAudioProviderGone();
+
+ class MessageHandler;
+ void OnMessage(rtc::Message* msg);
+
AudioObserverList audio_observers_;
+ rtc::CriticalSection sink_lock_;
+ std::list<AudioTrackSinkInterface*> sinks_;
+ rtc::Thread* const main_thread_;
+ SourceState state_;
};
} // namespace webrtc
diff --git a/talk/app/webrtc/remoteaudiotrack.cc b/talk/app/webrtc/remoteaudiotrack.cc
new file mode 100644
index 0000000..5f0b23e
--- /dev/null
+++ b/talk/app/webrtc/remoteaudiotrack.cc
@@ -0,0 +1,28 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+// TODO(tommi): Delete this file when removed from build files in Chromium.
diff --git a/talk/app/webrtc/remoteaudiotrack.h b/talk/app/webrtc/remoteaudiotrack.h
new file mode 100644
index 0000000..5f0b23e
--- /dev/null
+++ b/talk/app/webrtc/remoteaudiotrack.h
@@ -0,0 +1,28 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+// TODO(tommi): Delete this file when removed from build files in Chromium.
diff --git a/talk/app/webrtc/rtpreceiver.cc b/talk/app/webrtc/rtpreceiver.cc
index b88554f..9540f36 100644
--- a/talk/app/webrtc/rtpreceiver.cc
+++ b/talk/app/webrtc/rtpreceiver.cc
@@ -39,6 +39,7 @@
ssrc_(ssrc),
provider_(provider),
cached_track_enabled_(track->enabled()) {
+ RTC_DCHECK(track_->GetSource()->remote());
track_->RegisterObserver(this);
track_->GetSource()->RegisterAudioObserver(this);
Reconfigure();
@@ -85,6 +86,7 @@
uint32_t ssrc,
VideoProviderInterface* provider)
: id_(track->id()), track_(track), ssrc_(ssrc), provider_(provider) {
+ RTC_DCHECK(track_->GetSource()->remote());
provider_->SetVideoPlayout(ssrc_, true, track_->GetSource()->FrameInput());
}
diff --git a/talk/app/webrtc/rtpreceiver.h b/talk/app/webrtc/rtpreceiver.h
index a93ccbc..db021ba 100644
--- a/talk/app/webrtc/rtpreceiver.h
+++ b/talk/app/webrtc/rtpreceiver.h
@@ -68,10 +68,10 @@
private:
void Reconfigure();
- std::string id_;
- rtc::scoped_refptr<AudioTrackInterface> track_;
- uint32_t ssrc_;
- AudioProviderInterface* provider_;
+ const std::string id_;
+ const rtc::scoped_refptr<AudioTrackInterface> track_;
+ const uint32_t ssrc_;
+ AudioProviderInterface* provider_; // Set to null in Stop().
bool cached_track_enabled_;
};
diff --git a/talk/app/webrtc/rtpsender.cc b/talk/app/webrtc/rtpsender.cc
index 3a78f45..91e484b 100644
--- a/talk/app/webrtc/rtpsender.cc
+++ b/talk/app/webrtc/rtpsender.cc
@@ -29,6 +29,7 @@
#include "talk/app/webrtc/localaudiosource.h"
#include "talk/app/webrtc/videosourceinterface.h"
+#include "webrtc/base/helpers.h"
namespace webrtc {
@@ -43,7 +44,7 @@
void LocalAudioSinkAdapter::OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) {
rtc::CritScope lock(&lock_);
if (sink_) {
@@ -59,34 +60,49 @@
}
AudioRtpSender::AudioRtpSender(AudioTrackInterface* track,
- uint32_t ssrc,
- AudioProviderInterface* provider)
+ const std::string& stream_id,
+ AudioProviderInterface* provider,
+ StatsCollector* stats)
: id_(track->id()),
- track_(track),
- ssrc_(ssrc),
+ stream_id_(stream_id),
provider_(provider),
+ stats_(stats),
+ track_(track),
cached_track_enabled_(track->enabled()),
sink_adapter_(new LocalAudioSinkAdapter()) {
+ RTC_DCHECK(provider != nullptr);
track_->RegisterObserver(this);
track_->AddSink(sink_adapter_.get());
- Reconfigure();
}
+AudioRtpSender::AudioRtpSender(AudioProviderInterface* provider,
+ StatsCollector* stats)
+ : id_(rtc::CreateRandomUuid()),
+ stream_id_(rtc::CreateRandomUuid()),
+ provider_(provider),
+ stats_(stats),
+ sink_adapter_(new LocalAudioSinkAdapter()) {}
+
AudioRtpSender::~AudioRtpSender() {
- track_->RemoveSink(sink_adapter_.get());
- track_->UnregisterObserver(this);
Stop();
}
void AudioRtpSender::OnChanged() {
+ RTC_DCHECK(!stopped_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
- Reconfigure();
+ if (can_send_track()) {
+ SetAudioSend();
+ }
}
}
bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
- if (track->kind() != "audio") {
+ if (stopped_) {
+ LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
+ return false;
+ }
+ if (track && track->kind() != MediaStreamTrackInterface::kAudioKind) {
LOG(LS_ERROR) << "SetTrack called on audio RtpSender with " << track->kind()
<< " track.";
return false;
@@ -94,36 +110,84 @@
AudioTrackInterface* audio_track = static_cast<AudioTrackInterface*>(track);
// Detach from old track.
- track_->RemoveSink(sink_adapter_.get());
- track_->UnregisterObserver(this);
+ if (track_) {
+ track_->RemoveSink(sink_adapter_.get());
+ track_->UnregisterObserver(this);
+ }
+
+ if (can_send_track() && stats_) {
+ stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
+ }
// Attach to new track.
+ bool prev_can_send_track = can_send_track();
track_ = audio_track;
- cached_track_enabled_ = track_->enabled();
- track_->RegisterObserver(this);
- track_->AddSink(sink_adapter_.get());
- Reconfigure();
+ if (track_) {
+ cached_track_enabled_ = track_->enabled();
+ track_->RegisterObserver(this);
+ track_->AddSink(sink_adapter_.get());
+ }
+
+ // Update audio provider.
+ if (can_send_track()) {
+ SetAudioSend();
+ if (stats_) {
+ stats_->AddLocalAudioTrack(track_.get(), ssrc_);
+ }
+ } else if (prev_can_send_track) {
+ cricket::AudioOptions options;
+ provider_->SetAudioSend(ssrc_, false, options, nullptr);
+ }
return true;
}
+void AudioRtpSender::SetSsrc(uint32_t ssrc) {
+ if (stopped_ || ssrc == ssrc_) {
+ return;
+ }
+ // If we are already sending with a particular SSRC, stop sending.
+ if (can_send_track()) {
+ cricket::AudioOptions options;
+ provider_->SetAudioSend(ssrc_, false, options, nullptr);
+ if (stats_) {
+ stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
+ }
+ }
+ ssrc_ = ssrc;
+ if (can_send_track()) {
+ SetAudioSend();
+ if (stats_) {
+ stats_->AddLocalAudioTrack(track_.get(), ssrc_);
+ }
+ }
+}
+
void AudioRtpSender::Stop() {
// TODO(deadbeef): Need to do more here to fully stop sending packets.
- if (!provider_) {
+ if (stopped_) {
return;
}
- cricket::AudioOptions options;
- provider_->SetAudioSend(ssrc_, false, options, nullptr);
- provider_ = nullptr;
+ if (track_) {
+ track_->RemoveSink(sink_adapter_.get());
+ track_->UnregisterObserver(this);
+ }
+ if (can_send_track()) {
+ cricket::AudioOptions options;
+ provider_->SetAudioSend(ssrc_, false, options, nullptr);
+ if (stats_) {
+ stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
+ }
+ }
+ stopped_ = true;
}
-void AudioRtpSender::Reconfigure() {
- if (!provider_) {
- return;
- }
+void AudioRtpSender::SetAudioSend() {
+ RTC_DCHECK(!stopped_ && can_send_track());
cricket::AudioOptions options;
- if (track_->enabled() && track_->GetSource()) {
+ if (track_->enabled() && track_->GetSource() &&
+ !track_->GetSource()->remote()) {
// TODO(xians): Remove this static_cast since we should be able to connect
- // a remote audio track to peer connection.
+ // a remote audio track to a peer connection.
options = static_cast<LocalAudioSource*>(track_->GetSource())->options();
}
@@ -136,35 +200,42 @@
}
VideoRtpSender::VideoRtpSender(VideoTrackInterface* track,
- uint32_t ssrc,
+ const std::string& stream_id,
VideoProviderInterface* provider)
: id_(track->id()),
- track_(track),
- ssrc_(ssrc),
+ stream_id_(stream_id),
provider_(provider),
+ track_(track),
cached_track_enabled_(track->enabled()) {
+ RTC_DCHECK(provider != nullptr);
track_->RegisterObserver(this);
- VideoSourceInterface* source = track_->GetSource();
- if (source) {
- provider_->SetCaptureDevice(ssrc_, source->GetVideoCapturer());
- }
- Reconfigure();
}
+VideoRtpSender::VideoRtpSender(VideoProviderInterface* provider)
+ : id_(rtc::CreateRandomUuid()),
+ stream_id_(rtc::CreateRandomUuid()),
+ provider_(provider) {}
+
VideoRtpSender::~VideoRtpSender() {
- track_->UnregisterObserver(this);
Stop();
}
void VideoRtpSender::OnChanged() {
+ RTC_DCHECK(!stopped_);
if (cached_track_enabled_ != track_->enabled()) {
cached_track_enabled_ = track_->enabled();
- Reconfigure();
+ if (can_send_track()) {
+ SetVideoSend();
+ }
}
}
bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
- if (track->kind() != "video") {
+ if (stopped_) {
+ LOG(LS_ERROR) << "SetTrack can't be called on a stopped RtpSender.";
+ return false;
+ }
+ if (track && track->kind() != MediaStreamTrackInterface::kVideoKind) {
LOG(LS_ERROR) << "SetTrack called on video RtpSender with " << track->kind()
<< " track.";
return false;
@@ -172,30 +243,72 @@
VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track);
// Detach from old track.
- track_->UnregisterObserver(this);
+ if (track_) {
+ track_->UnregisterObserver(this);
+ }
// Attach to new track.
+ bool prev_can_send_track = can_send_track();
track_ = video_track;
- cached_track_enabled_ = track_->enabled();
- track_->RegisterObserver(this);
- Reconfigure();
+ if (track_) {
+ cached_track_enabled_ = track_->enabled();
+ track_->RegisterObserver(this);
+ }
+
+ // Update video provider.
+ if (can_send_track()) {
+ VideoSourceInterface* source = track_->GetSource();
+ // TODO(deadbeef): If SetTrack is called with a disabled track, and the
+ // previous track was enabled, this could cause a frame from the new track
+ // to slip out. Really, what we need is for SetCaptureDevice and
+ // SetVideoSend
+ // to be combined into one atomic operation, all the way down to
+ // WebRtcVideoSendStream.
+ provider_->SetCaptureDevice(ssrc_,
+ source ? source->GetVideoCapturer() : nullptr);
+ SetVideoSend();
+ } else if (prev_can_send_track) {
+ provider_->SetCaptureDevice(ssrc_, nullptr);
+ provider_->SetVideoSend(ssrc_, false, nullptr);
+ }
return true;
}
+void VideoRtpSender::SetSsrc(uint32_t ssrc) {
+ if (stopped_ || ssrc == ssrc_) {
+ return;
+ }
+ // If we are already sending with a particular SSRC, stop sending.
+ if (can_send_track()) {
+ provider_->SetCaptureDevice(ssrc_, nullptr);
+ provider_->SetVideoSend(ssrc_, false, nullptr);
+ }
+ ssrc_ = ssrc;
+ if (can_send_track()) {
+ VideoSourceInterface* source = track_->GetSource();
+ provider_->SetCaptureDevice(ssrc_,
+ source ? source->GetVideoCapturer() : nullptr);
+ SetVideoSend();
+ }
+}
+
void VideoRtpSender::Stop() {
// TODO(deadbeef): Need to do more here to fully stop sending packets.
- if (!provider_) {
+ if (stopped_) {
return;
}
- provider_->SetCaptureDevice(ssrc_, nullptr);
- provider_->SetVideoSend(ssrc_, false, nullptr);
- provider_ = nullptr;
+ if (track_) {
+ track_->UnregisterObserver(this);
+ }
+ if (can_send_track()) {
+ provider_->SetCaptureDevice(ssrc_, nullptr);
+ provider_->SetVideoSend(ssrc_, false, nullptr);
+ }
+ stopped_ = true;
}
-void VideoRtpSender::Reconfigure() {
- if (!provider_) {
- return;
- }
+void VideoRtpSender::SetVideoSend() {
+ RTC_DCHECK(!stopped_ && can_send_track());
const cricket::VideoOptions* options = nullptr;
VideoSourceInterface* source = track_->GetSource();
if (track_->enabled() && source) {
diff --git a/talk/app/webrtc/rtpsender.h b/talk/app/webrtc/rtpsender.h
index 3741909..dd846b5 100644
--- a/talk/app/webrtc/rtpsender.h
+++ b/talk/app/webrtc/rtpsender.h
@@ -36,6 +36,7 @@
#include "talk/app/webrtc/mediastreamprovider.h"
#include "talk/app/webrtc/rtpsenderinterface.h"
+#include "talk/app/webrtc/statscollector.h"
#include "talk/media/base/audiorenderer.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/criticalsection.h"
@@ -56,7 +57,7 @@
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) override;
// cricket::AudioRenderer implementation.
@@ -70,9 +71,15 @@
class AudioRtpSender : public ObserverInterface,
public rtc::RefCountedObject<RtpSenderInterface> {
public:
+ // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
+ // at the appropriate times.
AudioRtpSender(AudioTrackInterface* track,
- uint32_t ssrc,
- AudioProviderInterface* provider);
+ const std::string& stream_id,
+ AudioProviderInterface* provider,
+ StatsCollector* stats);
+
+ // Randomly generates id and stream_id.
+ AudioRtpSender(AudioProviderInterface* provider, StatsCollector* stats);
virtual ~AudioRtpSender();
@@ -85,18 +92,37 @@
return track_.get();
}
+ void SetSsrc(uint32_t ssrc) override;
+
+ uint32_t ssrc() const override { return ssrc_; }
+
+ cricket::MediaType media_type() const override {
+ return cricket::MEDIA_TYPE_AUDIO;
+ }
+
std::string id() const override { return id_; }
+ void set_stream_id(const std::string& stream_id) override {
+ stream_id_ = stream_id;
+ }
+ std::string stream_id() const override { return stream_id_; }
+
void Stop() override;
private:
- void Reconfigure();
+ bool can_send_track() const { return track_ && ssrc_; }
+ // Helper function to construct options for
+ // AudioProviderInterface::SetAudioSend.
+ void SetAudioSend();
std::string id_;
- rtc::scoped_refptr<AudioTrackInterface> track_;
- uint32_t ssrc_;
+ std::string stream_id_;
AudioProviderInterface* provider_;
- bool cached_track_enabled_;
+ StatsCollector* stats_;
+ rtc::scoped_refptr<AudioTrackInterface> track_;
+ uint32_t ssrc_ = 0;
+ bool cached_track_enabled_ = false;
+ bool stopped_ = false;
// Used to pass the data callback from the |track_| to the other end of
// cricket::AudioRenderer.
@@ -107,9 +133,12 @@
public rtc::RefCountedObject<RtpSenderInterface> {
public:
VideoRtpSender(VideoTrackInterface* track,
- uint32_t ssrc,
+ const std::string& stream_id,
VideoProviderInterface* provider);
+ // Randomly generates id and stream_id.
+ explicit VideoRtpSender(VideoProviderInterface* provider);
+
virtual ~VideoRtpSender();
// ObserverInterface implementation
@@ -121,18 +150,36 @@
return track_.get();
}
+ void SetSsrc(uint32_t ssrc) override;
+
+ uint32_t ssrc() const override { return ssrc_; }
+
+ cricket::MediaType media_type() const override {
+ return cricket::MEDIA_TYPE_VIDEO;
+ }
+
std::string id() const override { return id_; }
+ void set_stream_id(const std::string& stream_id) override {
+ stream_id_ = stream_id;
+ }
+ std::string stream_id() const override { return stream_id_; }
+
void Stop() override;
private:
- void Reconfigure();
+ bool can_send_track() const { return track_ && ssrc_; }
+ // Helper function to construct options for
+ // VideoProviderInterface::SetVideoSend.
+ void SetVideoSend();
std::string id_;
- rtc::scoped_refptr<VideoTrackInterface> track_;
- uint32_t ssrc_;
+ std::string stream_id_;
VideoProviderInterface* provider_;
- bool cached_track_enabled_;
+ rtc::scoped_refptr<VideoTrackInterface> track_;
+ uint32_t ssrc_ = 0;
+ bool cached_track_enabled_ = false;
+ bool stopped_ = false;
};
} // namespace webrtc
diff --git a/talk/app/webrtc/rtpsenderinterface.h b/talk/app/webrtc/rtpsenderinterface.h
index fca98f2..f54e8ca 100644
--- a/talk/app/webrtc/rtpsenderinterface.h
+++ b/talk/app/webrtc/rtpsenderinterface.h
@@ -35,6 +35,7 @@
#include "talk/app/webrtc/proxy.h"
#include "talk/app/webrtc/mediastreaminterface.h"
+#include "talk/session/media/mediasession.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
@@ -47,10 +48,24 @@
virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+ // Used to set the SSRC of the sender, once a local description has been set.
+ // If |ssrc| is 0, this indiates that the sender should disconnect from the
+ // underlying transport (this occurs if the sender isn't seen in a local
+ // description).
+ virtual void SetSsrc(uint32_t ssrc) = 0;
+ virtual uint32_t ssrc() const = 0;
+
+ // Audio or video sender?
+ virtual cricket::MediaType media_type() const = 0;
+
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
virtual std::string id() const = 0;
+ // TODO(deadbeef): Support one sender having multiple stream ids.
+ virtual void set_stream_id(const std::string& stream_id) = 0;
+ virtual std::string stream_id() const = 0;
+
virtual void Stop() = 0;
protected:
@@ -61,7 +76,12 @@
BEGIN_PROXY_MAP(RtpSender)
PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
+PROXY_METHOD1(void, SetSsrc, uint32_t)
+PROXY_CONSTMETHOD0(uint32_t, ssrc)
+PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
PROXY_CONSTMETHOD0(std::string, id)
+PROXY_METHOD1(void, set_stream_id, const std::string&)
+PROXY_CONSTMETHOD0(std::string, stream_id)
PROXY_METHOD0(void, Stop)
END_PROXY()
diff --git a/talk/app/webrtc/rtpsenderreceiver_unittest.cc b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
index c9d7e00..a590e1d 100644
--- a/talk/app/webrtc/rtpsenderreceiver_unittest.cc
+++ b/talk/app/webrtc/rtpsenderreceiver_unittest.cc
@@ -26,6 +26,7 @@
*/
#include <string>
+#include <utility>
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/mediastream.h"
@@ -48,14 +49,17 @@
static const char kVideoTrackId[] = "video_1";
static const char kAudioTrackId[] = "audio_1";
static const uint32_t kVideoSsrc = 98;
+static const uint32_t kVideoSsrc2 = 100;
static const uint32_t kAudioSsrc = 99;
+static const uint32_t kAudioSsrc2 = 101;
namespace webrtc {
// Helper class to test RtpSender/RtpReceiver.
class MockAudioProvider : public AudioProviderInterface {
public:
- virtual ~MockAudioProvider() {}
+ ~MockAudioProvider() override {}
+
MOCK_METHOD2(SetAudioPlayout,
void(uint32_t ssrc,
bool enable));
@@ -65,6 +69,14 @@
const cricket::AudioOptions& options,
cricket::AudioRenderer* renderer));
MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
+
+ void SetRawAudioSink(uint32_t,
+ rtc::scoped_ptr<AudioSinkInterface> sink) override {
+ sink_ = std::move(sink);
+ }
+
+ private:
+ rtc::scoped_ptr<AudioSinkInterface> sink_;
};
// Helper class to test RtpSender/RtpReceiver.
@@ -85,8 +97,8 @@
class FakeVideoSource : public Notifier<VideoSourceInterface> {
public:
- static rtc::scoped_refptr<FakeVideoSource> Create() {
- return new rtc::RefCountedObject<FakeVideoSource>();
+ static rtc::scoped_refptr<FakeVideoSource> Create(bool remote) {
+ return new rtc::RefCountedObject<FakeVideoSource>(remote);
}
virtual cricket::VideoCapturer* GetVideoCapturer() { return &fake_capturer_; }
virtual void Stop() {}
@@ -94,16 +106,18 @@
virtual void AddSink(cricket::VideoRenderer* output) {}
virtual void RemoveSink(cricket::VideoRenderer* output) {}
virtual SourceState state() const { return state_; }
+ virtual bool remote() const { return remote_; }
virtual const cricket::VideoOptions* options() const { return &options_; }
virtual cricket::VideoRenderer* FrameInput() { return NULL; }
protected:
- FakeVideoSource() : state_(kLive) {}
+ explicit FakeVideoSource(bool remote) : state_(kLive), remote_(remote) {}
~FakeVideoSource() {}
private:
cricket::FakeVideoCapturer fake_capturer_;
SourceState state_;
+ bool remote_;
cricket::VideoOptions options_;
};
@@ -111,7 +125,11 @@
public:
virtual void SetUp() {
stream_ = MediaStream::Create(kStreamLabel1);
- rtc::scoped_refptr<VideoSourceInterface> source(FakeVideoSource::Create());
+ }
+
+ void AddVideoTrack(bool remote) {
+ rtc::scoped_refptr<VideoSourceInterface> source(
+ FakeVideoSource::Create(remote));
video_track_ = VideoTrack::Create(kVideoTrackId, source);
EXPECT_TRUE(stream_->AddTrack(video_track_));
}
@@ -120,17 +138,21 @@
audio_track_ = AudioTrack::Create(kAudioTrackId, NULL);
EXPECT_TRUE(stream_->AddTrack(audio_track_));
EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
- audio_rtp_sender_ = new AudioRtpSender(stream_->GetAudioTracks()[0],
- kAudioSsrc, &audio_provider_);
+ audio_rtp_sender_ =
+ new AudioRtpSender(stream_->GetAudioTracks()[0], stream_->label(),
+ &audio_provider_, nullptr);
+ audio_rtp_sender_->SetSsrc(kAudioSsrc);
}
void CreateVideoRtpSender() {
+ AddVideoTrack(false);
EXPECT_CALL(video_provider_,
SetCaptureDevice(
kVideoSsrc, video_track_->GetSource()->GetVideoCapturer()));
EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
video_rtp_sender_ = new VideoRtpSender(stream_->GetVideoTracks()[0],
- kVideoSsrc, &video_provider_);
+ stream_->label(), &video_provider_);
+ video_rtp_sender_->SetSsrc(kVideoSsrc);
}
void DestroyAudioRtpSender() {
@@ -146,8 +168,8 @@
}
void CreateAudioRtpReceiver() {
- audio_track_ =
- AudioTrack::Create(kAudioTrackId, RemoteAudioSource::Create().get());
+ audio_track_ = AudioTrack::Create(
+ kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL));
EXPECT_TRUE(stream_->AddTrack(audio_track_));
EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));
audio_rtp_receiver_ = new AudioRtpReceiver(stream_->GetAudioTracks()[0],
@@ -155,6 +177,7 @@
}
void CreateVideoRtpReceiver() {
+ AddVideoTrack(true);
EXPECT_CALL(video_provider_,
SetVideoPlayout(kVideoSsrc, true,
video_track_->GetSource()->FrameInput()));
@@ -280,4 +303,212 @@
DestroyAudioRtpReceiver();
}
+// Test that provider methods aren't called without both a track and an SSRC.
+TEST_F(RtpSenderReceiverTest, AudioSenderWithoutTrackAndSsrc) {
+ rtc::scoped_refptr<AudioRtpSender> sender =
+ new AudioRtpSender(&audio_provider_, nullptr);
+ rtc::scoped_refptr<AudioTrackInterface> track =
+ AudioTrack::Create(kAudioTrackId, nullptr);
+ EXPECT_TRUE(sender->SetTrack(track));
+ EXPECT_TRUE(sender->SetTrack(nullptr));
+ sender->SetSsrc(kAudioSsrc);
+ sender->SetSsrc(0);
+ // Just let it get destroyed and make sure it doesn't call any methods on the
+ // provider interface.
+}
+
+// Test that provider methods aren't called without both a track and an SSRC.
+TEST_F(RtpSenderReceiverTest, VideoSenderWithoutTrackAndSsrc) {
+ rtc::scoped_refptr<VideoRtpSender> sender =
+ new VideoRtpSender(&video_provider_);
+ EXPECT_TRUE(sender->SetTrack(video_track_));
+ EXPECT_TRUE(sender->SetTrack(nullptr));
+ sender->SetSsrc(kVideoSsrc);
+ sender->SetSsrc(0);
+ // Just let it get destroyed and make sure it doesn't call any methods on the
+ // provider interface.
+}
+
+// Test that an audio sender calls the expected methods on the provider once
+// it has a track and SSRC, when the SSRC is set first.
+TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupSsrcThenTrack) {
+ rtc::scoped_refptr<AudioRtpSender> sender =
+ new AudioRtpSender(&audio_provider_, nullptr);
+ rtc::scoped_refptr<AudioTrackInterface> track =
+ AudioTrack::Create(kAudioTrackId, nullptr);
+ sender->SetSsrc(kAudioSsrc);
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
+ sender->SetTrack(track);
+
+ // Calls expected from destructor.
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
+}
+
+// Test that an audio sender calls the expected methods on the provider once
+// it has a track and SSRC, when the SSRC is set last.
+TEST_F(RtpSenderReceiverTest, AudioSenderEarlyWarmupTrackThenSsrc) {
+ rtc::scoped_refptr<AudioRtpSender> sender =
+ new AudioRtpSender(&audio_provider_, nullptr);
+ rtc::scoped_refptr<AudioTrackInterface> track =
+ AudioTrack::Create(kAudioTrackId, nullptr);
+ sender->SetTrack(track);
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
+ sender->SetSsrc(kAudioSsrc);
+
+ // Calls expected from destructor.
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
+}
+
+// Test that a video sender calls the expected methods on the provider once
+// it has a track and SSRC, when the SSRC is set first.
+TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupSsrcThenTrack) {
+ AddVideoTrack(false);
+ rtc::scoped_refptr<VideoRtpSender> sender =
+ new VideoRtpSender(&video_provider_);
+ sender->SetSsrc(kVideoSsrc);
+ EXPECT_CALL(video_provider_,
+ SetCaptureDevice(kVideoSsrc,
+ video_track_->GetSource()->GetVideoCapturer()));
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
+ sender->SetTrack(video_track_);
+
+ // Calls expected from destructor.
+ EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
+}
+
+// Test that a video sender calls the expected methods on the provider once
+// it has a track and SSRC, when the SSRC is set last.
+TEST_F(RtpSenderReceiverTest, VideoSenderEarlyWarmupTrackThenSsrc) {
+ AddVideoTrack(false);
+ rtc::scoped_refptr<VideoRtpSender> sender =
+ new VideoRtpSender(&video_provider_);
+ sender->SetTrack(video_track_);
+ EXPECT_CALL(video_provider_,
+ SetCaptureDevice(kVideoSsrc,
+ video_track_->GetSource()->GetVideoCapturer()));
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
+ sender->SetSsrc(kVideoSsrc);
+
+ // Calls expected from destructor.
+ EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
+}
+
+// Test that the sender is disconnected from the provider when its SSRC is
+// set to 0.
+TEST_F(RtpSenderReceiverTest, AudioSenderSsrcSetToZero) {
+ rtc::scoped_refptr<AudioTrackInterface> track =
+ AudioTrack::Create(kAudioTrackId, nullptr);
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
+ rtc::scoped_refptr<AudioRtpSender> sender =
+ new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
+ sender->SetSsrc(kAudioSsrc);
+
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
+ sender->SetSsrc(0);
+
+ // Make sure it's SetSsrc that called methods on the provider, and not the
+ // destructor.
+ EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
+}
+
+// Test that the sender is disconnected from the provider when its SSRC is
+// set to 0.
+TEST_F(RtpSenderReceiverTest, VideoSenderSsrcSetToZero) {
+ AddVideoTrack(false);
+ EXPECT_CALL(video_provider_,
+ SetCaptureDevice(kVideoSsrc,
+ video_track_->GetSource()->GetVideoCapturer()));
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
+ rtc::scoped_refptr<VideoRtpSender> sender =
+ new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
+ sender->SetSsrc(kVideoSsrc);
+
+ EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
+ sender->SetSsrc(0);
+
+ // Make sure it's SetSsrc that called methods on the provider, and not the
+ // destructor.
+ EXPECT_CALL(video_provider_, SetCaptureDevice(_, _)).Times(0);
+ EXPECT_CALL(video_provider_, SetVideoSend(_, _, _)).Times(0);
+}
+
+TEST_F(RtpSenderReceiverTest, AudioSenderTrackSetToNull) {
+ rtc::scoped_refptr<AudioTrackInterface> track =
+ AudioTrack::Create(kAudioTrackId, nullptr);
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
+ rtc::scoped_refptr<AudioRtpSender> sender =
+ new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
+ sender->SetSsrc(kAudioSsrc);
+
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
+ EXPECT_TRUE(sender->SetTrack(nullptr));
+
+ // Make sure it's SetTrack that called methods on the provider, and not the
+ // destructor.
+ EXPECT_CALL(audio_provider_, SetAudioSend(_, _, _, _)).Times(0);
+}
+
+TEST_F(RtpSenderReceiverTest, VideoSenderTrackSetToNull) {
+ AddVideoTrack(false);
+ EXPECT_CALL(video_provider_,
+ SetCaptureDevice(kVideoSsrc,
+ video_track_->GetSource()->GetVideoCapturer()));
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
+ rtc::scoped_refptr<VideoRtpSender> sender =
+ new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
+ sender->SetSsrc(kVideoSsrc);
+
+ EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
+ EXPECT_TRUE(sender->SetTrack(nullptr));
+
+ // Make sure it's SetTrack that called methods on the provider, and not the
+ // destructor.
+ EXPECT_CALL(video_provider_, SetCaptureDevice(_, _)).Times(0);
+ EXPECT_CALL(video_provider_, SetVideoSend(_, _, _)).Times(0);
+}
+
+TEST_F(RtpSenderReceiverTest, AudioSenderSsrcChanged) {
+ AddVideoTrack(false);
+ rtc::scoped_refptr<AudioTrackInterface> track =
+ AudioTrack::Create(kAudioTrackId, nullptr);
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, true, _, _));
+ rtc::scoped_refptr<AudioRtpSender> sender =
+ new AudioRtpSender(track, kStreamLabel1, &audio_provider_, nullptr);
+ sender->SetSsrc(kAudioSsrc);
+
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc, false, _, _)).Times(1);
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, true, _, _)).Times(1);
+ sender->SetSsrc(kAudioSsrc2);
+
+ // Calls expected from destructor.
+ EXPECT_CALL(audio_provider_, SetAudioSend(kAudioSsrc2, false, _, _)).Times(1);
+}
+
+TEST_F(RtpSenderReceiverTest, VideoSenderSsrcChanged) {
+ AddVideoTrack(false);
+ EXPECT_CALL(video_provider_,
+ SetCaptureDevice(kVideoSsrc,
+ video_track_->GetSource()->GetVideoCapturer()));
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, true, _));
+ rtc::scoped_refptr<VideoRtpSender> sender =
+ new VideoRtpSender(video_track_, kStreamLabel1, &video_provider_);
+ sender->SetSsrc(kVideoSsrc);
+
+ EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, nullptr)).Times(1);
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
+ EXPECT_CALL(video_provider_,
+ SetCaptureDevice(kVideoSsrc2,
+ video_track_->GetSource()->GetVideoCapturer()));
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _));
+ sender->SetSsrc(kVideoSsrc2);
+
+ // Calls expected from destructor.
+ EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1);
+ EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1);
+}
+
} // namespace webrtc
diff --git a/talk/app/webrtc/statscollector.cc b/talk/app/webrtc/statscollector.cc
index 347a846..b514b42 100644
--- a/talk/app/webrtc/statscollector.cc
+++ b/talk/app/webrtc/statscollector.cc
@@ -115,17 +115,17 @@
report->AddString(StatsReport::kStatsValueNameCodecName, info.codec_name);
}
-void SetAudioProcessingStats(StatsReport* report, int signal_level,
- bool typing_noise_detected, int echo_return_loss,
- int echo_return_loss_enhancement, int echo_delay_median_ms,
- float aec_quality_min, int echo_delay_std_ms) {
+void SetAudioProcessingStats(StatsReport* report,
+ bool typing_noise_detected,
+ int echo_return_loss,
+ int echo_return_loss_enhancement,
+ int echo_delay_median_ms,
+ float aec_quality_min,
+ int echo_delay_std_ms) {
report->AddBoolean(StatsReport::kStatsValueNameTypingNoiseState,
typing_noise_detected);
report->AddFloat(StatsReport::kStatsValueNameEchoCancellationQualityMin,
aec_quality_min);
- // Don't overwrite the previous signal level if it's not available now.
- if (signal_level >= 0)
- report->AddInt(StatsReport::kStatsValueNameAudioInputLevel, signal_level);
const IntForAdd ints[] = {
{ StatsReport::kStatsValueNameEchoReturnLoss, echo_return_loss },
{ StatsReport::kStatsValueNameEchoReturnLossEnhancement,
@@ -182,11 +182,14 @@
void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {
ExtractCommonSendProperties(info, report);
- SetAudioProcessingStats(report, info.audio_level, info.typing_noise_detected,
- info.echo_return_loss, info.echo_return_loss_enhancement,
- info.echo_delay_median_ms, info.aec_quality_min, info.echo_delay_std_ms);
+ SetAudioProcessingStats(
+ report, info.typing_noise_detected, info.echo_return_loss,
+ info.echo_return_loss_enhancement, info.echo_delay_median_ms,
+ info.aec_quality_min, info.echo_delay_std_ms);
+ RTC_DCHECK_GE(info.audio_level, 0);
const IntForAdd ints[] = {
+ { StatsReport::kStatsValueNameAudioInputLevel, info.audio_level},
{ StatsReport::kStatsValueNameJitterReceived, info.jitter_ms },
{ StatsReport::kStatsValueNamePacketsLost, info.packets_lost },
{ StatsReport::kStatsValueNamePacketsSent, info.packets_sent },
@@ -198,6 +201,8 @@
void ExtractStats(const cricket::VideoReceiverInfo& info, StatsReport* report) {
ExtractCommonReceiveProperties(info, report);
+ report->AddString(StatsReport::kStatsValueNameCodecImplementationName,
+ info.decoder_implementation_name);
report->AddInt64(StatsReport::kStatsValueNameBytesReceived,
info.bytes_rcvd);
report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
@@ -230,6 +235,8 @@
void ExtractStats(const cricket::VideoSenderInfo& info, StatsReport* report) {
ExtractCommonSendProperties(info, report);
+ report->AddString(StatsReport::kStatsValueNameCodecImplementationName,
+ info.encoder_implementation_name);
report->AddBoolean(StatsReport::kStatsValueNameBandwidthLimitedResolution,
(info.adapt_reason & 0x2) > 0);
report->AddBoolean(StatsReport::kStatsValueNameCpuLimitedResolution,
@@ -730,17 +737,20 @@
channel_report->AddId(StatsReport::kStatsValueNameRemoteCertificateId,
remote_cert_report_id);
}
- const std::string& srtp_cipher = channel_iter.srtp_cipher;
- if (!srtp_cipher.empty()) {
- channel_report->AddString(StatsReport::kStatsValueNameSrtpCipher,
- srtp_cipher);
+ int srtp_crypto_suite = channel_iter.srtp_crypto_suite;
+ if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE &&
+ rtc::SrtpCryptoSuiteToName(srtp_crypto_suite).length()) {
+ channel_report->AddString(
+ StatsReport::kStatsValueNameSrtpCipher,
+ rtc::SrtpCryptoSuiteToName(srtp_crypto_suite));
}
- int ssl_cipher = channel_iter.ssl_cipher;
- if (ssl_cipher &&
- rtc::SSLStreamAdapter::GetSslCipherSuiteName(ssl_cipher).length()) {
+ int ssl_cipher_suite = channel_iter.ssl_cipher_suite;
+ if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL &&
+ rtc::SSLStreamAdapter::SslCipherSuiteToName(ssl_cipher_suite)
+ .length()) {
channel_report->AddString(
StatsReport::kStatsValueNameDtlsCipher,
- rtc::SSLStreamAdapter::GetSslCipherSuiteName(ssl_cipher));
+ rtc::SSLStreamAdapter::SslCipherSuiteToName(ssl_cipher_suite));
}
int connection_id = 0;
@@ -888,21 +898,24 @@
RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent());
RTC_DCHECK(track != NULL);
- int signal_level = 0;
- if (!track->GetSignalLevel(&signal_level))
- signal_level = -1;
+ // Don't overwrite report values if they're not available.
+ int signal_level;
+ if (track->GetSignalLevel(&signal_level)) {
+ RTC_DCHECK_GE(signal_level, 0);
+ report->AddInt(StatsReport::kStatsValueNameAudioInputLevel, signal_level);
+ }
- rtc::scoped_refptr<AudioProcessorInterface> audio_processor(
- track->GetAudioProcessor());
+ auto audio_processor(track->GetAudioProcessor());
- AudioProcessorInterface::AudioProcessorStats stats;
- if (audio_processor.get())
+ if (audio_processor.get()) {
+ AudioProcessorInterface::AudioProcessorStats stats;
audio_processor->GetStats(&stats);
- SetAudioProcessingStats(report, signal_level, stats.typing_noise_detected,
- stats.echo_return_loss, stats.echo_return_loss_enhancement,
- stats.echo_delay_median_ms, stats.aec_quality_min,
- stats.echo_delay_std_ms);
+ SetAudioProcessingStats(
+ report, stats.typing_noise_detected, stats.echo_return_loss,
+ stats.echo_return_loss_enhancement, stats.echo_delay_median_ms,
+ stats.aec_quality_min, stats.echo_delay_std_ms);
+ }
}
bool StatsCollector::GetTrackIdBySsrc(uint32_t ssrc,
diff --git a/talk/app/webrtc/statscollector.h b/talk/app/webrtc/statscollector.h
index 18a345d..56db79d 100644
--- a/talk/app/webrtc/statscollector.h
+++ b/talk/app/webrtc/statscollector.h
@@ -36,7 +36,6 @@
#include <vector>
#include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/statstypes.h"
#include "talk/app/webrtc/webrtcsession.h"
diff --git a/talk/app/webrtc/statscollector_unittest.cc b/talk/app/webrtc/statscollector_unittest.cc
index 9121c69..e7ee911 100644
--- a/talk/app/webrtc/statscollector_unittest.cc
+++ b/talk/app/webrtc/statscollector_unittest.cc
@@ -35,7 +35,6 @@
#include "talk/app/webrtc/peerconnectionfactory.h"
#include "talk/app/webrtc/mediastream.h"
#include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/mediastreamtrack.h"
#include "talk/app/webrtc/test/fakedatachannelprovider.h"
#include "talk/app/webrtc/videotrack.h"
@@ -683,8 +682,8 @@
// Fake stats to process.
cricket::TransportChannelStats channel_stats;
channel_stats.component = 1;
- channel_stats.srtp_cipher = "the-srtp-cipher";
- channel_stats.ssl_cipher = TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA;
+ channel_stats.srtp_crypto_suite = rtc::SRTP_AES128_CM_SHA1_80;
+ channel_stats.ssl_cipher_suite = TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA;
cricket::TransportStats transport_stats;
transport_stats.transport_name = "audio";
@@ -697,8 +696,7 @@
// Fake certificate to report
rtc::scoped_refptr<rtc::RTCCertificate> local_certificate(
rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::FakeSSLIdentity>(
- new rtc::FakeSSLIdentity(local_cert))
- .Pass()));
+ new rtc::FakeSSLIdentity(local_cert))));
// Configure MockWebRtcSession
EXPECT_CALL(session_,
@@ -747,18 +745,17 @@
}
// Check negotiated ciphers.
- std::string dtls_cipher = ExtractStatsValue(
- StatsReport::kStatsReportTypeComponent,
- reports,
- StatsReport::kStatsValueNameDtlsCipher);
- EXPECT_EQ(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ std::string dtls_cipher_suite =
+ ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports,
+ StatsReport::kStatsValueNameDtlsCipher);
+ EXPECT_EQ(rtc::SSLStreamAdapter::SslCipherSuiteToName(
TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA),
- dtls_cipher);
- std::string srtp_cipher = ExtractStatsValue(
- StatsReport::kStatsReportTypeComponent,
- reports,
- StatsReport::kStatsValueNameSrtpCipher);
- EXPECT_EQ("the-srtp-cipher", srtp_cipher);
+ dtls_cipher_suite);
+ std::string srtp_crypto_suite =
+ ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports,
+ StatsReport::kStatsValueNameSrtpCipher);
+ EXPECT_EQ(rtc::SrtpCryptoSuiteToName(rtc::SRTP_AES128_CM_SHA1_80),
+ srtp_crypto_suite);
}
cricket::FakeMediaEngine* media_engine_;
@@ -1407,16 +1404,14 @@
ASSERT_EQ(kNotFound, remote_certificate_id);
// Check that the negotiated ciphers are absent.
- std::string dtls_cipher = ExtractStatsValue(
- StatsReport::kStatsReportTypeComponent,
- reports,
- StatsReport::kStatsValueNameDtlsCipher);
- ASSERT_EQ(kNotFound, dtls_cipher);
- std::string srtp_cipher = ExtractStatsValue(
- StatsReport::kStatsReportTypeComponent,
- reports,
- StatsReport::kStatsValueNameSrtpCipher);
- ASSERT_EQ(kNotFound, srtp_cipher);
+ std::string dtls_cipher_suite =
+ ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports,
+ StatsReport::kStatsValueNameDtlsCipher);
+ ASSERT_EQ(kNotFound, dtls_cipher_suite);
+ std::string srtp_crypto_suite =
+ ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports,
+ StatsReport::kStatsValueNameSrtpCipher);
+ ASSERT_EQ(kNotFound, srtp_crypto_suite);
}
// This test verifies that the stats are generated correctly when the transport
diff --git a/talk/app/webrtc/statstypes.cc b/talk/app/webrtc/statstypes.cc
index e45833c..19cb1f5 100644
--- a/talk/app/webrtc/statstypes.cc
+++ b/talk/app/webrtc/statstypes.cc
@@ -408,6 +408,8 @@
return "state";
case kStatsValueNameDataChannelId:
return "datachannelid";
+ case kStatsValueNameCodecImplementationName:
+ return "codecImplementationName";
// 'goog' prefixed constants.
case kStatsValueNameAccelerateRate:
@@ -592,9 +594,6 @@
return "googViewLimitedResolution";
case kStatsValueNameWritable:
return "googWritable";
- default:
- RTC_DCHECK(false);
- break;
}
return nullptr;
diff --git a/talk/app/webrtc/statstypes.h b/talk/app/webrtc/statstypes.h
index 7fa9f32..60439b9 100644
--- a/talk/app/webrtc/statstypes.h
+++ b/talk/app/webrtc/statstypes.h
@@ -120,6 +120,7 @@
kStatsValueNameAudioOutputLevel,
kStatsValueNameBytesReceived,
kStatsValueNameBytesSent,
+ kStatsValueNameCodecImplementationName,
kStatsValueNameDataChannelId,
kStatsValueNamePacketsLost,
kStatsValueNamePacketsReceived,
diff --git a/talk/app/webrtc/test/DEPS b/talk/app/webrtc/test/DEPS
new file mode 100644
index 0000000..a814b15
--- /dev/null
+++ b/talk/app/webrtc/test/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+ # Allow include of Chrome base/android to allow inclusion of headers needed
+ # for accessing the JVM and Application context in gtest.
+ "+base/android",
+]
diff --git a/talk/app/webrtc/test/androidtestinitializer.cc b/talk/app/webrtc/test/androidtestinitializer.cc
new file mode 100644
index 0000000..883c2d8
--- /dev/null
+++ b/talk/app/webrtc/test/androidtestinitializer.cc
@@ -0,0 +1,74 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "talk/app/webrtc/test/androidtestinitializer.h"
+
+#include <pthread.h>
+
+// Note: this dependency is dangerous since it reaches into Chromium's base.
+// There's a risk of e.g. macro clashes. This file may only be used in tests.
+// Since we use Chromes build system for creating the gtest binary, this should
+// be fine.
+#include "base/android/context_utils.h"
+#include "base/android/jni_android.h"
+
+#include "talk/app/webrtc/java/jni/classreferenceholder.h"
+#include "talk/app/webrtc/java/jni/jni_helpers.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/ssladapter.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+
+namespace webrtc {
+
+namespace {
+
+static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
+
+// There can only be one JNI_OnLoad in each binary. So since this is a GTEST
+// C++ runner binary, we want to initialize the same global objects we normally
+// do if this had been a Java binary.
+void EnsureInitializedOnce() {
+ RTC_CHECK(::base::android::IsVMInitialized());
+ JNIEnv* jni = ::base::android::AttachCurrentThread();
+ JavaVM* jvm = NULL;
+ RTC_CHECK_EQ(0, jni->GetJavaVM(&jvm));
+ jobject context = ::base::android::GetApplicationContext();
+
+ RTC_CHECK_GE(webrtc_jni::InitGlobalJniVariables(jvm), 0);
+ RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
+ webrtc_jni::LoadGlobalClassReferenceHolder();
+
+ webrtc::VoiceEngine::SetAndroidObjects(jvm, context);
+}
+
+} // anonymous namespace
+
+void InitializeAndroidObjects() {
+ RTC_CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
+}
+
+} // namespace webrtc
diff --git a/talk/app/webrtc/test/androidtestinitializer.h b/talk/app/webrtc/test/androidtestinitializer.h
new file mode 100644
index 0000000..e699282
--- /dev/null
+++ b/talk/app/webrtc/test/androidtestinitializer.h
@@ -0,0 +1,37 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef TALK_APP_WEBRTC_TEST_ANDROIDTESTINITIALIZER_H_
+#define TALK_APP_WEBRTC_TEST_ANDROIDTESTINITIALIZER_H_
+
+namespace webrtc {
+
+void InitializeAndroidObjects();
+
+} // namespace webrtc
+
+#endif // TALK_APP_WEBRTC_TEST_ANDROIDTESTINITIALIZER_H_
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
index e2dc123..6b675a9 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
@@ -58,7 +58,7 @@
int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
@@ -82,7 +82,7 @@
// ADM is pulling data.
int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
- const uint8_t nChannels,
+ const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
diff --git a/talk/app/webrtc/test/fakedtlsidentitystore.h b/talk/app/webrtc/test/fakedtlsidentitystore.h
index 0f9bdb9..98074c7 100644
--- a/talk/app/webrtc/test/fakedtlsidentitystore.h
+++ b/talk/app/webrtc/test/fakedtlsidentitystore.h
@@ -29,41 +29,73 @@
#define TALK_APP_WEBRTC_TEST_FAKEDTLSIDENTITYSERVICE_H_
#include <string>
+#include <utility>
#include "talk/app/webrtc/dtlsidentitystore.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "webrtc/base/rtccertificate.h"
-static const char kRSA_PRIVATE_KEY_PEM[] =
- "-----BEGIN RSA PRIVATE KEY-----\n"
- "MIICdwIBADANBgkqhkiG9w0BAQEFAASCAmEwggJdAgEAAoGBAMYRkbhmI7kVA/rM\n"
- "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
- "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
- "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAECgYAvgOs4FJcgvp+TuREx7YtiYVsH\n"
- "mwQPTum2z/8VzWGwR8BBHBvIpVe1MbD/Y4seyI2aco/7UaisatSgJhsU46/9Y4fq\n"
- "2TwXH9QANf4at4d9n/R6rzwpAJOpgwZgKvdQjkfrKTtgLV+/dawvpxUYkRH4JZM1\n"
- "CVGukMfKNrSVH4Ap4QJBAOJmGV1ASPnB4r4nc99at7JuIJmd7fmuVUwUgYi4XgaR\n"
- "WhScBsgYwZ/JoywdyZJgnbcrTDuVcWG56B3vXbhdpMsCQQDf9zeJrjnPZ3Cqm79y\n"
- "kdqANep0uwZciiNiWxsQrCHztywOvbFhdp8iYVFG9EK8DMY41Y5TxUwsHD+67zao\n"
- "ZNqJAkEA1suLUP/GvL8IwuRneQd2tWDqqRQ/Td3qq03hP7e77XtF/buya3Ghclo5\n"
- "54czUR89QyVfJEC6278nzA7n2h1uVQJAcG6mztNL6ja/dKZjYZye2CY44QjSlLo0\n"
- "MTgTSjdfg/28fFn2Jjtqf9Pi/X+50LWI/RcYMC2no606wRk9kyOuIQJBAK6VSAim\n"
- "1pOEjsYQn0X5KEIrz1G3bfCbB848Ime3U2/FWlCHMr6ch8kCZ5d1WUeJD3LbwMNG\n"
- "UCXiYxSsu20QNVw=\n"
- "-----END RSA PRIVATE KEY-----\n";
-
-static const char kCERT_PEM[] =
- "-----BEGIN CERTIFICATE-----\n"
- "MIIBmTCCAQKgAwIBAgIEbzBSAjANBgkqhkiG9w0BAQsFADARMQ8wDQYDVQQDEwZX\n"
- "ZWJSVEMwHhcNMTQwMTAyMTgyNDQ3WhcNMTQwMjAxMTgyNDQ3WjARMQ8wDQYDVQQD\n"
- "EwZXZWJSVEMwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAMYRkbhmI7kVA/rM\n"
- "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
- "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
- "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAEwDQYJKoZIhvcNAQELBQADgYEAUflI\n"
- "VUe5Krqf5RVa5C3u/UTAOAUJBiDS3VANTCLBxjuMsvqOG0WvaYWP3HYPgrz0jXK2\n"
- "LJE/mGw3MyFHEqi81jh95J+ypl6xKW6Rm8jKLR87gUvCaVYn/Z4/P3AqcQTB7wOv\n"
- "UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n"
- "-----END CERTIFICATE-----\n";
+static const struct {
+ const char* rsa_private_key_pem;
+ const char* cert_pem;
+} kKeysAndCerts[] = {
+ {"-----BEGIN RSA PRIVATE KEY-----\n"
+ "MIICdwIBADANBgkqhkiG9w0BAQEFAASCAmEwggJdAgEAAoGBAMYRkbhmI7kVA/rM\n"
+ "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
+ "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
+ "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAECgYAvgOs4FJcgvp+TuREx7YtiYVsH\n"
+ "mwQPTum2z/8VzWGwR8BBHBvIpVe1MbD/Y4seyI2aco/7UaisatSgJhsU46/9Y4fq\n"
+ "2TwXH9QANf4at4d9n/R6rzwpAJOpgwZgKvdQjkfrKTtgLV+/dawvpxUYkRH4JZM1\n"
+ "CVGukMfKNrSVH4Ap4QJBAOJmGV1ASPnB4r4nc99at7JuIJmd7fmuVUwUgYi4XgaR\n"
+ "WhScBsgYwZ/JoywdyZJgnbcrTDuVcWG56B3vXbhdpMsCQQDf9zeJrjnPZ3Cqm79y\n"
+ "kdqANep0uwZciiNiWxsQrCHztywOvbFhdp8iYVFG9EK8DMY41Y5TxUwsHD+67zao\n"
+ "ZNqJAkEA1suLUP/GvL8IwuRneQd2tWDqqRQ/Td3qq03hP7e77XtF/buya3Ghclo5\n"
+ "54czUR89QyVfJEC6278nzA7n2h1uVQJAcG6mztNL6ja/dKZjYZye2CY44QjSlLo0\n"
+ "MTgTSjdfg/28fFn2Jjtqf9Pi/X+50LWI/RcYMC2no606wRk9kyOuIQJBAK6VSAim\n"
+ "1pOEjsYQn0X5KEIrz1G3bfCbB848Ime3U2/FWlCHMr6ch8kCZ5d1WUeJD3LbwMNG\n"
+ "UCXiYxSsu20QNVw=\n"
+ "-----END RSA PRIVATE KEY-----\n",
+ "-----BEGIN CERTIFICATE-----\n"
+ "MIIBmTCCAQKgAwIBAgIEbzBSAjANBgkqhkiG9w0BAQsFADARMQ8wDQYDVQQDEwZX\n"
+ "ZWJSVEMwHhcNMTQwMTAyMTgyNDQ3WhcNMTQwMjAxMTgyNDQ3WjARMQ8wDQYDVQQD\n"
+ "EwZXZWJSVEMwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAMYRkbhmI7kVA/rM\n"
+ "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
+ "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
+ "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAEwDQYJKoZIhvcNAQELBQADgYEAUflI\n"
+ "VUe5Krqf5RVa5C3u/UTAOAUJBiDS3VANTCLBxjuMsvqOG0WvaYWP3HYPgrz0jXK2\n"
+ "LJE/mGw3MyFHEqi81jh95J+ypl6xKW6Rm8jKLR87gUvCaVYn/Z4/P3AqcQTB7wOv\n"
+ "UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n"
+ "-----END CERTIFICATE-----\n"},
+ {"-----BEGIN RSA PRIVATE KEY-----\n"
+ "MIICXQIBAAKBgQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgj\n"
+ "Bl8CPZMvDh9EwB4Ji9ytA8dZZbQ4WbJWPr73zPpJSCvQqz6sOXSlenBRi72acNaQ\n"
+ "sOR/qPvviJx5I6Hqo4qemfnjZhAW85a5BpgrAwKgMLIQTHCTLWwVSyrDrwIDAQAB\n"
+ "AoGARni9eY8/hv+SX+I+05EdXt6MQXNUbQ+cSykBNCfVccLzIFEWUQMT2IHqwl6X\n"
+ "ShIXcq7/n1QzOAEiuzixauM3YHg4xZ1Um2Ha9a7ig5Xg4v6b43bmMkNE6LkoAtYs\n"
+ "qnQdfMh442b1liDud6IMb1Qk0amt3fSrgRMc547TZQVx4QECQQDxUeDm94r3p4ng\n"
+ "5rCLLC1K5/6HSTZsh7jatKPlz7GfP/IZlYV7iE5784/n0wRiCjZOS7hQRy/8m2Gp\n"
+ "pf4aZq+DAkEA6+np4d36FYikydvUrupLT3FkdRHGn/v83qOll/VmeNh+L1xMZlIP\n"
+ "tM26hAXCcQb7O5+J9y3cx2CAQsBS11ZXZQJAfGgTo76WG9p5UEJdXUInD2jOZPwv\n"
+ "XIATolxh6kXKcijLLLlSmT7KB0inNYIpzkkpee+7U1d/u6B3FriGaSHq9QJBAM/J\n"
+ "ICnDdLCgwNvWVraVQC3BpwSB2pswvCFwq7py94V60XFvbw80Ogc6qIv98qvQxVlX\n"
+ "hJIEgA/PjEi+0ng94Q0CQQDm8XSDby35gmjO+6eRmJtAjtB7nguLvrPXM6CPXRmD\n"
+ "sRoBocpHw6j9UdzZ6qYG0FkdXZghezXFY58ro2BYYRR3\n"
+ "-----END RSA PRIVATE KEY-----\n",
+ "-----BEGIN CERTIFICATE-----\n"
+ "MIICWDCCAcGgAwIBAgIJALgDjxMbBOhbMA0GCSqGSIb3DQEBCwUAMEUxCzAJBgNV\n"
+ "BAYTAkFVMRMwEQYDVQQIDApTb21lLVN0YXRlMSEwHwYDVQQKDBhJbnRlcm5ldCBX\n"
+ "aWRnaXRzIFB0eSBMdGQwHhcNMTUxMTEzMjIzMjEzWhcNMTYxMTEyMjIzMjEzWjBF\n"
+ "MQswCQYDVQQGEwJBVTETMBEGA1UECAwKU29tZS1TdGF0ZTEhMB8GA1UECgwYSW50\n"
+ "ZXJuZXQgV2lkZ2l0cyBQdHkgTHRkMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKB\n"
+ "gQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgjBl8CPZMvDh9E\n"
+ "wB4Ji9ytA8dZZbQ4WbJWPr73zPpJSCvQqz6sOXSlenBRi72acNaQsOR/qPvviJx5\n"
+ "I6Hqo4qemfnjZhAW85a5BpgrAwKgMLIQTHCTLWwVSyrDrwIDAQABo1AwTjAdBgNV\n"
+ "HQ4EFgQUx2tbJdlcSTCepn09UdYORXKuSTAwHwYDVR0jBBgwFoAUx2tbJdlcSTCe\n"
+ "pn09UdYORXKuSTAwDAYDVR0TBAUwAwEB/zANBgkqhkiG9w0BAQsFAAOBgQAmp9Id\n"
+ "E716gHMqeBG4S2FCgVFCr0a0ugkaneQAN/c2L9CbMemEN9W6jvucUIVOtYd90dDW\n"
+ "lXuowWmT/JctPe3D2qt4yvYW3puECHk2tVQmrJOZiZiTRtWm6HxkmoUYHYp/DtaS\n"
+ "1Xe29gSTnZtI5sQCrGMzk3SGRSSs7ejLKiVDBQ==\n"
+ "-----END CERTIFICATE-----\n"}};
class FakeDtlsIdentityStore : public webrtc::DtlsIdentityStoreInterface,
public rtc::MessageHandler {
@@ -77,6 +109,9 @@
should_fail_ = should_fail;
}
+ void use_original_key() { key_index_ = 0; }
+ void use_alternate_key() { key_index_ = 1; }
+
void RequestIdentity(
rtc::KeyType key_type,
const rtc::scoped_refptr<webrtc::DtlsIdentityRequestObserver>&
@@ -92,8 +127,9 @@
static rtc::scoped_refptr<rtc::RTCCertificate> GenerateCertificate() {
std::string cert;
std::string key;
- rtc::SSLIdentity::PemToDer("CERTIFICATE", kCERT_PEM, &cert);
- rtc::SSLIdentity::PemToDer("RSA PRIVATE KEY", kRSA_PRIVATE_KEY_PEM, &key);
+ rtc::SSLIdentity::PemToDer("CERTIFICATE", kKeysAndCerts[0].cert_pem, &cert);
+ rtc::SSLIdentity::PemToDer("RSA PRIVATE KEY",
+ kKeysAndCerts[0].rsa_private_key_pem, &key);
std::string pem_cert = rtc::SSLIdentity::DerToPem(
rtc::kPemTypeCertificate,
@@ -106,7 +142,7 @@
rtc::scoped_ptr<rtc::SSLIdentity> identity(
rtc::SSLIdentity::FromPEMStrings(pem_key, pem_cert));
- return rtc::RTCCertificate::Create(identity.Pass());
+ return rtc::RTCCertificate::Create(std::move(identity));
}
private:
@@ -115,6 +151,11 @@
MSG_FAILURE,
};
+ const char* get_key() {
+ return kKeysAndCerts[key_index_].rsa_private_key_pem;
+ }
+ const char* get_cert() { return kKeysAndCerts[key_index_].cert_pem; }
+
// rtc::MessageHandler implementation.
void OnMessage(rtc::Message* msg) {
MessageData* message_data = static_cast<MessageData*>(msg->pdata);
@@ -124,9 +165,8 @@
case MSG_SUCCESS: {
std::string cert;
std::string key;
- rtc::SSLIdentity::PemToDer("CERTIFICATE", kCERT_PEM, &cert);
- rtc::SSLIdentity::PemToDer("RSA PRIVATE KEY", kRSA_PRIVATE_KEY_PEM,
- &key);
+ rtc::SSLIdentity::PemToDer("CERTIFICATE", get_cert(), &cert);
+ rtc::SSLIdentity::PemToDer("RSA PRIVATE KEY", get_key(), &key);
observer->OnSuccess(cert, key);
break;
}
@@ -138,6 +178,7 @@
}
bool should_fail_;
+ int key_index_ = 0;
};
#endif // TALK_APP_WEBRTC_TEST_FAKEDTLSIDENTITYSERVICE_H_
diff --git a/talk/app/webrtc/test/fakemediastreamsignaling.h b/talk/app/webrtc/test/fakemediastreamsignaling.h
deleted file mode 100644
index 562c4ad..0000000
--- a/talk/app/webrtc/test/fakemediastreamsignaling.h
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- * libjingle
- * Copyright 2013 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
-#define TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
-
-#include "talk/app/webrtc/audiotrack.h"
-#include "talk/app/webrtc/mediastreamsignaling.h"
-#include "talk/app/webrtc/videotrack.h"
-
-static const char kStream1[] = "stream1";
-static const char kVideoTrack1[] = "video1";
-static const char kAudioTrack1[] = "audio1";
-
-static const char kStream2[] = "stream2";
-static const char kVideoTrack2[] = "video2";
-static const char kAudioTrack2[] = "audio2";
-
-class FakeMediaStreamSignaling : public webrtc::MediaStreamSignaling,
- public webrtc::MediaStreamSignalingObserver {
- public:
- explicit FakeMediaStreamSignaling(cricket::ChannelManager* channel_manager) :
- webrtc::MediaStreamSignaling(rtc::Thread::Current(), this,
- channel_manager) {
- }
-
- void SendAudioVideoStream1() {
- ClearLocalStreams();
- AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
- }
-
- void SendAudioVideoStream2() {
- ClearLocalStreams();
- AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
- }
-
- void SendAudioVideoStream1And2() {
- ClearLocalStreams();
- AddLocalStream(CreateStream(kStream1, kAudioTrack1, kVideoTrack1));
- AddLocalStream(CreateStream(kStream2, kAudioTrack2, kVideoTrack2));
- }
-
- void SendNothing() {
- ClearLocalStreams();
- }
-
- void UseOptionsAudioOnly() {
- ClearLocalStreams();
- AddLocalStream(CreateStream(kStream2, kAudioTrack2, ""));
- }
-
- void UseOptionsVideoOnly() {
- ClearLocalStreams();
- AddLocalStream(CreateStream(kStream2, "", kVideoTrack2));
- }
-
- void ClearLocalStreams() {
- while (local_streams()->count() != 0) {
- RemoveLocalStream(local_streams()->at(0));
- }
- }
-
- // Implements MediaStreamSignalingObserver.
- virtual void OnAddRemoteStream(webrtc::MediaStreamInterface* stream) {}
- virtual void OnRemoveRemoteStream(webrtc::MediaStreamInterface* stream) {}
- virtual void OnAddDataChannel(webrtc::DataChannelInterface* data_channel) {}
- virtual void OnAddLocalAudioTrack(webrtc::MediaStreamInterface* stream,
- webrtc::AudioTrackInterface* audio_track,
- uint32_t ssrc) {}
- virtual void OnAddLocalVideoTrack(webrtc::MediaStreamInterface* stream,
- webrtc::VideoTrackInterface* video_track,
- uint32_t ssrc) {}
- virtual void OnAddRemoteAudioTrack(webrtc::MediaStreamInterface* stream,
- webrtc::AudioTrackInterface* audio_track,
- uint32_t ssrc) {}
- virtual void OnAddRemoteVideoTrack(webrtc::MediaStreamInterface* stream,
- webrtc::VideoTrackInterface* video_track,
- uint32_t ssrc) {}
- virtual void OnRemoveRemoteAudioTrack(
- webrtc::MediaStreamInterface* stream,
- webrtc::AudioTrackInterface* audio_track) {}
- virtual void OnRemoveRemoteVideoTrack(
- webrtc::MediaStreamInterface* stream,
- webrtc::VideoTrackInterface* video_track) {}
- virtual void OnRemoveLocalAudioTrack(webrtc::MediaStreamInterface* stream,
- webrtc::AudioTrackInterface* audio_track,
- uint32_t ssrc) {}
- virtual void OnRemoveLocalVideoTrack(
- webrtc::MediaStreamInterface* stream,
- webrtc::VideoTrackInterface* video_track) {}
- virtual void OnRemoveLocalStream(webrtc::MediaStreamInterface* stream) {}
-
- private:
- rtc::scoped_refptr<webrtc::MediaStreamInterface> CreateStream(
- const std::string& stream_label,
- const std::string& audio_track_id,
- const std::string& video_track_id) {
- rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
- webrtc::MediaStream::Create(stream_label));
-
- if (!audio_track_id.empty()) {
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
- webrtc::AudioTrack::Create(audio_track_id, NULL));
- stream->AddTrack(audio_track);
- }
-
- if (!video_track_id.empty()) {
- rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
- webrtc::VideoTrack::Create(video_track_id, NULL));
- stream->AddTrack(video_track);
- }
- return stream;
- }
-};
-
-#endif // TALK_APP_WEBRTC_TEST_FAKEMEDIASTREAMSIGNALING_H_
diff --git a/talk/app/webrtc/test/peerconnectiontestwrapper.cc b/talk/app/webrtc/test/peerconnectiontestwrapper.cc
index 2eb24d9..86b7842 100644
--- a/talk/app/webrtc/test/peerconnectiontestwrapper.cc
+++ b/talk/app/webrtc/test/peerconnectiontestwrapper.cc
@@ -25,13 +25,15 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
-#include "talk/app/webrtc/fakeportallocatorfactory.h"
+#include <utility>
+
#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
#include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
#include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
#include "talk/app/webrtc/test/peerconnectiontestwrapper.h"
#include "talk/app/webrtc/videosourceinterface.h"
#include "webrtc/base/gunit.h"
+#include "webrtc/p2p/client/fakeportallocator.h"
static const char kStreamLabelBase[] = "stream_label";
static const char kVideoTrackLabelBase[] = "video_track";
@@ -70,10 +72,8 @@
bool PeerConnectionTestWrapper::CreatePc(
const MediaConstraintsInterface* constraints) {
- allocator_factory_ = webrtc::FakePortAllocatorFactory::Create();
- if (!allocator_factory_) {
- return false;
- }
+ rtc::scoped_ptr<cricket::PortAllocator> port_allocator(
+ new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
if (fake_audio_capture_module_ == NULL) {
@@ -87,17 +87,17 @@
return false;
}
- // CreatePeerConnection with IceServers.
- webrtc::PeerConnectionInterface::IceServers ice_servers;
+ // CreatePeerConnection with RTCConfiguration.
+ webrtc::PeerConnectionInterface::RTCConfiguration config;
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = "stun:stun.l.google.com:19302";
- ice_servers.push_back(ice_server);
+ config.servers.push_back(ice_server);
rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store(
rtc::SSLStreamAdapter::HaveDtlsSrtp() ?
new FakeDtlsIdentityStore() : nullptr);
peer_connection_ = peer_connection_factory_->CreatePeerConnection(
- ice_servers, constraints, allocator_factory_.get(),
- dtls_identity_store.Pass(), this);
+ config, constraints, std::move(port_allocator),
+ std::move(dtls_identity_store), this);
return peer_connection_.get() != NULL;
}
diff --git a/talk/app/webrtc/test/peerconnectiontestwrapper.h b/talk/app/webrtc/test/peerconnectiontestwrapper.h
index b654263..883f2f2 100644
--- a/talk/app/webrtc/test/peerconnectiontestwrapper.h
+++ b/talk/app/webrtc/test/peerconnectiontestwrapper.h
@@ -34,11 +34,6 @@
#include "talk/app/webrtc/test/fakevideotrackrenderer.h"
#include "webrtc/base/sigslot.h"
-namespace webrtc {
-class DtlsIdentityStoreInterface;
-class PortAllocatorFactoryInterface;
-}
-
class PeerConnectionTestWrapper
: public webrtc::PeerConnectionObserver,
public webrtc::CreateSessionDescriptionObserver,
@@ -110,8 +105,6 @@
bool video, const webrtc::FakeConstraints& video_constraints);
std::string name_;
- rtc::scoped_refptr<webrtc::PortAllocatorFactoryInterface>
- allocator_factory_;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
peer_connection_factory_;
diff --git a/talk/app/webrtc/videosource.cc b/talk/app/webrtc/videosource.cc
index b33f5f9..4b371e3 100644
--- a/talk/app/webrtc/videosource.cc
+++ b/talk/app/webrtc/videosource.cc
@@ -32,6 +32,7 @@
#include "talk/app/webrtc/mediaconstraintsinterface.h"
#include "talk/session/media/channelmanager.h"
+#include "webrtc/base/arraysize.h"
using cricket::CaptureState;
using webrtc::MediaConstraintsInterface;
@@ -267,11 +268,12 @@
// Set |option| to the highest-priority value of |key| in the constraints.
// Return false if the key is mandatory, and the value is invalid.
bool ExtractOption(const MediaConstraintsInterface* all_constraints,
- const std::string& key, cricket::Settable<bool>* option) {
+ const std::string& key,
+ rtc::Optional<bool>* option) {
size_t mandatory = 0;
bool value;
if (FindConstraint(all_constraints, key, &value, &mandatory)) {
- option->Set(value);
+ *option = rtc::Optional<bool>(value);
return true;
}
@@ -302,8 +304,6 @@
virtual ~FrameInputWrapper() {}
// VideoRenderer implementation.
- bool SetSize(int width, int height, int reserved) override { return true; }
-
bool RenderFrame(const cricket::VideoFrame* frame) override {
if (!capturer_->IsRunning()) {
return true;
@@ -329,21 +329,23 @@
rtc::scoped_refptr<VideoSource> VideoSource::Create(
cricket::ChannelManager* channel_manager,
cricket::VideoCapturer* capturer,
- const webrtc::MediaConstraintsInterface* constraints) {
+ const webrtc::MediaConstraintsInterface* constraints,
+ bool remote) {
ASSERT(channel_manager != NULL);
ASSERT(capturer != NULL);
- rtc::scoped_refptr<VideoSource> source(
- new rtc::RefCountedObject<VideoSource>(channel_manager,
- capturer));
+ rtc::scoped_refptr<VideoSource> source(new rtc::RefCountedObject<VideoSource>(
+ channel_manager, capturer, remote));
source->Initialize(constraints);
return source;
}
VideoSource::VideoSource(cricket::ChannelManager* channel_manager,
- cricket::VideoCapturer* capturer)
+ cricket::VideoCapturer* capturer,
+ bool remote)
: channel_manager_(channel_manager),
video_capturer_(capturer),
- state_(kInitializing) {
+ state_(kInitializing),
+ remote_(remote) {
channel_manager_->SignalVideoCaptureStateChange.connect(
this, &VideoSource::OnStateChange);
}
@@ -368,7 +370,7 @@
} else {
// The VideoCapturer implementation doesn't support capability
// enumeration. We need to guess what the camera supports.
- for (int i = 0; i < ARRAY_SIZE(kVideoFormats); ++i) {
+ for (int i = 0; i < arraysize(kVideoFormats); ++i) {
formats.push_back(cricket::VideoFormat(kVideoFormats[i]));
}
}
@@ -460,7 +462,9 @@
}
void VideoSource::SetState(SourceState new_state) {
- if (VERIFY(state_ != new_state)) {
+ // TODO(hbos): Temporarily disabled VERIFY due to webrtc:4776.
+ // if (VERIFY(state_ != new_state)) {
+ if (state_ != new_state) {
state_ = new_state;
FireOnChanged();
}
diff --git a/talk/app/webrtc/videosource.h b/talk/app/webrtc/videosource.h
index 8253cba..98c1e08 100644
--- a/talk/app/webrtc/videosource.h
+++ b/talk/app/webrtc/videosource.h
@@ -66,9 +66,12 @@
static rtc::scoped_refptr<VideoSource> Create(
cricket::ChannelManager* channel_manager,
cricket::VideoCapturer* capturer,
- const webrtc::MediaConstraintsInterface* constraints);
+ const webrtc::MediaConstraintsInterface* constraints,
+ bool remote);
- virtual SourceState state() const { return state_; }
+ SourceState state() const override { return state_; }
+ bool remote() const override { return remote_; }
+
virtual const cricket::VideoOptions* options() const { return &options_; }
virtual cricket::VideoRenderer* FrameInput();
@@ -86,7 +89,8 @@
protected:
VideoSource(cricket::ChannelManager* channel_manager,
- cricket::VideoCapturer* capturer);
+ cricket::VideoCapturer* capturer,
+ bool remote);
virtual ~VideoSource();
void Initialize(const webrtc::MediaConstraintsInterface* constraints);
@@ -104,6 +108,7 @@
cricket::VideoFormat format_;
cricket::VideoOptions options_;
SourceState state_;
+ const bool remote_;
};
} // namespace webrtc
diff --git a/talk/app/webrtc/videosource_unittest.cc b/talk/app/webrtc/videosource_unittest.cc
index 2efcc1d..6f1df34 100644
--- a/talk/app/webrtc/videosource_unittest.cc
+++ b/talk/app/webrtc/videosource_unittest.cc
@@ -144,9 +144,9 @@
void CreateVideoSource(
const webrtc::MediaConstraintsInterface* constraints) {
// VideoSource take ownership of |capturer_|
- source_ = VideoSource::Create(channel_manager_.get(),
- capturer_cleanup_.release(),
- constraints);
+ source_ =
+ VideoSource::Create(channel_manager_.get(), capturer_cleanup_.release(),
+ constraints, false);
ASSERT_TRUE(source_.get() != NULL);
EXPECT_EQ(capturer_, source_->GetVideoCapturer());
@@ -210,8 +210,7 @@
// RemoteVideoCapturer and takes video frames from FrameInput.
TEST_F(VideoSourceTest, StartStopRemote) {
source_ = VideoSource::Create(channel_manager_.get(),
- new webrtc::RemoteVideoCapturer(),
- NULL);
+ new webrtc::RemoteVideoCapturer(), NULL, true);
ASSERT_TRUE(source_.get() != NULL);
EXPECT_TRUE(NULL != source_->GetVideoCapturer());
@@ -392,16 +391,14 @@
CreateVideoSource(&constraints);
- bool value = true;
- EXPECT_TRUE(source_->options()->video_noise_reduction.Get(&value));
- EXPECT_FALSE(value);
+ EXPECT_EQ(rtc::Optional<bool>(false),
+ source_->options()->video_noise_reduction);
}
TEST_F(VideoSourceTest, OptionNotSet) {
FakeConstraints constraints;
CreateVideoSource(&constraints);
- bool value;
- EXPECT_FALSE(source_->options()->video_noise_reduction.Get(&value));
+ EXPECT_EQ(rtc::Optional<bool>(), source_->options()->video_noise_reduction);
}
TEST_F(VideoSourceTest, MandatoryOptionOverridesOptional) {
@@ -413,9 +410,8 @@
CreateVideoSource(&constraints);
- bool value = false;
- EXPECT_TRUE(source_->options()->video_noise_reduction.Get(&value));
- EXPECT_TRUE(value);
+ EXPECT_EQ(rtc::Optional<bool>(true),
+ source_->options()->video_noise_reduction);
}
TEST_F(VideoSourceTest, InvalidOptionKeyOptional) {
@@ -428,9 +424,8 @@
EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
kMaxWaitMs);
- bool value = true;
- EXPECT_TRUE(source_->options()->video_noise_reduction.Get(&value));
- EXPECT_FALSE(value);
+ EXPECT_EQ(rtc::Optional<bool>(false),
+ source_->options()->video_noise_reduction);
}
TEST_F(VideoSourceTest, InvalidOptionKeyMandatory) {
@@ -443,8 +438,7 @@
EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
kMaxWaitMs);
- bool value;
- EXPECT_FALSE(source_->options()->video_noise_reduction.Get(&value));
+ EXPECT_EQ(rtc::Optional<bool>(), source_->options()->video_noise_reduction);
}
TEST_F(VideoSourceTest, InvalidOptionValueOptional) {
@@ -456,8 +450,7 @@
EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
kMaxWaitMs);
- bool value = false;
- EXPECT_FALSE(source_->options()->video_noise_reduction.Get(&value));
+ EXPECT_EQ(rtc::Optional<bool>(), source_->options()->video_noise_reduction);
}
TEST_F(VideoSourceTest, InvalidOptionValueMandatory) {
@@ -473,8 +466,7 @@
EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
kMaxWaitMs);
- bool value;
- EXPECT_FALSE(source_->options()->video_noise_reduction.Get(&value));
+ EXPECT_EQ(rtc::Optional<bool>(), source_->options()->video_noise_reduction);
}
TEST_F(VideoSourceTest, MixedOptionsAndConstraints) {
@@ -497,9 +489,8 @@
EXPECT_EQ(288, format->height);
EXPECT_EQ(30, format->framerate());
- bool value = true;
- EXPECT_TRUE(source_->options()->video_noise_reduction.Get(&value));
- EXPECT_FALSE(value);
+ EXPECT_EQ(rtc::Optional<bool>(false),
+ source_->options()->video_noise_reduction);
}
// Tests that the source starts video with the default resolution for
diff --git a/talk/app/webrtc/videosourceproxy.h b/talk/app/webrtc/videosourceproxy.h
index 677fa9c..ce96e8e 100644
--- a/talk/app/webrtc/videosourceproxy.h
+++ b/talk/app/webrtc/videosourceproxy.h
@@ -38,6 +38,7 @@
// signaling thread.
BEGIN_PROXY_MAP(VideoSource)
PROXY_CONSTMETHOD0(SourceState, state)
+ PROXY_CONSTMETHOD0(bool, remote)
PROXY_METHOD0(cricket::VideoCapturer*, GetVideoCapturer)
PROXY_METHOD0(void, Stop)
PROXY_METHOD0(void, Restart)
diff --git a/talk/app/webrtc/videotrack.cc b/talk/app/webrtc/videotrack.cc
index 7c78aea..f138240 100644
--- a/talk/app/webrtc/videotrack.cc
+++ b/talk/app/webrtc/videotrack.cc
@@ -31,7 +31,7 @@
namespace webrtc {
-static const char kVideoTrackKind[] = "video";
+const char MediaStreamTrackInterface::kVideoKind[] = "video";
VideoTrack::VideoTrack(const std::string& label,
VideoSourceInterface* video_source)
@@ -47,7 +47,7 @@
}
std::string VideoTrack::kind() const {
- return kVideoTrackKind;
+ return kVideoKind;
}
void VideoTrack::AddRenderer(VideoRendererInterface* renderer) {
diff --git a/talk/app/webrtc/videotrack_unittest.cc b/talk/app/webrtc/videotrack_unittest.cc
index 609ee80..013d925 100644
--- a/talk/app/webrtc/videotrack_unittest.cc
+++ b/talk/app/webrtc/videotrack_unittest.cc
@@ -62,7 +62,7 @@
video_track_ = VideoTrack::Create(
kVideoTrackId,
VideoSource::Create(channel_manager_.get(),
- new webrtc::RemoteVideoCapturer(), NULL));
+ new webrtc::RemoteVideoCapturer(), NULL, true));
}
protected:
diff --git a/talk/app/webrtc/videotrackrenderers.cc b/talk/app/webrtc/videotrackrenderers.cc
index 3c47c6e..3f9301b 100644
--- a/talk/app/webrtc/videotrackrenderers.cc
+++ b/talk/app/webrtc/videotrackrenderers.cc
@@ -54,10 +54,6 @@
enabled_ = enable;
}
-bool VideoTrackRenderers::SetSize(int width, int height, int reserved) {
- return true;
-}
-
bool VideoTrackRenderers::RenderFrame(const cricket::VideoFrame* frame) {
rtc::CritScope cs(&critical_section_);
if (!enabled_) {
diff --git a/talk/app/webrtc/videotrackrenderers.h b/talk/app/webrtc/videotrackrenderers.h
index 15274a1..3262e22 100644
--- a/talk/app/webrtc/videotrackrenderers.h
+++ b/talk/app/webrtc/videotrackrenderers.h
@@ -48,7 +48,6 @@
~VideoTrackRenderers();
// Implements cricket::VideoRenderer
- virtual bool SetSize(int width, int height, int reserved);
virtual bool RenderFrame(const cricket::VideoFrame* frame);
void AddRenderer(VideoRendererInterface* renderer);
diff --git a/talk/app/webrtc/webrtcsdp.cc b/talk/app/webrtc/webrtcsdp.cc
index 3fa9a7d..e287e90 100644
--- a/talk/app/webrtc/webrtcsdp.cc
+++ b/talk/app/webrtc/webrtcsdp.cc
@@ -45,6 +45,7 @@
#include "webrtc/p2p/base/constants.h"
#include "webrtc/p2p/base/port.h"
#include "talk/session/media/mediasession.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/messagedigest.h"
@@ -121,6 +122,7 @@
static const char kAttributeGroup[] = "group";
static const char kAttributeMid[] = "mid";
static const char kAttributeRtcpMux[] = "rtcp-mux";
+static const char kAttributeRtcpReducedSize[] = "rtcp-rsize";
static const char kAttributeSsrc[] = "ssrc";
static const char kSsrcAttributeCname[] = "cname";
static const char kAttributeExtmap[] = "extmap";
@@ -138,8 +140,8 @@
static const char kAttributeCandidateTyp[] = "typ";
static const char kAttributeCandidateRaddr[] = "raddr";
static const char kAttributeCandidateRport[] = "rport";
-static const char kAttributeCandidateUsername[] = "username";
-static const char kAttributeCandidatePassword[] = "password";
+static const char kAttributeCandidateUfrag[] = "ufrag";
+static const char kAttributeCandidatePwd[] = "pwd";
static const char kAttributeCandidateGeneration[] = "generation";
static const char kAttributeFingerprint[] = "fingerprint";
static const char kAttributeSetup[] = "setup";
@@ -260,6 +262,7 @@
const MediaType media_type,
std::string* message);
static void BuildCandidate(const std::vector<Candidate>& candidates,
+ bool include_ufrag,
std::string* message);
static void BuildIceOptions(const std::vector<std::string>& transport_options,
std::string* message);
@@ -876,7 +879,7 @@
std::string message;
std::vector<cricket::Candidate> candidates;
candidates.push_back(candidate.candidate());
- BuildCandidate(candidates, &message);
+ BuildCandidate(candidates, true, &message);
// From WebRTC draft section 4.8.1.1 candidate-attribute will be
// just candidate:<candidate> not a=candidate:<blah>CRLF
ASSERT(message.find("a=") == 0);
@@ -1070,10 +1073,9 @@
}
// Extension
- // Empty string as the candidate username and password.
- // Will be updated later with the ice-ufrag and ice-pwd.
- // TODO: Remove the username/password extension, which is currently
- // kept for backwards compatibility.
+ // Though non-standard, we support the ICE ufrag and pwd being signaled on
+ // the candidate to avoid issues with confusing which generation a candidate
+ // belongs to when trickling multiple generations at the same time.
std::string username;
std::string password;
uint32_t generation = 0;
@@ -1084,9 +1086,9 @@
if (!GetValueFromString(first_line, fields[++i], &generation, error)) {
return false;
}
- } else if (fields[i] == kAttributeCandidateUsername) {
+ } else if (fields[i] == kAttributeCandidateUfrag) {
username = fields[++i];
- } else if (fields[i] == kAttributeCandidatePassword) {
+ } else if (fields[i] == kAttributeCandidatePwd) {
password = fields[++i];
} else {
// Skip the unknown extension.
@@ -1283,8 +1285,9 @@
}
}
- // Build the a=candidate lines.
- BuildCandidate(candidates, message);
+ // Build the a=candidate lines. We don't include ufrag and pwd in the
+ // candidates in the SDP to avoid redundancy.
+ BuildCandidate(candidates, false, message);
// Use the transport_info to build the media level ice-ufrag and ice-pwd.
if (transport_info) {
@@ -1292,13 +1295,17 @@
// ice-pwd-att = "ice-pwd" ":" password
// ice-ufrag-att = "ice-ufrag" ":" ufrag
// ice-ufrag
- InitAttrLine(kAttributeIceUfrag, &os);
- os << kSdpDelimiterColon << transport_info->description.ice_ufrag;
- AddLine(os.str(), message);
+ if (!transport_info->description.ice_ufrag.empty()) {
+ InitAttrLine(kAttributeIceUfrag, &os);
+ os << kSdpDelimiterColon << transport_info->description.ice_ufrag;
+ AddLine(os.str(), message);
+ }
// ice-pwd
- InitAttrLine(kAttributeIcePwd, &os);
- os << kSdpDelimiterColon << transport_info->description.ice_pwd;
- AddLine(os.str(), message);
+ if (!transport_info->description.ice_pwd.empty()) {
+ InitAttrLine(kAttributeIcePwd, &os);
+ os << kSdpDelimiterColon << transport_info->description.ice_pwd;
+ AddLine(os.str(), message);
+ }
// draft-petithuguenin-mmusic-ice-attributes-level-03
BuildIceOptions(transport_info->description.transport_options, message);
@@ -1399,6 +1406,13 @@
AddLine(os.str(), message);
}
+ // RFC 5506
+ // a=rtcp-rsize
+ if (media_desc->rtcp_reduced_size()) {
+ InitAttrLine(kAttributeRtcpReducedSize, &os);
+ AddLine(os.str(), message);
+ }
+
// RFC 4568
// a=crypto:<tag> <crypto-suite> <key-params> [<session-params>]
for (std::vector<CryptoParams>::const_iterator it =
@@ -1525,7 +1539,7 @@
kCodecParamMaxAverageBitrate, kCodecParamMaxPlaybackRate,
kCodecParamAssociatedPayloadType
};
- for (size_t i = 0; i < ARRAY_SIZE(kFmtpParams); ++i) {
+ for (size_t i = 0; i < arraysize(kFmtpParams); ++i) {
if (_stricmp(name.c_str(), kFmtpParams[i]) == 0) {
return true;
}
@@ -1708,6 +1722,7 @@
}
void BuildCandidate(const std::vector<Candidate>& candidates,
+ bool include_ufrag,
std::string* message) {
std::ostringstream os;
@@ -1757,6 +1772,9 @@
// Extensions
os << kAttributeCandidateGeneration << " " << it->generation();
+ if (include_ufrag && !it->username().empty()) {
+ os << " " << kAttributeCandidateUfrag << " " << it->username();
+ }
AddLine(os.str(), message);
}
@@ -2046,7 +2064,7 @@
struct StaticPayloadAudioCodec {
const char* name;
int clockrate;
- int channels;
+ size_t channels;
};
static const StaticPayloadAudioCodec kStaticPayloadAudioCodecs[] = {
{ "PCMU", 8000, 1 },
@@ -2082,10 +2100,10 @@
int payload_type = *it;
if (!media_desc->HasCodec(payload_type) &&
payload_type >= 0 &&
- payload_type < ARRAY_SIZE(kStaticPayloadAudioCodecs)) {
+ payload_type < arraysize(kStaticPayloadAudioCodecs)) {
std::string encoding_name = kStaticPayloadAudioCodecs[payload_type].name;
int clock_rate = kStaticPayloadAudioCodecs[payload_type].clockrate;
- int channels = kStaticPayloadAudioCodecs[payload_type].channels;
+ size_t channels = kStaticPayloadAudioCodecs[payload_type].channels;
media_desc->AddCodec(cricket::AudioCodec(payload_type, encoding_name,
clock_rate, 0, channels,
preference));
@@ -2552,6 +2570,8 @@
//
if (HasAttribute(line, kAttributeRtcpMux)) {
media_desc->set_rtcp_mux(true);
+ } else if (HasAttribute(line, kAttributeRtcpReducedSize)) {
+ media_desc->set_rtcp_reduced_size(true);
} else if (HasAttribute(line, kAttributeSsrcGroup)) {
if (!ParseSsrcGroupAttribute(line, &ssrc_groups, error)) {
return false;
@@ -2666,7 +2686,8 @@
// Update the candidates with the media level "ice-pwd" and "ice-ufrag".
for (Candidates::iterator it = candidates_orig.begin();
it != candidates_orig.end(); ++it) {
- ASSERT((*it).username().empty());
+ ASSERT((*it).username().empty() ||
+ (*it).username() == transport->ice_ufrag);
(*it).set_username(transport->ice_ufrag);
ASSERT((*it).password().empty());
(*it).set_password(transport->ice_pwd);
@@ -2817,7 +2838,7 @@
// Updates or creates a new codec entry in the audio description with according
// to |name|, |clockrate|, |bitrate|, |channels| and |preference|.
void UpdateCodec(int payload_type, const std::string& name, int clockrate,
- int bitrate, int channels, int preference,
+ int bitrate, size_t channels, int preference,
AudioContentDescription* audio_desc) {
// Codec may already be populated with (only) optional parameters
// (from an fmtp).
@@ -2916,7 +2937,7 @@
// of audio channels. This parameter is OPTIONAL and may be
// omitted if the number of channels is one, provided that no
// additional parameters are needed.
- int channels = 1;
+ size_t channels = 1;
if (codec_params.size() == 3) {
if (!GetValueFromString(line, codec_params[2], &channels, error)) {
return false;
diff --git a/talk/app/webrtc/webrtcsdp_unittest.cc b/talk/app/webrtc/webrtcsdp_unittest.cc
index cb6a392..15fc808 100644
--- a/talk/app/webrtc/webrtcsdp_unittest.cc
+++ b/talk/app/webrtc/webrtcsdp_unittest.cc
@@ -30,6 +30,9 @@
#include <vector>
#include "talk/app/webrtc/jsepsessiondescription.h"
+#ifdef WEBRTC_ANDROID
+#include "talk/app/webrtc/test/androidtestinitializer.h"
+#endif
#include "talk/app/webrtc/webrtcsdp.h"
#include "talk/media/base/constants.h"
#include "webrtc/p2p/base/constants.h"
@@ -80,11 +83,13 @@
static const uint32_t kCandidatePriority = 2130706432U; // pref = 1.0
static const char kCandidateUfragVoice[] = "ufrag_voice";
static const char kCandidatePwdVoice[] = "pwd_voice";
+static const char kAttributeIceUfragVoice[] = "a=ice-ufrag:ufrag_voice\r\n";
static const char kAttributeIcePwdVoice[] = "a=ice-pwd:pwd_voice\r\n";
static const char kCandidateUfragVideo[] = "ufrag_video";
static const char kCandidatePwdVideo[] = "pwd_video";
static const char kCandidateUfragData[] = "ufrag_data";
static const char kCandidatePwdData[] = "pwd_data";
+static const char kAttributeIceUfragVideo[] = "a=ice-ufrag:ufrag_video\r\n";
static const char kAttributeIcePwdVideo[] = "a=ice-pwd:pwd_video\r\n";
static const uint32_t kCandidateGeneration = 2;
static const char kCandidateFoundation1[] = "a0+B/1";
@@ -153,6 +158,7 @@
"a=mid:audio_content_name\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
+ "a=rtcp-rsize\r\n"
"a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
"inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
"dummy_session_params\r\n"
@@ -220,6 +226,7 @@
"a=mid:audio_content_name\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
+ "a=rtcp-rsize\r\n"
"a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
"inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
"dummy_session_params\r\n"
@@ -394,9 +401,9 @@
"abcd::abcd::abcd::abcd::abcd::abcd::abcd::abcd 1234 typ host generation 2";
// One candidate reference string.
-static const char kSdpOneCandidateOldFormat[] =
+static const char kSdpOneCandidateWithUfragPwd[] =
"a=candidate:a0+B/1 1 udp 2130706432 192.168.1.5 1234 typ host network_name"
- " eth0 username user_rtp password password_rtp generation 2\r\n";
+ " eth0 ufrag user_rtp pwd password_rtp generation 2\r\n";
// Session id and version
static const char kSessionId[] = "18446744069414584320";
@@ -523,10 +530,14 @@
static void ReplaceRejected(bool audio_rejected, bool video_rejected,
std::string* message) {
if (audio_rejected) {
- Replace("m=audio 2345", "m=audio 0", message);
+ Replace("m=audio 9", "m=audio 0", message);
+ Replace(kAttributeIceUfragVoice, "", message);
+ Replace(kAttributeIcePwdVoice, "", message);
}
if (video_rejected) {
- Replace("m=video 3457", "m=video 0", message);
+ Replace("m=video 9", "m=video 0", message);
+ Replace(kAttributeIceUfragVideo, "", message);
+ Replace(kAttributeIcePwdVideo, "", message);
}
}
@@ -536,6 +547,9 @@
public:
WebRtcSdpTest()
: jdesc_(kDummyString) {
+#ifdef WEBRTC_ANDROID
+ webrtc::InitializeAndroidObjects();
+#endif
// AudioContentDescription
audio_desc_ = CreateAudioContentDescription();
AudioCodec opus(111, "opus", 48000, 0, 2, 3);
@@ -704,6 +718,7 @@
AudioContentDescription* CreateAudioContentDescription() {
AudioContentDescription* audio = new AudioContentDescription();
audio->set_rtcp_mux(true);
+ audio->set_rtcp_reduced_size(true);
StreamParams audio_stream1;
audio_stream1.id = kAudioTrackId1;
audio_stream1.cname = kStream1Cname;
@@ -735,6 +750,9 @@
// rtcp_mux
EXPECT_EQ(cd1->rtcp_mux(), cd2->rtcp_mux());
+ // rtcp_reduced_size
+ EXPECT_EQ(cd1->rtcp_reduced_size(), cd2->rtcp_reduced_size());
+
// cryptos
EXPECT_EQ(cd1->cryptos().size(), cd2->cryptos().size());
if (cd1->cryptos().size() != cd2->cryptos().size()) {
@@ -979,6 +997,18 @@
desc_.AddTransportInfo(transport_info);
}
+ void SetIceUfragPwd(const std::string& content_name,
+ const std::string& ice_ufrag,
+ const std::string& ice_pwd) {
+ ASSERT_TRUE(desc_.GetTransportInfoByName(content_name) != NULL);
+ cricket::TransportInfo transport_info =
+ *(desc_.GetTransportInfoByName(content_name));
+ desc_.RemoveTransportInfoByName(content_name);
+ transport_info.description.ice_ufrag = ice_ufrag;
+ transport_info.description.ice_pwd = ice_pwd;
+ desc_.AddTransportInfo(transport_info);
+ }
+
void AddFingerprint() {
desc_.RemoveTransportInfoByName(kAudioContentName);
desc_.RemoveTransportInfoByName(kVideoContentName);
@@ -1050,15 +1080,22 @@
audio_desc_);
desc_.AddContent(kVideoContentName, NS_JINGLE_RTP, video_rejected,
video_desc_);
- std::string new_sdp = kSdpFullString;
+ SetIceUfragPwd(kAudioContentName,
+ audio_rejected ? "" : kCandidateUfragVoice,
+ audio_rejected ? "" : kCandidatePwdVoice);
+ SetIceUfragPwd(kVideoContentName,
+ video_rejected ? "" : kCandidateUfragVideo,
+ video_rejected ? "" : kCandidatePwdVideo);
+
+ std::string new_sdp = kSdpString;
ReplaceRejected(audio_rejected, video_rejected, &new_sdp);
- if (!jdesc_.Initialize(desc_.Copy(),
- jdesc_.session_id(),
- jdesc_.session_version())) {
+ JsepSessionDescription jdesc_no_candidates(kDummyString);
+ if (!jdesc_no_candidates.Initialize(desc_.Copy(), kSessionId,
+ kSessionVersion)) {
return false;
}
- std::string message = webrtc::SdpSerialize(jdesc_);
+ std::string message = webrtc::SdpSerialize(jdesc_no_candidates);
EXPECT_EQ(new_sdp, message);
return true;
}
@@ -1121,11 +1158,11 @@
}
bool TestDeserializeRejected(bool audio_rejected, bool video_rejected) {
- std::string new_sdp = kSdpFullString;
+ std::string new_sdp = kSdpString;
ReplaceRejected(audio_rejected, video_rejected, &new_sdp);
JsepSessionDescription new_jdesc(JsepSessionDescription::kOffer);
-
EXPECT_TRUE(SdpDeserialize(new_sdp, &new_jdesc));
+
audio_desc_ = static_cast<AudioContentDescription*>(
audio_desc_->Copy());
video_desc_ = static_cast<VideoContentDescription*>(
@@ -1136,12 +1173,18 @@
audio_desc_);
desc_.AddContent(kVideoContentName, NS_JINGLE_RTP, video_rejected,
video_desc_);
- if (!jdesc_.Initialize(desc_.Copy(),
- jdesc_.session_id(),
- jdesc_.session_version())) {
+ SetIceUfragPwd(kAudioContentName,
+ audio_rejected ? "" : kCandidateUfragVoice,
+ audio_rejected ? "" : kCandidatePwdVoice);
+ SetIceUfragPwd(kVideoContentName,
+ video_rejected ? "" : kCandidateUfragVideo,
+ video_rejected ? "" : kCandidatePwdVideo);
+ JsepSessionDescription jdesc_no_candidates(kDummyString);
+ if (!jdesc_no_candidates.Initialize(desc_.Copy(), jdesc_.session_id(),
+ jdesc_.session_version())) {
return false;
}
- EXPECT_TRUE(CompareSessionDescription(jdesc_, new_jdesc));
+ EXPECT_TRUE(CompareSessionDescription(jdesc_no_candidates, new_jdesc));
return true;
}
@@ -1540,8 +1583,8 @@
TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithoutCandidates) {
// JsepSessionDescription with desc but without candidates.
JsepSessionDescription jdesc_no_candidates(kDummyString);
- ASSERT_TRUE(jdesc_no_candidates.Initialize(desc_.Copy(),
- kSessionId, kSessionVersion));
+ ASSERT_TRUE(jdesc_no_candidates.Initialize(desc_.Copy(), kSessionId,
+ kSessionVersion));
std::string message = webrtc::SdpSerialize(jdesc_no_candidates);
EXPECT_EQ(std::string(kSdpString), message);
}
@@ -1721,6 +1764,13 @@
TEST_F(WebRtcSdpTest, SerializeCandidates) {
std::string message = webrtc::SdpSerializeCandidate(*jcandidate_);
EXPECT_EQ(std::string(kRawCandidate), message);
+
+ Candidate candidate_with_ufrag(candidates_.front());
+ candidate_with_ufrag.set_username("ABC");
+ jcandidate_.reset(new JsepIceCandidate(std::string("audio_content_name"), 0,
+ candidate_with_ufrag));
+ message = webrtc::SdpSerializeCandidate(*jcandidate_);
+ EXPECT_EQ(std::string(kRawCandidate) + " ufrag ABC", message);
}
// TODO(mallinath) : Enable this test once WebRTCSdp capable of parsing
@@ -2317,9 +2367,10 @@
EXPECT_TRUE(jcandidate.candidate().IsEquivalent(jcandidate_->candidate()));
}
-TEST_F(WebRtcSdpTest, DeserializeCandidateOldFormat) {
+TEST_F(WebRtcSdpTest, DeserializeCandidateWithUfragPwd) {
JsepIceCandidate jcandidate(kDummyMid, kDummyIndex);
- EXPECT_TRUE(SdpDeserializeCandidate(kSdpOneCandidateOldFormat,&jcandidate));
+ EXPECT_TRUE(
+ SdpDeserializeCandidate(kSdpOneCandidateWithUfragPwd, &jcandidate));
EXPECT_EQ(kDummyMid, jcandidate.sdp_mid());
EXPECT_EQ(kDummyIndex, jcandidate.sdp_mline_index());
Candidate ref_candidate = jcandidate_->candidate();
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 95abeab..d8f7637 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -30,13 +30,13 @@
#include <limits.h>
#include <algorithm>
-#include <vector>
#include <set>
+#include <utility>
+#include <vector>
#include "talk/app/webrtc/jsepicecandidate.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/mediaconstraintsinterface.h"
-#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/sctputils.h"
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
@@ -45,6 +45,7 @@
#include "talk/session/media/channel.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediasession.h"
+#include "webrtc/audio/audio_sink.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/helpers.h"
@@ -441,10 +442,11 @@
// Set |option| to the highest-priority value of |key| in the optional
// constraints if the key is found and has a valid value.
-template<typename T>
+template <typename T>
static void SetOptionFromOptionalConstraint(
const MediaConstraintsInterface* constraints,
- const std::string& key, cricket::Settable<T>* option) {
+ const std::string& key,
+ rtc::Optional<T>* option) {
if (!constraints) {
return;
}
@@ -452,7 +454,7 @@
T value;
if (constraints->GetOptional().FindFirst(key, &string_value)) {
if (rtc::FromString(string_value, &value)) {
- option->Set(value);
+ *option = rtc::Optional<T>(value);
}
}
}
@@ -492,9 +494,13 @@
}
}
+ // This method has two purposes: 1. Return whether |new_desc| requests
+ // an ICE restart (i.e., new ufrag/pwd). 2. If it requests an ICE restart
+ // and it is an OFFER, remember this in |ice_restart_| so that the next
+ // Local Answer will be created with new ufrag and pwd.
bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
const SessionDescriptionInterface* new_desc) {
- if (!old_desc || new_desc->type() != SessionDescriptionInterface::kOffer) {
+ if (!old_desc) {
return false;
}
const SessionDescription* new_sd = new_desc->description();
@@ -520,7 +526,9 @@
new_transport_desc->ice_ufrag,
new_transport_desc->ice_pwd)) {
LOG(LS_INFO) << "Remote peer request ice restart.";
- ice_restart_ = true;
+ if (new_desc->type() == SessionDescriptionInterface::kOffer) {
+ ice_restart_ = true;
+ }
return true;
}
}
@@ -593,6 +601,8 @@
const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
bundle_policy_ = rtc_configuration.bundle_policy;
rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
+ video_options_.disable_prerenderer_smoothing =
+ rtc::Optional<bool>(rtc_configuration.disable_prerenderer_smoothing);
transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version);
// Obtain a certificate from RTCConfiguration if any were provided (optional).
@@ -644,8 +654,8 @@
constraints,
MediaConstraintsInterface::kEnableDscp,
&value, NULL)) {
- audio_options_.dscp.Set(value);
- video_options_.dscp.Set(value);
+ audio_options_.dscp = rtc::Optional<bool>(value);
+ video_options_.dscp = rtc::Optional<bool>(value);
}
// Find Suspend Below Min Bitrate constraint.
@@ -654,7 +664,7 @@
MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
&value,
NULL)) {
- video_options_.suspend_below_min_bitrate.Set(value);
+ video_options_.suspend_below_min_bitrate = rtc::Optional<bool>(value);
}
SetOptionFromOptionalConstraint(constraints,
@@ -684,12 +694,10 @@
SetOptionFromOptionalConstraint(constraints,
MediaConstraintsInterface::kNumUnsignalledRecvStreams,
&video_options_.unsignalled_recv_stream_limit);
- if (video_options_.unsignalled_recv_stream_limit.IsSet()) {
- int stream_limit;
- video_options_.unsignalled_recv_stream_limit.Get(&stream_limit);
- stream_limit = std::min(kMaxUnsignalledRecvStreams, stream_limit);
- stream_limit = std::max(0, stream_limit);
- video_options_.unsignalled_recv_stream_limit.Set(stream_limit);
+ if (video_options_.unsignalled_recv_stream_limit) {
+ video_options_.unsignalled_recv_stream_limit = rtc::Optional<int>(
+ std::max(0, std::min(kMaxUnsignalledRecvStreams,
+ *video_options_.unsignalled_recv_stream_limit)));
}
SetOptionFromOptionalConstraint(constraints,
@@ -700,22 +708,12 @@
MediaConstraintsInterface::kCombinedAudioVideoBwe,
&audio_options_.combined_audio_video_bwe);
- audio_options_.audio_jitter_buffer_max_packets.Set(
- rtc_configuration.audio_jitter_buffer_max_packets);
+ audio_options_.audio_jitter_buffer_max_packets =
+ rtc::Optional<int>(rtc_configuration.audio_jitter_buffer_max_packets);
- audio_options_.audio_jitter_buffer_fast_accelerate.Set(
+ audio_options_.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(
rtc_configuration.audio_jitter_buffer_fast_accelerate);
- const cricket::VideoCodec default_codec(
- JsepSessionDescription::kDefaultVideoCodecId,
- JsepSessionDescription::kDefaultVideoCodecName,
- JsepSessionDescription::kMaxVideoCodecWidth,
- JsepSessionDescription::kMaxVideoCodecHeight,
- JsepSessionDescription::kDefaultVideoCodecFramerate,
- JsepSessionDescription::kDefaultVideoCodecPreference);
- channel_manager_->SetDefaultVideoEncoderConfig(
- cricket::VideoEncoderConfig(default_codec));
-
if (!dtls_enabled_) {
// Construct with DTLS disabled.
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
@@ -726,7 +724,7 @@
// Use the |dtls_identity_store| to generate a certificate.
RTC_DCHECK(dtls_identity_store);
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
- signaling_thread(), channel_manager_, dtls_identity_store.Pass(),
+ signaling_thread(), channel_manager_, std::move(dtls_identity_store),
this, id()));
} else {
// Use the already generated certificate.
@@ -744,12 +742,6 @@
port_allocator()->set_candidate_filter(
ConvertIceTransportTypeToCandidateFilter(rtc_configuration.type));
- if (rtc_configuration.enable_localhost_ice_candidate) {
- port_allocator()->set_flags(
- port_allocator()->flags() |
- cricket::PORTALLOCATOR_ENABLE_LOCALHOST_CANDIDATE);
- }
-
return true;
}
@@ -769,14 +761,20 @@
return webrtc_session_desc_factory_->SdesPolicy();
}
-bool WebRtcSession::GetSslRole(rtc::SSLRole* role) {
+bool WebRtcSession::GetSslRole(const std::string& transport_name,
+ rtc::SSLRole* role) {
if (!local_desc_ || !remote_desc_) {
LOG(LS_INFO) << "Local and Remote descriptions must be applied to get "
<< "SSL Role of the session.";
return false;
}
- return transport_controller_->GetSslRole(role);
+ return transport_controller_->GetSslRole(transport_name, role);
+}
+
+bool WebRtcSession::GetSslRole(const cricket::BaseChannel* channel,
+ rtc::SSLRole* role) {
+ return channel && GetSslRole(channel->transport_name(), role);
}
void WebRtcSession::CreateOffer(
@@ -978,15 +976,12 @@
return BadPranswerSdp(source, GetSessionErrorMsg(), err_desc);
}
} else if (action == kAnswer) {
- if (!PushdownTransportDescription(source, cricket::CA_ANSWER, &td_err)) {
- return BadAnswerSdp(source, MakeTdErrorString(td_err), err_desc);
- }
const cricket::ContentGroup* local_bundle =
local_desc_->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
const cricket::ContentGroup* remote_bundle =
remote_desc_->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
if (local_bundle && remote_bundle) {
- // The answerer decides the transport to bundle on
+ // The answerer decides the transport to bundle on.
const cricket::ContentGroup* answer_bundle =
(source == cricket::CS_LOCAL ? local_bundle : remote_bundle);
if (!EnableBundle(*answer_bundle)) {
@@ -994,6 +989,11 @@
return BadAnswerSdp(source, kEnableBundleFailed, err_desc);
}
}
+ // Only push down the transport description after enabling BUNDLE; we don't
+ // want to push down a description on a transport about to be destroyed.
+ if (!PushdownTransportDescription(source, cricket::CA_ANSWER, &td_err)) {
+ return BadAnswerSdp(source, MakeTdErrorString(td_err), err_desc);
+ }
EnableChannels();
SetState(STATE_INPROGRESS);
if (!PushdownMediaDescription(cricket::CA_ANSWER, source, err_desc)) {
@@ -1250,6 +1250,8 @@
const PeerConnectionInterface::RTCConfiguration& config) const {
cricket::IceConfig ice_config;
ice_config.receiving_timeout_ms = config.ice_connection_receiving_timeout;
+ ice_config.backup_connection_ping_interval =
+ config.ice_backup_candidate_pair_ping_interval;
ice_config.gather_continually = (config.continual_gathering_policy ==
PeerConnectionInterface::GATHER_CONTINUALLY);
return ice_config;
@@ -1326,6 +1328,15 @@
}
}
+void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
+ rtc::scoped_ptr<AudioSinkInterface> sink) {
+ ASSERT(signaling_thread()->IsCurrent());
+ if (!voice_channel_)
+ return;
+
+ voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
+}
+
bool WebRtcSession::SetCaptureDevice(uint32_t ssrc,
cricket::VideoCapturer* camera) {
ASSERT(signaling_thread()->IsCurrent());
@@ -1409,8 +1420,7 @@
LOG(LS_ERROR) << "InsertDtmf: Track does not exist: " << track_id;
return false;
}
- if (!voice_channel_->InsertDtmf(send_ssrc, code, duration,
- cricket::DF_SEND)) {
+ if (!voice_channel_->InsertDtmf(send_ssrc, code, duration)) {
LOG(LS_ERROR) << "Failed to insert DTMF to channel.";
return false;
}
@@ -1747,7 +1757,6 @@
cricket::GetFirstVideoContent(desc);
if ((!video_info || video_info->rejected) && video_channel_) {
SignalVideoChannelDestroyed();
- const std::string content_name = video_channel_->content_name();
channel_manager_->DestroyVideoChannel(video_channel_.release());
}
@@ -1755,7 +1764,6 @@
cricket::GetFirstAudioContent(desc);
if ((!voice_info || voice_info->rejected) && voice_channel_) {
SignalVoiceChannelDestroyed();
- const std::string content_name = voice_channel_->content_name();
channel_manager_->DestroyVoiceChannel(voice_channel_.release());
}
@@ -1763,7 +1771,6 @@
cricket::GetFirstDataContent(desc);
if ((!data_info || data_info->rejected) && data_channel_) {
SignalDataChannelDestroyed();
- const std::string content_name = data_channel_->content_name();
channel_manager_->DestroyDataChannel(data_channel_.release());
}
}
@@ -2164,9 +2171,10 @@
return;
}
- const std::string& srtp_cipher = stats.channel_stats[0].srtp_cipher;
- int ssl_cipher = stats.channel_stats[0].ssl_cipher;
- if (srtp_cipher.empty() && !ssl_cipher) {
+ int srtp_crypto_suite = stats.channel_stats[0].srtp_crypto_suite;
+ int ssl_cipher_suite = stats.channel_stats[0].ssl_cipher_suite;
+ if (srtp_crypto_suite == rtc::SRTP_INVALID_CRYPTO_SUITE &&
+ ssl_cipher_suite == rtc::TLS_NULL_WITH_NULL_NULL) {
return;
}
@@ -2186,12 +2194,13 @@
return;
}
- if (!srtp_cipher.empty()) {
- metrics_observer_->IncrementSparseEnumCounter(
- srtp_counter_type, rtc::GetSrtpCryptoSuiteFromName(srtp_cipher));
+ if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
+ metrics_observer_->IncrementSparseEnumCounter(srtp_counter_type,
+ srtp_crypto_suite);
}
- if (ssl_cipher) {
- metrics_observer_->IncrementSparseEnumCounter(ssl_counter_type, ssl_cipher);
+ if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
+ metrics_observer_->IncrementSparseEnumCounter(ssl_counter_type,
+ ssl_cipher_suite);
}
}
diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
index d9c40d1..b79e0ec 100644
--- a/talk/app/webrtc/webrtcsession.h
+++ b/talk/app/webrtc/webrtcsession.h
@@ -38,11 +38,11 @@
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/statstypes.h"
#include "talk/media/base/mediachannel.h"
-#include "webrtc/p2p/base/transportcontroller.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/sslidentity.h"
#include "webrtc/base/thread.h"
+#include "webrtc/p2p/base/transportcontroller.h"
namespace cricket {
@@ -204,7 +204,11 @@
cricket::SecurePolicy SdesPolicy() const;
// Get current ssl role from transport.
- bool GetSslRole(rtc::SSLRole* role);
+ bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role);
+
+ // Get current SSL role for this channel's transport.
+ // If |transport| is null, returns false.
+ bool GetSslRole(const cricket::BaseChannel* channel, rtc::SSLRole* role);
void CreateOffer(
CreateSessionDescriptionObserver* observer,
@@ -250,6 +254,8 @@
const cricket::AudioOptions& options,
cricket::AudioRenderer* renderer) override;
void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
+ void SetRawAudioSink(uint32_t ssrc,
+ rtc::scoped_ptr<AudioSinkInterface> sink) override;
// Implements VideoMediaProviderInterface.
bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index 3eb46f1..e81b8b5 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -25,6 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
+#include <utility>
#include <vector>
#include "talk/app/webrtc/audiotrack.h"
@@ -33,7 +34,6 @@
#include "talk/app/webrtc/jsepicecandidate.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
#include "talk/app/webrtc/peerconnection.h"
-#include "talk/app/webrtc/mediastreamsignaling.h"
#include "talk/app/webrtc/sctputils.h"
#include "talk/app/webrtc/streamcollection.h"
#include "talk/app/webrtc/streamcollection.h"
@@ -72,8 +72,6 @@
return; \
}
-using cricket::DF_PLAY;
-using cricket::DF_SEND;
using cricket::FakeVoiceMediaChannel;
using cricket::TransportInfo;
using rtc::SocketAddress;
@@ -173,15 +171,6 @@
enum RTCCertificateGenerationMethod { ALREADY_GENERATED, DTLS_IDENTITY_STORE };
-// Add some extra |newlines| to the |message| after |line|.
-static void InjectAfter(const std::string& line,
- const std::string& newlines,
- std::string* message) {
- const std::string tmp = line + newlines;
- rtc::replace_substrs(line.c_str(), line.length(), tmp.c_str(), tmp.length(),
- message);
-}
-
class MockIceObserver : public webrtc::IceObserver {
public:
MockIceObserver()
@@ -428,7 +417,7 @@
observer_.ice_gathering_state_);
EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
- dtls_identity_store.Pass(),
+ std::move(dtls_identity_store),
rtc_configuration));
session_->set_metrics_observer(metrics_observer_);
}
@@ -479,7 +468,7 @@
} else {
RTC_CHECK(false);
}
- Init(dtls_identity_store.Pass(), configuration);
+ Init(std::move(dtls_identity_store), configuration);
}
// Init with DTLS with a store that will fail to generate a certificate.
@@ -488,7 +477,7 @@
new FakeDtlsIdentityStore());
dtls_identity_store->set_should_fail(true);
PeerConnectionInterface::RTCConfiguration configuration;
- Init(dtls_identity_store.Pass(), configuration);
+ Init(std::move(dtls_identity_store), configuration);
}
void InitWithDtmfCodec() {
@@ -726,9 +715,9 @@
std::string identity_name = "WebRTC" +
rtc::ToString(rtc::CreateRandomId());
// Confirmed to work with KT_RSA and KT_ECDSA.
- tdesc_factory_->set_certificate(rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(rtc::SSLIdentity::Generate(
- identity_name, rtc::KT_DEFAULT)).Pass()));
+ tdesc_factory_->set_certificate(
+ rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+ rtc::SSLIdentity::Generate(identity_name, rtc::KT_DEFAULT))));
tdesc_factory_->set_secure(cricket::SEC_REQUIRED);
}
@@ -789,7 +778,7 @@
ASSERT_TRUE(video_channel_ != NULL);
const cricket::VideoOptions& video_options = video_channel_->options();
EXPECT_EQ(value_expected,
- video_options.unsignalled_recv_stream_limit.GetWithDefaultIfUnset(-1));
+ video_options.unsignalled_recv_stream_limit.value_or(-1));
}
void CompareIceUfragAndPassword(const cricket::SessionDescription* desc1,
@@ -1442,12 +1431,12 @@
}
void ConfigureAllocatorWithTurn() {
- cricket::RelayServerConfig relay_server(cricket::RELAY_TURN);
+ cricket::RelayServerConfig turn_server(cricket::RELAY_TURN);
cricket::RelayCredentials credentials(kTurnUsername, kTurnPassword);
- relay_server.credentials = credentials;
- relay_server.ports.push_back(cricket::ProtocolAddress(
- kTurnUdpIntAddr, cricket::PROTO_UDP, false));
- allocator_->AddRelay(relay_server);
+ turn_server.credentials = credentials;
+ turn_server.ports.push_back(
+ cricket::ProtocolAddress(kTurnUdpIntAddr, cricket::PROTO_UDP, false));
+ allocator_->AddTurnServer(turn_server);
allocator_->set_step_delay(cricket::kMinimumStepDelay);
allocator_->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP);
}
@@ -1968,6 +1957,67 @@
SetLocalDescriptionWithoutError(answer);
}
+// Test that we can create and set an answer correctly when different
+// SSL roles have been negotiated for different transports.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4525
+TEST_P(WebRtcSessionTest, TestCreateAnswerWithDifferentSslRoles) {
+ SendAudioVideoStream1();
+ InitWithDtls(GetParam());
+ SetFactoryDtlsSrtp();
+
+ SessionDescriptionInterface* offer = CreateOffer();
+ SetLocalDescriptionWithoutError(offer);
+
+ cricket::MediaSessionOptions options;
+ options.recv_video = true;
+
+ // First, negotiate different SSL roles.
+ SessionDescriptionInterface* answer =
+ CreateRemoteAnswer(offer, options, cricket::SEC_DISABLED);
+ TransportInfo* audio_transport_info =
+ answer->description()->GetTransportInfoByName("audio");
+ audio_transport_info->description.connection_role =
+ cricket::CONNECTIONROLE_ACTIVE;
+ TransportInfo* video_transport_info =
+ answer->description()->GetTransportInfoByName("video");
+ video_transport_info->description.connection_role =
+ cricket::CONNECTIONROLE_PASSIVE;
+ SetRemoteDescriptionWithoutError(answer);
+
+ // Now create an offer in the reverse direction, and ensure the initial
+ // offerer responds with an answer with correct SSL roles.
+ offer = CreateRemoteOfferWithVersion(options, cricket::SEC_DISABLED,
+ kSessionVersion,
+ session_->remote_description());
+ SetRemoteDescriptionWithoutError(offer);
+
+ answer = CreateAnswer(nullptr);
+ audio_transport_info = answer->description()->GetTransportInfoByName("audio");
+ EXPECT_EQ(cricket::CONNECTIONROLE_PASSIVE,
+ audio_transport_info->description.connection_role);
+ video_transport_info = answer->description()->GetTransportInfoByName("video");
+ EXPECT_EQ(cricket::CONNECTIONROLE_ACTIVE,
+ video_transport_info->description.connection_role);
+ SetLocalDescriptionWithoutError(answer);
+
+ // Lastly, start BUNDLE-ing on "audio", expecting that the "passive" role of
+ // audio is transferred over to video in the answer that completes the BUNDLE
+ // negotiation.
+ options.bundle_enabled = true;
+ offer = CreateRemoteOfferWithVersion(options, cricket::SEC_DISABLED,
+ kSessionVersion,
+ session_->remote_description());
+ SetRemoteDescriptionWithoutError(offer);
+ answer = CreateAnswer(nullptr);
+ audio_transport_info = answer->description()->GetTransportInfoByName("audio");
+ EXPECT_EQ(cricket::CONNECTIONROLE_PASSIVE,
+ audio_transport_info->description.connection_role);
+ video_transport_info = answer->description()->GetTransportInfoByName("video");
+ EXPECT_EQ(cricket::CONNECTIONROLE_PASSIVE,
+ video_transport_info->description.connection_role);
+ SetLocalDescriptionWithoutError(answer);
+}
+
TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
Init();
SendNothing();
@@ -2809,10 +2859,9 @@
EXPECT_FALSE(session_->SetRemoteDescription(modified_offer, &error));
}
-// Test that if the remote description indicates the peer requested ICE restart
-// (via a new ufrag or pwd), the old ICE candidates are not copied,
-// and vice versa.
-TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithIceRestart) {
+// Test that if the remote offer indicates the peer requested ICE restart (via
+// a new ufrag or pwd), the old ICE candidates are not copied, and vice versa.
+TEST_F(WebRtcSessionTest, TestSetRemoteOfferWithIceRestart) {
Init();
scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
@@ -2866,6 +2915,64 @@
EXPECT_EQ(0, session_->remote_description()->candidates(0)->count());
}
+// Test that if the remote answer indicates the peer requested ICE restart (via
+// a new ufrag or pwd), the old ICE candidates are not copied, and vice versa.
+TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithIceRestart) {
+ Init();
+ SessionDescriptionInterface* offer = CreateOffer();
+ SetLocalDescriptionWithoutError(offer);
+ scoped_ptr<SessionDescriptionInterface> answer(CreateRemoteAnswer(offer));
+
+ // Create the first answer.
+ std::string sdp;
+ ModifyIceUfragPwdLines(answer.get(), "0123456789012345",
+ "abcdefghijklmnopqrstuvwx", &sdp);
+ SessionDescriptionInterface* answer1 =
+ CreateSessionDescription(JsepSessionDescription::kPrAnswer, sdp, NULL);
+ cricket::Candidate candidate1(1, "udp", rtc::SocketAddress("1.1.1.1", 5000),
+ 0, "", "", "relay", 0, "");
+ JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0,
+ candidate1);
+ EXPECT_TRUE(answer1->AddCandidate(&ice_candidate1));
+ SetRemoteDescriptionWithoutError(answer1);
+ EXPECT_EQ(1, session_->remote_description()->candidates(0)->count());
+
+ // The second answer has the same ufrag and pwd but different address.
+ sdp.clear();
+ ModifyIceUfragPwdLines(answer.get(), "0123456789012345",
+ "abcdefghijklmnopqrstuvwx", &sdp);
+ SessionDescriptionInterface* answer2 =
+ CreateSessionDescription(JsepSessionDescription::kPrAnswer, sdp, NULL);
+ candidate1.set_address(rtc::SocketAddress("1.1.1.1", 6000));
+ JsepIceCandidate ice_candidate2(kMediaContentName0, kMediaContentIndex0,
+ candidate1);
+ EXPECT_TRUE(answer2->AddCandidate(&ice_candidate2));
+ SetRemoteDescriptionWithoutError(answer2);
+ EXPECT_EQ(2, session_->remote_description()->candidates(0)->count());
+
+ // The third answer has a different ufrag and different address.
+ sdp.clear();
+ ModifyIceUfragPwdLines(answer.get(), "0123456789012333",
+ "abcdefghijklmnopqrstuvwx", &sdp);
+ SessionDescriptionInterface* answer3 =
+ CreateSessionDescription(JsepSessionDescription::kPrAnswer, sdp, NULL);
+ candidate1.set_address(rtc::SocketAddress("1.1.1.1", 7000));
+ JsepIceCandidate ice_candidate3(kMediaContentName0, kMediaContentIndex0,
+ candidate1);
+ EXPECT_TRUE(answer3->AddCandidate(&ice_candidate3));
+ SetRemoteDescriptionWithoutError(answer3);
+ EXPECT_EQ(1, session_->remote_description()->candidates(0)->count());
+
+ // The fourth answer has no candidate but a different ufrag/pwd.
+ sdp.clear();
+ ModifyIceUfragPwdLines(answer.get(), "0123456789012444",
+ "abcdefghijklmnopqrstuvyz", &sdp);
+ SessionDescriptionInterface* offer4 =
+ CreateSessionDescription(JsepSessionDescription::kPrAnswer, sdp, NULL);
+ SetRemoteDescriptionWithoutError(offer4);
+ EXPECT_EQ(0, session_->remote_description()->candidates(0)->count());
+}
+
// Test that candidates sent to the "video" transport do not get pushed down to
// the "audio" transport channel when bundling.
TEST_F(WebRtcSessionTest, TestIgnoreCandidatesForUnusedTransportWhenBundling) {
@@ -3297,20 +3404,18 @@
EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
cricket::AudioOptions options;
- options.echo_cancellation.Set(true);
+ options.echo_cancellation = rtc::Optional<bool>(true);
rtc::scoped_ptr<FakeAudioRenderer> renderer(new FakeAudioRenderer());
session_->SetAudioSend(send_ssrc, false, options, renderer.get());
EXPECT_TRUE(channel->IsStreamMuted(send_ssrc));
- EXPECT_FALSE(channel->options().echo_cancellation.IsSet());
+ EXPECT_EQ(rtc::Optional<bool>(), channel->options().echo_cancellation);
EXPECT_TRUE(renderer->sink() != NULL);
// This will trigger SetSink(NULL) to the |renderer|.
session_->SetAudioSend(send_ssrc, true, options, NULL);
EXPECT_FALSE(channel->IsStreamMuted(send_ssrc));
- bool value;
- EXPECT_TRUE(channel->options().echo_cancellation.Get(&value));
- EXPECT_TRUE(value);
+ EXPECT_EQ(rtc::Optional<bool>(true), channel->options().echo_cancellation);
EXPECT_TRUE(renderer->sink() == NULL);
}
@@ -3387,7 +3492,6 @@
EXPECT_EQ(0U, channel->dtmf_info_queue().size());
// Insert DTMF
- const int expected_flags = DF_SEND;
const int expected_duration = 90;
session_->InsertDtmf(kAudioTrack1, 0, expected_duration);
session_->InsertDtmf(kAudioTrack1, 1, expected_duration);
@@ -3397,11 +3501,11 @@
ASSERT_EQ(3U, channel->dtmf_info_queue().size());
const uint32_t send_ssrc = channel->send_streams()[0].first_ssrc();
EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[0], send_ssrc, 0,
- expected_duration, expected_flags));
+ expected_duration));
EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[1], send_ssrc, 1,
- expected_duration, expected_flags));
+ expected_duration));
EXPECT_TRUE(CompareDtmfInfo(channel->dtmf_info_queue()[2], send_ssrc, 2,
- expected_duration, expected_flags));
+ expected_duration));
}
// This test verifies the |initial_offerer| flag when session initiates the
@@ -3582,7 +3686,9 @@
SetLocalDescriptionWithoutError(answer.release());
// Receive an offer with new ufrag and password.
- options.transport_options.ice_restart = true;
+ options.audio_transport_options.ice_restart = true;
+ options.video_transport_options.ice_restart = true;
+ options.data_transport_options.ice_restart = true;
rtc::scoped_ptr<JsepSessionDescription> updated_offer1(
CreateRemoteOffer(options, session_->remote_description()));
SetRemoteDescriptionWithoutError(updated_offer1.release());
@@ -3613,7 +3719,9 @@
SetLocalDescriptionWithoutError(answer.release());
// Receive an offer without changed ufrag or password.
- options.transport_options.ice_restart = false;
+ options.audio_transport_options.ice_restart = false;
+ options.video_transport_options.ice_restart = false;
+ options.data_transport_options.ice_restart = false;
rtc::scoped_ptr<JsepSessionDescription> updated_offer2(
CreateRemoteOffer(options, session_->remote_description()));
SetRemoteDescriptionWithoutError(updated_offer2.release());
@@ -3993,10 +4101,8 @@
ASSERT_TRUE(voice_channel_ != NULL);
const cricket::AudioOptions& audio_options = voice_channel_->options();
const cricket::VideoOptions& video_options = video_channel_->options();
- EXPECT_TRUE(audio_options.dscp.IsSet());
- EXPECT_TRUE(audio_options.dscp.GetWithDefaultIfUnset(false));
- EXPECT_TRUE(video_options.dscp.IsSet());
- EXPECT_TRUE(video_options.dscp.GetWithDefaultIfUnset(false));
+ EXPECT_EQ(rtc::Optional<bool>(true), audio_options.dscp);
+ EXPECT_EQ(rtc::Optional<bool>(true), video_options.dscp);
}
TEST_F(WebRtcSessionTest, TestSuspendBelowMinBitrateConstraint) {
@@ -4014,8 +4120,7 @@
ASSERT_TRUE(video_channel_ != NULL);
const cricket::VideoOptions& video_options = video_channel_->options();
- EXPECT_TRUE(
- video_options.suspend_below_min_bitrate.GetWithDefaultIfUnset(false));
+ EXPECT_EQ(rtc::Optional<bool>(true), video_options.suspend_below_min_bitrate);
}
TEST_F(WebRtcSessionTest, TestNumUnsignalledRecvStreamsConstraint) {
@@ -4042,8 +4147,7 @@
ASSERT_TRUE(voice_channel_ != NULL);
const cricket::AudioOptions& audio_options = voice_channel_->options();
- EXPECT_TRUE(
- audio_options.combined_audio_video_bwe.GetWithDefaultIfUnset(false));
+ EXPECT_EQ(rtc::Optional<bool>(true), audio_options.combined_audio_video_bwe);
}
// Tests that we can renegotiate new media content with ICE candidates in the
diff --git a/talk/app/webrtc/webrtcsessiondescriptionfactory.cc b/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
index 25965af..f08b77e 100644
--- a/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
+++ b/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
@@ -27,6 +27,8 @@
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
+#include <utility>
+
#include "talk/app/webrtc/dtlsidentitystore.h"
#include "talk/app/webrtc/jsep.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
@@ -99,12 +101,12 @@
der_private_key.length());
rtc::scoped_ptr<rtc::SSLIdentity> identity(
rtc::SSLIdentity::FromPEMStrings(pem_key, pem_cert));
- SignalCertificateReady(rtc::RTCCertificate::Create(identity.Pass()));
+ SignalCertificateReady(rtc::RTCCertificate::Create(std::move(identity)));
}
void WebRtcIdentityRequestObserver::OnSuccess(
rtc::scoped_ptr<rtc::SSLIdentity> identity) {
- SignalCertificateReady(rtc::RTCCertificate::Create(identity.Pass()));
+ SignalCertificateReady(rtc::RTCCertificate::Create(std::move(identity)));
}
// static
@@ -143,7 +145,7 @@
// to just use a random number as session id and start version from
// |kInitSessionVersion|.
session_version_(kInitSessionVersion),
- dtls_identity_store_(dtls_identity_store.Pass()),
+ dtls_identity_store_(std::move(dtls_identity_store)),
identity_request_observer_(identity_request_observer),
session_(session),
session_id_(session_id),
@@ -177,7 +179,7 @@
: WebRtcSessionDescriptionFactory(
signaling_thread,
channel_manager,
- dtls_identity_store.Pass(),
+ std::move(dtls_identity_store),
new rtc::RefCountedObject<WebRtcIdentityRequestObserver>(),
session,
session_id,
@@ -390,7 +392,9 @@
return;
}
if (session_->local_description() &&
- !request.options.transport_options.ice_restart) {
+ !request.options.audio_transport_options.ice_restart &&
+ !request.options.video_transport_options.ice_restart &&
+ !request.options.data_transport_options.ice_restart) {
// Include all local ice candidates in the SessionDescription unless
// the an ice restart has been requested.
CopyCandidatesFromSessionDescription(session_->local_description(), offer);
@@ -403,12 +407,25 @@
// According to http://tools.ietf.org/html/rfc5245#section-9.2.1.1
// an answer should also contain new ice ufrag and password if an offer has
// been received with new ufrag and password.
- request.options.transport_options.ice_restart = session_->IceRestartPending();
+ request.options.audio_transport_options.ice_restart =
+ session_->IceRestartPending();
+ request.options.video_transport_options.ice_restart =
+ session_->IceRestartPending();
+ request.options.data_transport_options.ice_restart =
+ session_->IceRestartPending();
// We should pass current ssl role to the transport description factory, if
// there is already an existing ongoing session.
rtc::SSLRole ssl_role;
- if (session_->GetSslRole(&ssl_role)) {
- request.options.transport_options.prefer_passive_role =
+ if (session_->GetSslRole(session_->voice_channel(), &ssl_role)) {
+ request.options.audio_transport_options.prefer_passive_role =
+ (rtc::SSL_SERVER == ssl_role);
+ }
+ if (session_->GetSslRole(session_->video_channel(), &ssl_role)) {
+ request.options.video_transport_options.prefer_passive_role =
+ (rtc::SSL_SERVER == ssl_role);
+ }
+ if (session_->GetSslRole(session_->data_channel(), &ssl_role)) {
+ request.options.data_transport_options.prefer_passive_role =
(rtc::SSL_SERVER == ssl_role);
}
@@ -437,7 +454,9 @@
return;
}
if (session_->local_description() &&
- !request.options.transport_options.ice_restart) {
+ !request.options.audio_transport_options.ice_restart &&
+ !request.options.video_transport_options.ice_restart &&
+ !request.options.data_transport_options.ice_restart) {
// Include all local ice candidates in the SessionDescription unless
// the remote peer has requested an ice restart.
CopyCandidatesFromSessionDescription(session_->local_description(), answer);
diff --git a/talk/build/common.gypi b/talk/build/common.gypi
index 36a96c5..061b06b 100644
--- a/talk/build/common.gypi
+++ b/talk/build/common.gypi
@@ -41,7 +41,6 @@
],
# Disable these to not build components which can be externally provided.
'build_expat%': 1,
- 'build_icu%': 1,
'build_json%': 1,
'build_libsrtp%': 1,
'build_libyuv%': 1,
@@ -61,17 +60,9 @@
'../../webrtc',
],
'defines': [
- 'EXPAT_RELATIVE_PATH',
- 'FEATURE_ENABLE_VOICEMAIL',
- 'GTEST_RELATIVE_PATH',
- 'JSONCPP_RELATIVE_PATH',
- 'LOGGING=1',
'SRTP_RELATIVE_PATH',
# Feature selection
- 'FEATURE_ENABLE_SSL',
- 'FEATURE_ENABLE_VOICEMAIL',
- 'FEATURE_ENABLE_PSTN',
'HAVE_SCTP',
'HAVE_SRTP',
'HAVE_WEBRTC_VIDEO',
@@ -80,7 +71,6 @@
'conditions': [
['OS=="linux"', {
'defines': [
- 'LINUX',
'WEBRTC_LINUX',
],
# Remove Chromium's disabling of the -Wformat warning.
@@ -112,7 +102,6 @@
}],
['OS=="mac"', {
'defines': [
- 'OSX',
'WEBRTC_MAC',
],
}],
@@ -129,7 +118,6 @@
}],
['OS=="ios"', {
'defines': [
- 'IOS',
'WEBRTC_MAC',
'WEBRTC_IOS',
],
diff --git a/talk/build/merge_ios_libs.gyp b/talk/build/merge_ios_libs.gyp
index 0c7114d..f7e4875 100644
--- a/talk/build/merge_ios_libs.gyp
+++ b/talk/build/merge_ios_libs.gyp
@@ -27,7 +27,7 @@
{
'includes': ['common.gypi',],
'conditions': [
- ['OS=="ios" or (OS=="mac" and mac_sdk>="10.8")', {
+ ['OS=="ios" or OS=="mac"', {
'targets': [
{
'target_name': 'libjingle_peerconnection_objc_no_op',
diff --git a/talk/codereview.settings b/talk/codereview.settings
index 97bee14..c441cc6 100644
--- a/talk/codereview.settings
+++ b/talk/codereview.settings
@@ -1,4 +1,5 @@
-Creating CLs from this location is not supported!
-Please create a full WebRTC checkout using 'fetch webrtc'
-or by cloning https://chromium.googlesource.com/external/webrtc
+Creating CLs from this location is not supported! Please make sure the current
+working directory is the parent directory of this directory.
+If you're working with a Chromium checkout, you'll have to create a full WebRTC
+checkout and upload a CL from that. See http://www.webrtc.org for instructions.
diff --git a/talk/libjingle.gyp b/talk/libjingle.gyp
index 81d723a..6e0f8a3 100755
--- a/talk/libjingle.gyp
+++ b/talk/libjingle.gyp
@@ -43,8 +43,8 @@
['OS=="linux" or OS=="android"', {
'targets': [
{
- 'target_name': 'libjingle_peerconnection_so',
- 'type': 'shared_library',
+ 'target_name': 'libjingle_peerconnection_jni',
+ 'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'libjingle_peerconnection',
@@ -62,11 +62,55 @@
'<(libyuv_dir)/include',
],
'conditions': [
- ['build_icu==1', {
- 'dependencies': [
- '<(DEPTH)/third_party/icu/icu.gyp:icuuc',
+ ['OS=="linux"', {
+ 'include_dirs': [
+ '<(java_home)/include',
+ '<(java_home)/include/linux',
],
}],
+ ['build_json==1', {
+ 'dependencies': [
+ '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
+ ],
+ 'export_dependent_settings': [
+ '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
+ ],
+ }],
+ ['OS=="android"', {
+ 'sources': [
+ 'app/webrtc/androidvideocapturer.cc',
+ 'app/webrtc/androidvideocapturer.h',
+ 'app/webrtc/java/jni/androidmediacodeccommon.h',
+ 'app/webrtc/java/jni/androidmediadecoder_jni.cc',
+ 'app/webrtc/java/jni/androidmediadecoder_jni.h',
+ 'app/webrtc/java/jni/androidmediaencoder_jni.cc',
+ 'app/webrtc/java/jni/androidmediaencoder_jni.h',
+ 'app/webrtc/java/jni/androidnetworkmonitor_jni.cc',
+ 'app/webrtc/java/jni/androidnetworkmonitor_jni.h',
+ 'app/webrtc/java/jni/androidvideocapturer_jni.cc',
+ 'app/webrtc/java/jni/androidvideocapturer_jni.h',
+ 'app/webrtc/java/jni/surfacetexturehelper_jni.cc',
+ 'app/webrtc/java/jni/surfacetexturehelper_jni.h',
+ ]
+ }],
+ ],
+ },
+ {
+ 'target_name': 'libjingle_peerconnection_so',
+ 'type': 'shared_library',
+ 'dependencies': [
+ 'libjingle_peerconnection',
+ 'libjingle_peerconnection_jni',
+ ],
+ 'sources': [
+ 'app/webrtc/java/jni/jni_onload.cc',
+ ],
+ 'variables': {
+ # This library uses native JNI exports; tell GYP so that the
+ # required symbols will be kept.
+ 'use_native_jni_exports': 1,
+ },
+ 'conditions': [
['OS=="linux"', {
'defines': [
'HAVE_GTK',
@@ -86,30 +130,6 @@
}],
],
}],
- ['OS=="android"', {
- 'sources': [
- 'app/webrtc/java/jni/androidvideocapturer_jni.cc',
- 'app/webrtc/java/jni/androidvideocapturer_jni.h',
- ],
- 'variables': {
- # This library uses native JNI exports; tell GYP so that the
- # required symbols will be kept.
- 'use_native_jni_exports': 1,
- },
- }],
- ['OS=="android" and build_with_chromium==0', {
- 'sources': [
- 'app/webrtc/java/jni/androidmediacodeccommon.h',
- 'app/webrtc/java/jni/androidmediadecoder_jni.cc',
- 'app/webrtc/java/jni/androidmediadecoder_jni.h',
- 'app/webrtc/java/jni/androidmediaencoder_jni.cc',
- 'app/webrtc/java/jni/androidmediaencoder_jni.h',
- 'app/webrtc/java/jni/androidnetworkmonitor_jni.cc',
- 'app/webrtc/java/jni/androidnetworkmonitor_jni.h',
- 'app/webrtc/java/jni/surfacetexturehelper_jni.cc',
- 'app/webrtc/java/jni/surfacetexturehelper_jni.h',
- ]
- }],
],
},
{
@@ -154,6 +174,8 @@
'app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java',
'app/webrtc/java/android/org/webrtc/CameraEnumerator.java',
'app/webrtc/java/android/org/webrtc/EglBase.java',
+ 'app/webrtc/java/android/org/webrtc/EglBase10.java',
+ 'app/webrtc/java/android/org/webrtc/EglBase14.java',
'app/webrtc/java/android/org/webrtc/GlRectDrawer.java',
'app/webrtc/java/android/org/webrtc/GlShader.java',
'app/webrtc/java/android/org/webrtc/GlUtil.java',
@@ -232,6 +254,12 @@
'libjingle_peerconnection_so',
],
'variables': {
+ # Designate as Chromium code and point to our lint settings to
+ # enable linting of the WebRTC code (this is the only way to make
+ # lint_action invoke the Android linter).
+ 'android_manifest_path': '<(webrtc_root)/build/android/AndroidManifest.xml',
+ 'suppressions_file': '<(webrtc_root)/build/android/suppressions.xml',
+ 'chromium_code': 1,
'java_in_dir': 'app/webrtc/java',
'webrtc_base_dir': '<(webrtc_root)/base',
'webrtc_modules_dir': '<(webrtc_root)/modules',
@@ -246,7 +274,7 @@
}, # libjingle_peerconnection_java
]
}],
- ['OS=="ios" or (OS=="mac" and target_arch!="ia32" and mac_sdk>="10.7")', {
+ ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
# The >= 10.7 above is required for ARC.
'targets': [
{
@@ -354,6 +382,9 @@
# common.gypi enables this for mac but we want this to be disabled
# like it is for ios.
'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
+ # Disabled due to failing when compiled with -Wall, see
+ # https://bugs.chromium.org/p/webrtc/issues/detail?id=5397
+ 'WARNING_CFLAGS': ['-Wno-unused-property-ivar'],
},
'conditions': [
['OS=="ios"', {
@@ -366,6 +397,9 @@
'app/webrtc/objc/public/RTCEAGLVideoView.h',
'app/webrtc/objc/public/RTCAVFoundationVideoSource.h',
],
+ 'dependencies': [
+ '<(webrtc_root)/base/base.gyp:rtc_base_objc',
+ ],
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
@@ -534,7 +568,7 @@
'include_dirs': [
# TODO(jiayl): move this into the direct_dependent_settings of
# usrsctp.gyp.
- '<(DEPTH)/third_party/usrsctp',
+ '<(DEPTH)/third_party/usrsctp/usrsctplib',
],
'dependencies': [
'<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
@@ -678,10 +712,16 @@
'include_dirs': [
'<(DEPTH)/testing/gtest/include',
],
+ 'include_dirs!': [
+ '<(DEPTH)/webrtc',
+ ],
'direct_dependent_settings': {
'include_dirs': [
'<(DEPTH)/testing/gtest/include',
],
+ 'include_dirs!': [
+ '<(DEPTH)/webrtc',
+ ],
},
'sources': [
'session/media/audiomonitor.cc',
@@ -725,7 +765,6 @@
'app/webrtc/dtmfsender.cc',
'app/webrtc/dtmfsender.h',
'app/webrtc/dtmfsenderinterface.h',
- 'app/webrtc/fakeportallocatorfactory.h',
'app/webrtc/jsep.h',
'app/webrtc/jsepicecandidate.cc',
'app/webrtc/jsepicecandidate.h',
@@ -740,10 +779,10 @@
'app/webrtc/mediastream.cc',
'app/webrtc/mediastream.h',
'app/webrtc/mediastreaminterface.h',
+ 'app/webrtc/mediastreamobserver.cc',
+ 'app/webrtc/mediastreamobserver.h',
'app/webrtc/mediastreamprovider.h',
'app/webrtc/mediastreamproxy.h',
- 'app/webrtc/mediastreamsignaling.cc',
- 'app/webrtc/mediastreamsignaling.h',
'app/webrtc/mediastreamtrack.h',
'app/webrtc/mediastreamtrackproxy.h',
'app/webrtc/notifier.h',
@@ -754,8 +793,6 @@
'app/webrtc/peerconnectionfactoryproxy.h',
'app/webrtc/peerconnectioninterface.h',
'app/webrtc/peerconnectionproxy.h',
- 'app/webrtc/portallocatorfactory.cc',
- 'app/webrtc/portallocatorfactory.h',
'app/webrtc/proxy.h',
'app/webrtc/remoteaudiosource.cc',
'app/webrtc/remoteaudiosource.h',
@@ -789,14 +826,6 @@
'app/webrtc/webrtcsessiondescriptionfactory.cc',
'app/webrtc/webrtcsessiondescriptionfactory.h',
],
- 'conditions': [
- ['OS=="android" and build_with_chromium==0', {
- 'sources': [
- 'app/webrtc/androidvideocapturer.h',
- 'app/webrtc/androidvideocapturer.cc',
- ],
- }],
- ],
}, # target libjingle_peerconnection
],
}
diff --git a/talk/libjingle_tests.gyp b/talk/libjingle_tests.gyp
index 41b38b3..1dc3649 100755
--- a/talk/libjingle_tests.gyp
+++ b/talk/libjingle_tests.gyp
@@ -91,15 +91,15 @@
'media/base/videocapturer_unittest.cc',
'media/base/videocommon_unittest.cc',
'media/base/videoengine_unittest.h',
+ 'media/base/videoframe_unittest.h',
'media/devices/dummydevicemanager_unittest.cc',
'media/devices/filevideocapturer_unittest.cc',
'media/sctp/sctpdataengine_unittest.cc',
'media/webrtc/simulcast_unittest.cc',
+ 'media/webrtc/webrtcmediaengine_unittest.cc',
'media/webrtc/webrtcvideocapturer_unittest.cc',
- 'media/base/videoframe_unittest.h',
'media/webrtc/webrtcvideoframe_unittest.cc',
'media/webrtc/webrtcvideoframefactory_unittest.cc',
-
# Disabled because some tests fail.
# TODO(ronghuawu): Reenable these tests.
# 'media/devices/devicemanager_unittest.cc',
@@ -128,6 +128,17 @@
},
},
}],
+ ['OS=="win" and clang==1', {
+ 'msvs_settings': {
+ 'VCCLCompilerTool': {
+ 'AdditionalOptions': [
+ # Disable warnings failing when compiling with Clang on Windows.
+ # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+ '-Wno-unused-function',
+ ],
+ },
+ },
+ },],
['OS=="ios"', {
'sources!': [
'media/sctp/sctpdataengine_unittest.cc',
@@ -176,7 +187,7 @@
}, # target libjingle_p2p_unittest
{
'target_name': 'libjingle_peerconnection_unittest',
- 'type': 'executable',
+ 'type': '<(gtest_target_type)',
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
@@ -207,7 +218,6 @@
# 'app/webrtc/peerconnectionproxy_unittest.cc',
'app/webrtc/remotevideocapturer_unittest.cc',
'app/webrtc/rtpsenderreceiver_unittest.cc',
- 'app/webrtc/sctputils.cc',
'app/webrtc/statscollector_unittest.cc',
'app/webrtc/test/fakeaudiocapturemodule.cc',
'app/webrtc/test/fakeaudiocapturemodule.h',
@@ -215,7 +225,6 @@
'app/webrtc/test/fakeconstraints.h',
'app/webrtc/test/fakedatachannelprovider.h',
'app/webrtc/test/fakedtlsidentitystore.h',
- 'app/webrtc/test/fakemediastreamsignaling.h',
'app/webrtc/test/fakeperiodicvideocapturer.h',
'app/webrtc/test/fakevideotrackrenderer.h',
'app/webrtc/test/mockpeerconnectionobservers.h',
@@ -229,17 +238,25 @@
],
'conditions': [
['OS=="android"', {
- # We want gmock features that use tr1::tuple, but we currently
- # don't support the variadic templates used by libstdc++'s
- # implementation. gmock supports this scenario by providing its
- # own implementation but we must opt in to it.
- 'defines': [
- 'GTEST_USE_OWN_TR1_TUPLE=1',
- # GTEST_USE_OWN_TR1_TUPLE only works if GTEST_HAS_TR1_TUPLE is set.
- # gmock r625 made it so that GTEST_HAS_TR1_TUPLE is set to 0
- # automatically on android, so it has to be set explicitly here.
- 'GTEST_HAS_TR1_TUPLE=1',
- ],
+ 'sources': [
+ 'app/webrtc/test/androidtestinitializer.cc',
+ 'app/webrtc/test/androidtestinitializer.h',
+ ],
+ 'dependencies': [
+ '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
+ 'libjingle.gyp:libjingle_peerconnection_jni',
+ ],
+ }],
+ ['OS=="win" and clang==1', {
+ 'msvs_settings': {
+ 'VCCLCompilerTool': {
+ 'AdditionalOptions': [
+ # Disable warnings failing when compiling with Clang on Windows.
+ # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+ '-Wno-unused-function',
+ ],
+ },
+ },
}],
],
}, # target libjingle_peerconnection_unittest
@@ -333,7 +350,7 @@
},
], # targets
}], # OS=="android"
- ['OS=="ios" or (OS=="mac" and target_arch!="ia32" and mac_sdk>="10.7")', {
+ ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
# The >=10.7 above is required to make ARC link cleanly (e.g. as
# opposed to _compile_ cleanly, which the library under test
# does just fine on 10.6 too).
@@ -378,7 +395,7 @@
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'<(DEPTH)/third_party/ocmock/ocmock.gyp:ocmock',
- '<(webrtc_root)/libjingle_examples.gyp:apprtc_signaling',
+ '<(webrtc_root)/webrtc_examples.gyp:apprtc_signaling',
],
'sources': [
'app/webrtc/objctests/mac/main.mm',
@@ -394,6 +411,17 @@
}, # target apprtc_signaling_gunit_test
],
}],
+ ['OS=="android"', {
+ 'targets': [
+ {
+ 'target_name': 'libjingle_peerconnection_unittest_apk_target',
+ 'type': 'none',
+ 'dependencies': [
+ '<(DEPTH)/webrtc/build/apk_tests.gyp:libjingle_peerconnection_unittest_apk',
+ ],
+ },
+ ],
+ }],
['test_isolation_mode != "noop"', {
'targets': [
{
diff --git a/talk/media/base/audiorenderer.h b/talk/media/base/audiorenderer.h
index 229c36e..a42cd7d 100644
--- a/talk/media/base/audiorenderer.h
+++ b/talk/media/base/audiorenderer.h
@@ -41,7 +41,7 @@
virtual void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) = 0;
// Called when the AudioRenderer is going away.
diff --git a/talk/media/base/capturemanager_unittest.cc b/talk/media/base/capturemanager_unittest.cc
index e990342..84086ab 100644
--- a/talk/media/base/capturemanager_unittest.cc
+++ b/talk/media/base/capturemanager_unittest.cc
@@ -29,6 +29,7 @@
#include "talk/media/base/fakevideocapturer.h"
#include "talk/media/base/fakevideorenderer.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/sigslot.h"
@@ -57,7 +58,7 @@
}
void PopulateSupportedFormats() {
std::vector<cricket::VideoFormat> formats;
- for (int i = 0; i < ARRAY_SIZE(kCameraFormats); ++i) {
+ for (int i = 0; i < arraysize(kCameraFormats); ++i) {
formats.push_back(cricket::VideoFormat(kCameraFormats[i]));
}
video_capturer_.ResetSupportedFormats(formats);
diff --git a/talk/media/base/codec.cc b/talk/media/base/codec.cc
index 5b747d1..59708b3 100644
--- a/talk/media/base/codec.cc
+++ b/talk/media/base/codec.cc
@@ -163,13 +163,15 @@
feedback_params.Intersect(other.feedback_params);
}
-AudioCodec::AudioCodec(int pt,
- const std::string& nm,
- int cr,
- int br,
- int cs,
- int pr)
- : Codec(pt, nm, cr, pr), bitrate(br), channels(cs) {
+AudioCodec::AudioCodec(int id,
+ const std::string& name,
+ int clockrate,
+ int bitrate,
+ size_t channels,
+ int preference)
+ : Codec(id, name, clockrate, preference),
+ bitrate(bitrate),
+ channels(channels) {
}
AudioCodec::AudioCodec() : Codec(), bitrate(0), channels(0) {
@@ -219,20 +221,20 @@
return os.str();
}
-VideoCodec::VideoCodec(int pt,
- const std::string& nm,
- int w,
- int h,
- int fr,
- int pr)
- : Codec(pt, nm, kVideoCodecClockrate, pr),
- width(w),
- height(h),
- framerate(fr) {
+VideoCodec::VideoCodec(int id,
+ const std::string& name,
+ int width,
+ int height,
+ int framerate,
+ int preference)
+ : Codec(id, name, kVideoCodecClockrate, preference),
+ width(width),
+ height(height),
+ framerate(framerate) {
}
-VideoCodec::VideoCodec(int pt, const std::string& nm)
- : Codec(pt, nm, kVideoCodecClockrate, 0),
+VideoCodec::VideoCodec(int id, const std::string& name)
+ : Codec(id, name, kVideoCodecClockrate, 0),
width(0),
height(0),
framerate(0) {
@@ -334,6 +336,11 @@
FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
}
+bool HasTransportCc(const VideoCodec& codec) {
+ return codec.HasFeedbackParam(
+ FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
+}
+
bool CodecNamesEq(const std::string& name1, const std::string& name2) {
return _stricmp(name1.c_str(), name2.c_str()) == 0;
}
diff --git a/talk/media/base/codec.h b/talk/media/base/codec.h
index 3bb08e7..da78e1c 100644
--- a/talk/media/base/codec.h
+++ b/talk/media/base/codec.h
@@ -128,10 +128,15 @@
struct AudioCodec : public Codec {
int bitrate;
- int channels;
+ size_t channels;
// Creates a codec with the given parameters.
- AudioCodec(int pt, const std::string& nm, int cr, int br, int cs, int pr);
+ AudioCodec(int id,
+ const std::string& name,
+ int clockrate,
+ int bitrate,
+ size_t channels,
+ int preference);
// Creates an empty codec.
AudioCodec();
AudioCodec(const AudioCodec& c);
@@ -161,8 +166,13 @@
int framerate;
// Creates a codec with the given parameters.
- VideoCodec(int pt, const std::string& nm, int w, int h, int fr, int pr);
- VideoCodec(int pt, const std::string& nm);
+ VideoCodec(int id,
+ const std::string& name,
+ int width,
+ int height,
+ int framerate,
+ int preference);
+ VideoCodec(int id, const std::string& name);
// Creates an empty codec.
VideoCodec();
VideoCodec(const VideoCodec& c);
@@ -209,50 +219,6 @@
std::string ToString() const;
};
-struct VideoEncoderConfig {
- static const int kDefaultMaxThreads = -1;
- static const int kDefaultCpuProfile = -1;
-
- VideoEncoderConfig()
- : max_codec(),
- num_threads(kDefaultMaxThreads),
- cpu_profile(kDefaultCpuProfile) {
- }
-
- VideoEncoderConfig(const VideoCodec& c)
- : max_codec(c),
- num_threads(kDefaultMaxThreads),
- cpu_profile(kDefaultCpuProfile) {
- }
-
- VideoEncoderConfig(const VideoCodec& c, int t, int p)
- : max_codec(c),
- num_threads(t),
- cpu_profile(p) {
- }
-
- VideoEncoderConfig& operator=(const VideoEncoderConfig& config) {
- max_codec = config.max_codec;
- num_threads = config.num_threads;
- cpu_profile = config.cpu_profile;
- return *this;
- }
-
- bool operator==(const VideoEncoderConfig& config) const {
- return max_codec == config.max_codec &&
- num_threads == config.num_threads &&
- cpu_profile == config.cpu_profile;
- }
-
- bool operator!=(const VideoEncoderConfig& config) const {
- return !(*this == config);
- }
-
- VideoCodec max_codec;
- int num_threads;
- int cpu_profile;
-};
-
// Get the codec setting associated with |payload_type|. If there
// is no codec associated with that payload type it returns false.
template <class Codec>
@@ -271,6 +237,7 @@
bool CodecNamesEq(const std::string& name1, const std::string& name2);
bool HasNack(const VideoCodec& codec);
bool HasRemb(const VideoCodec& codec);
+bool HasTransportCc(const VideoCodec& codec);
} // namespace cricket
diff --git a/talk/media/base/codec_unittest.cc b/talk/media/base/codec_unittest.cc
index 7bd3735..b2aff50 100644
--- a/talk/media/base/codec_unittest.cc
+++ b/talk/media/base/codec_unittest.cc
@@ -33,7 +33,6 @@
using cricket::DataCodec;
using cricket::FeedbackParam;
using cricket::VideoCodec;
-using cricket::VideoEncoderConfig;
using cricket::kCodecParamAssociatedPayloadType;
using cricket::kCodecParamMaxBitrate;
using cricket::kCodecParamMinBitrate;
@@ -214,54 +213,6 @@
EXPECT_FALSE(c1.Matches(VideoCodec(95, "V", 640, 400, 15, 0)));
}
-TEST_F(CodecTest, TestVideoEncoderConfigOperators) {
- VideoEncoderConfig c1(VideoCodec(
- 96, "SVC", 320, 200, 30, 3), 1, 2);
- VideoEncoderConfig c2(VideoCodec(
- 95, "SVC", 320, 200, 30, 3), 1, 2);
- VideoEncoderConfig c3(VideoCodec(
- 96, "xxx", 320, 200, 30, 3), 1, 2);
- VideoEncoderConfig c4(VideoCodec(
- 96, "SVC", 120, 200, 30, 3), 1, 2);
- VideoEncoderConfig c5(VideoCodec(
- 96, "SVC", 320, 100, 30, 3), 1, 2);
- VideoEncoderConfig c6(VideoCodec(
- 96, "SVC", 320, 200, 10, 3), 1, 2);
- VideoEncoderConfig c7(VideoCodec(
- 96, "SVC", 320, 200, 30, 1), 1, 2);
- VideoEncoderConfig c8(VideoCodec(
- 96, "SVC", 320, 200, 30, 3), 0, 2);
- VideoEncoderConfig c9(VideoCodec(
- 96, "SVC", 320, 200, 30, 3), 1, 1);
- EXPECT_TRUE(c1 != c2);
- EXPECT_TRUE(c1 != c2);
- EXPECT_TRUE(c1 != c3);
- EXPECT_TRUE(c1 != c4);
- EXPECT_TRUE(c1 != c5);
- EXPECT_TRUE(c1 != c6);
- EXPECT_TRUE(c1 != c7);
- EXPECT_TRUE(c1 != c8);
- EXPECT_TRUE(c1 != c9);
-
- VideoEncoderConfig c10;
- VideoEncoderConfig c11(VideoCodec(
- 0, "", 0, 0, 0, 0));
- VideoEncoderConfig c12(VideoCodec(
- 0, "", 0, 0, 0, 0),
- VideoEncoderConfig::kDefaultMaxThreads,
- VideoEncoderConfig::kDefaultCpuProfile);
- VideoEncoderConfig c13 = c1;
- VideoEncoderConfig c14(VideoCodec(
- 0, "", 0, 0, 0, 0), 0, 0);
-
- EXPECT_TRUE(c11 == c10);
- EXPECT_TRUE(c12 == c10);
- EXPECT_TRUE(c13 != c10);
- EXPECT_TRUE(c13 == c1);
- EXPECT_TRUE(c14 != c11);
- EXPECT_TRUE(c14 != c12);
-}
-
TEST_F(CodecTest, TestDataCodecMatches) {
// Test a codec with a static payload type.
DataCodec c0(95, "D", 0);
diff --git a/talk/media/base/constants.cc b/talk/media/base/constants.cc
index 4063004..2361be6 100644
--- a/talk/media/base/constants.cc
+++ b/talk/media/base/constants.cc
@@ -90,6 +90,7 @@
const char kRtcpFbParamNack[] = "nack";
const char kRtcpFbNackParamPli[] = "pli";
const char kRtcpFbParamRemb[] = "goog-remb";
+const char kRtcpFbParamTransportCc[] = "transport-cc";
const char kRtcpFbParamCcm[] = "ccm";
const char kRtcpFbCcmParamFir[] = "fir";
diff --git a/talk/media/base/constants.h b/talk/media/base/constants.h
index b6a9e56..706a7bd 100644
--- a/talk/media/base/constants.h
+++ b/talk/media/base/constants.h
@@ -107,6 +107,9 @@
// rtcp-fb messages according to
// http://tools.ietf.org/html/draft-alvestrand-rmcat-remb-00
extern const char kRtcpFbParamRemb[];
+// rtcp-fb messages according to
+// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
+extern const char kRtcpFbParamTransportCc[];
// ccm submessages according to RFC 5104
extern const char kRtcpFbParamCcm[];
extern const char kRtcpFbCcmParamFir[];
diff --git a/talk/media/base/cryptoparams.h b/talk/media/base/cryptoparams.h
index 9dd1db5..589953d 100644
--- a/talk/media/base/cryptoparams.h
+++ b/talk/media/base/cryptoparams.h
@@ -35,8 +35,10 @@
// Parameters for SRTP negotiation, as described in RFC 4568.
struct CryptoParams {
CryptoParams() : tag(0) {}
- CryptoParams(int t, const std::string& cs,
- const std::string& kp, const std::string& sp)
+ CryptoParams(int t,
+ const std::string& cs,
+ const std::string& kp,
+ const std::string& sp)
: tag(t), cipher_suite(cs), key_params(kp), session_params(sp) {}
bool Matches(const CryptoParams& params) const {
diff --git a/talk/media/base/executablehelpers.h b/talk/media/base/executablehelpers.h
index 401890f..dd165c2 100644
--- a/talk/media/base/executablehelpers.h
+++ b/talk/media/base/executablehelpers.h
@@ -28,7 +28,7 @@
#ifndef TALK_MEDIA_BASE_EXECUTABLEHELPERS_H_
#define TALK_MEDIA_BASE_EXECUTABLEHELPERS_H_
-#ifdef OSX
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#include <mach-o/dyld.h>
#endif
@@ -62,15 +62,15 @@
#else // UNICODE
rtc::Pathname path(exe_path_buffer);
#endif // UNICODE
-#elif defined(OSX) || defined(LINUX)
+#elif (defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)) || defined(WEBRTC_LINUX)
char exe_path_buffer[kMaxExePathSize];
-#ifdef OSX
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
uint32_t copied_length = kMaxExePathSize - 1;
if (_NSGetExecutablePath(exe_path_buffer, &copied_length) == -1) {
LOG(LS_ERROR) << "Buffer too small";
return rtc::Pathname();
}
-#elif defined LINUX
+#elif defined WEBRTC_LINUX
int32_t copied_length = kMaxExePathSize - 1;
const char* kProcExeFmt = "/proc/%d/exe";
char proc_exe_link[40];
@@ -86,11 +86,11 @@
return rtc::Pathname();
}
exe_path_buffer[copied_length] = '\0';
-#endif // LINUX
+#endif // WEBRTC_LINUX
rtc::Pathname path(exe_path_buffer);
-#else // Android || IOS
+#else // Android || iOS
rtc::Pathname path;
-#endif // OSX || LINUX
+#endif // Mac || Linux
return path;
}
diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index a6fa960..149704f 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -38,9 +38,10 @@
#include "talk/media/base/mediaengine.h"
#include "talk/media/base/rtputils.h"
#include "talk/media/base/streamparams.h"
-#include "webrtc/p2p/base/sessiondescription.h"
+#include "webrtc/audio/audio_sink.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/stringutils.h"
+#include "webrtc/p2p/base/sessiondescription.h"
namespace cricket {
@@ -229,15 +230,13 @@
class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
public:
struct DtmfInfo {
- DtmfInfo(uint32_t ssrc, int event_code, int duration, int flags)
+ DtmfInfo(uint32_t ssrc, int event_code, int duration)
: ssrc(ssrc),
event_code(event_code),
- duration(duration),
- flags(flags) {}
+ duration(duration) {}
uint32_t ssrc;
int event_code;
int duration;
- int flags;
};
explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
const AudioOptions& options)
@@ -321,9 +320,8 @@
}
virtual bool InsertDtmf(uint32_t ssrc,
int event_code,
- int duration,
- int flags) {
- dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration, flags));
+ int duration) {
+ dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration));
return true;
}
@@ -349,6 +347,12 @@
virtual bool GetStats(VoiceMediaInfo* info) { return false; }
+ virtual void SetRawAudioSink(
+ uint32_t ssrc,
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+ sink_ = std::move(sink);
+ }
+
private:
class VoiceChannelAudioSink : public AudioRenderer::Sink {
public:
@@ -364,7 +368,7 @@
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) override {}
void OnClose() override { renderer_ = NULL; }
AudioRenderer* renderer() const { return renderer_; }
@@ -421,16 +425,16 @@
int time_since_last_typing_;
AudioOptions options_;
std::map<uint32_t, VoiceChannelAudioSink*> local_renderers_;
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;
};
// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
uint32_t ssrc,
int event_code,
- int duration,
- int flags) {
+ int duration) {
return (info.duration == duration && info.event_code == event_code &&
- info.flags == flags && info.ssrc == ssrc);
+ info.ssrc == ssrc);
}
class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
@@ -694,33 +698,23 @@
class FakeBaseEngine {
public:
FakeBaseEngine()
- : loglevel_(-1),
- options_changed_(false),
+ : options_changed_(false),
fail_create_channel_(false) {}
- void SetLogging(int level, const char* filter) {
- loglevel_ = level;
- logfilter_ = filter;
- }
-
void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
- const std::vector<RtpHeaderExtension>& rtp_header_extensions() const {
- return rtp_header_extensions_;
- }
+ RtpCapabilities GetCapabilities() const { return capabilities_; }
void set_rtp_header_extensions(
const std::vector<RtpHeaderExtension>& extensions) {
- rtp_header_extensions_ = extensions;
+ capabilities_.header_extensions = extensions;
}
protected:
- int loglevel_;
- std::string logfilter_;
// Flag used by optionsmessagehandler_unittest for checking whether any
// relevant setting has been updated.
// TODO(thaloun): Replace with explicit checks of before & after values.
bool options_changed_;
bool fail_create_channel_;
- std::vector<RtpHeaderExtension> rtp_header_extensions_;
+ RtpCapabilities capabilities_;
};
class FakeVoiceEngine : public FakeBaseEngine {
@@ -733,14 +727,8 @@
}
bool Init(rtc::Thread* worker_thread) { return true; }
void Terminate() {}
- webrtc::VoiceEngine* GetVoE() { return nullptr; }
- AudioOptions GetOptions() const {
- return options_;
- }
- bool SetOptions(const AudioOptions& options) {
- options_ = options;
- options_changed_ = true;
- return true;
+ rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
+ return rtc::scoped_refptr<webrtc::AudioState>();
}
VoiceMediaChannel* CreateChannel(webrtc::Call* call,
@@ -763,21 +751,12 @@
const std::vector<AudioCodec>& codecs() { return codecs_; }
void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }
- bool SetDevices(const Device* in_device, const Device* out_device) {
- in_device_ = (in_device) ? in_device->name : "";
- out_device_ = (out_device) ? out_device->name : "";
- options_changed_ = true;
- return true;
- }
-
bool GetOutputVolume(int* level) {
*level = output_volume_;
return true;
}
-
bool SetOutputVolume(int level) {
output_volume_ = level;
- options_changed_ = true;
return true;
}
@@ -795,9 +774,6 @@
std::vector<FakeVoiceMediaChannel*> channels_;
std::vector<AudioCodec> codecs_;
int output_volume_;
- std::string in_device_;
- std::string out_device_;
- AudioOptions options_;
friend class FakeMediaEngine;
};
@@ -815,13 +791,6 @@
options_changed_ = true;
return true;
}
- bool SetDefaultEncoderConfig(const VideoEncoderConfig& config) {
- default_encoder_config_ = config;
- return true;
- }
- const VideoEncoderConfig& default_encoder_config() const {
- return default_encoder_config_;
- }
VideoMediaChannel* CreateChannel(webrtc::Call* call,
const VideoOptions& options) {
@@ -864,7 +833,6 @@
private:
std::vector<FakeVideoMediaChannel*> channels_;
std::vector<VideoCodec> codecs_;
- VideoEncoderConfig default_encoder_config_;
std::string in_device_;
bool capture_;
VideoOptions options_;
@@ -875,10 +843,7 @@
class FakeMediaEngine :
public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
public:
- FakeMediaEngine() {
- voice_ = FakeVoiceEngine();
- video_ = FakeVideoEngine();
- }
+ FakeMediaEngine() {}
virtual ~FakeMediaEngine() {}
void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
@@ -904,24 +869,13 @@
return video_.GetChannel(index);
}
- AudioOptions audio_options() const { return voice_.options_; }
int output_volume() const { return voice_.output_volume_; }
- const VideoEncoderConfig& default_video_encoder_config() const {
- return video_.default_encoder_config_;
- }
- const std::string& audio_in_device() const { return voice_.in_device_; }
- const std::string& audio_out_device() const { return voice_.out_device_; }
- int voice_loglevel() const { return voice_.loglevel_; }
- const std::string& voice_logfilter() const { return voice_.logfilter_; }
- int video_loglevel() const { return video_.loglevel_; }
- const std::string& video_logfilter() const { return video_.logfilter_; }
bool capture() const { return video_.capture_; }
bool options_changed() const {
- return voice_.options_changed_ || video_.options_changed_;
+ return video_.options_changed_;
}
void clear_options_changed() {
video_.options_changed_ = false;
- voice_.options_changed_ = false;
}
void set_fail_create_channel(bool fail) {
voice_.set_fail_create_channel(fail);
diff --git a/talk/media/base/fakemediaprocessor.h b/talk/media/base/fakemediaprocessor.h
deleted file mode 100644
index 8de2678..0000000
--- a/talk/media/base/fakemediaprocessor.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * libjingle
- * Copyright 2004 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-// TODO(solenberg): Remove this file once Chromium's libjingle.gyp/.gn are
-// updated.
diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h
index 1466084..f6fb77d 100644
--- a/talk/media/base/mediachannel.h
+++ b/talk/media/base/mediachannel.h
@@ -38,6 +38,7 @@
#include "webrtc/base/buffer.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/logging.h"
+#include "webrtc/base/optional.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/socket.h"
#include "webrtc/base/window.h"
@@ -50,88 +51,30 @@
class Timing;
}
+namespace webrtc {
+class AudioSinkInterface;
+}
+
namespace cricket {
class AudioRenderer;
-struct RtpHeader;
class ScreencastId;
-struct VideoFormat;
class VideoCapturer;
class VideoRenderer;
+struct RtpHeader;
+struct VideoFormat;
const int kMinRtpHeaderExtensionId = 1;
const int kMaxRtpHeaderExtensionId = 255;
const int kScreencastDefaultFps = 5;
-// Used in AudioOptions and VideoOptions to signify "unset" values.
template <class T>
-class Settable {
- public:
- Settable() : set_(false), val_() {}
- explicit Settable(T val) : set_(true), val_(val) {}
-
- bool IsSet() const {
- return set_;
- }
-
- bool Get(T* out) const {
- *out = val_;
- return set_;
- }
-
- T GetWithDefaultIfUnset(const T& default_value) const {
- return set_ ? val_ : default_value;
- }
-
- void Set(T val) {
- set_ = true;
- val_ = val;
- }
-
- void Clear() {
- Set(T());
- set_ = false;
- }
-
- void SetFrom(const Settable<T>& o) {
- // Set this value based on the value of o, iff o is set. If this value is
- // set and o is unset, the current value will be unchanged.
- T val;
- if (o.Get(&val)) {
- Set(val);
- }
- }
-
- std::string ToString() const {
- return set_ ? rtc::ToString(val_) : "";
- }
-
- bool operator==(const Settable<T>& o) const {
- // Equal if both are unset with any value or both set with the same value.
- return (set_ == o.set_) && (!set_ || (val_ == o.val_));
- }
-
- bool operator!=(const Settable<T>& o) const {
- return !operator==(o);
- }
-
- protected:
- void InitializeValue(const T &val) {
- val_ = val;
- }
-
- private:
- bool set_;
- T val_;
-};
-
-template <class T>
-static std::string ToStringIfSet(const char* key, const Settable<T>& val) {
+static std::string ToStringIfSet(const char* key, const rtc::Optional<T>& val) {
std::string str;
- if (val.IsSet()) {
+ if (val) {
str = key;
str += ": ";
- str += val.ToString();
+ str += val ? rtc::ToString(*val) : "";
str += ", ";
}
return str;
@@ -157,32 +100,32 @@
// but some things currently still use flags.
struct AudioOptions {
void SetAll(const AudioOptions& change) {
- echo_cancellation.SetFrom(change.echo_cancellation);
- auto_gain_control.SetFrom(change.auto_gain_control);
- noise_suppression.SetFrom(change.noise_suppression);
- highpass_filter.SetFrom(change.highpass_filter);
- stereo_swapping.SetFrom(change.stereo_swapping);
- audio_jitter_buffer_max_packets.SetFrom(
- change.audio_jitter_buffer_max_packets);
- audio_jitter_buffer_fast_accelerate.SetFrom(
- change.audio_jitter_buffer_fast_accelerate);
- typing_detection.SetFrom(change.typing_detection);
- aecm_generate_comfort_noise.SetFrom(change.aecm_generate_comfort_noise);
- conference_mode.SetFrom(change.conference_mode);
- adjust_agc_delta.SetFrom(change.adjust_agc_delta);
- experimental_agc.SetFrom(change.experimental_agc);
- extended_filter_aec.SetFrom(change.extended_filter_aec);
- delay_agnostic_aec.SetFrom(change.delay_agnostic_aec);
- experimental_ns.SetFrom(change.experimental_ns);
- aec_dump.SetFrom(change.aec_dump);
- tx_agc_target_dbov.SetFrom(change.tx_agc_target_dbov);
- tx_agc_digital_compression_gain.SetFrom(
- change.tx_agc_digital_compression_gain);
- tx_agc_limiter.SetFrom(change.tx_agc_limiter);
- recording_sample_rate.SetFrom(change.recording_sample_rate);
- playout_sample_rate.SetFrom(change.playout_sample_rate);
- dscp.SetFrom(change.dscp);
- combined_audio_video_bwe.SetFrom(change.combined_audio_video_bwe);
+ SetFrom(&echo_cancellation, change.echo_cancellation);
+ SetFrom(&auto_gain_control, change.auto_gain_control);
+ SetFrom(&noise_suppression, change.noise_suppression);
+ SetFrom(&highpass_filter, change.highpass_filter);
+ SetFrom(&stereo_swapping, change.stereo_swapping);
+ SetFrom(&audio_jitter_buffer_max_packets,
+ change.audio_jitter_buffer_max_packets);
+ SetFrom(&audio_jitter_buffer_fast_accelerate,
+ change.audio_jitter_buffer_fast_accelerate);
+ SetFrom(&typing_detection, change.typing_detection);
+ SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
+ SetFrom(&conference_mode, change.conference_mode);
+ SetFrom(&adjust_agc_delta, change.adjust_agc_delta);
+ SetFrom(&experimental_agc, change.experimental_agc);
+ SetFrom(&extended_filter_aec, change.extended_filter_aec);
+ SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
+ SetFrom(&experimental_ns, change.experimental_ns);
+ SetFrom(&aec_dump, change.aec_dump);
+ SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
+ SetFrom(&tx_agc_digital_compression_gain,
+ change.tx_agc_digital_compression_gain);
+ SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
+ SetFrom(&recording_sample_rate, change.recording_sample_rate);
+ SetFrom(&playout_sample_rate, change.playout_sample_rate);
+ SetFrom(&dscp, change.dscp);
+ SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
}
bool operator==(const AudioOptions& o) const {
@@ -247,39 +190,47 @@
// Audio processing that attempts to filter away the output signal from
// later inbound pickup.
- Settable<bool> echo_cancellation;
+ rtc::Optional<bool> echo_cancellation;
// Audio processing to adjust the sensitivity of the local mic dynamically.
- Settable<bool> auto_gain_control;
+ rtc::Optional<bool> auto_gain_control;
// Audio processing to filter out background noise.
- Settable<bool> noise_suppression;
+ rtc::Optional<bool> noise_suppression;
// Audio processing to remove background noise of lower frequencies.
- Settable<bool> highpass_filter;
+ rtc::Optional<bool> highpass_filter;
// Audio processing to swap the left and right channels.
- Settable<bool> stereo_swapping;
+ rtc::Optional<bool> stereo_swapping;
// Audio receiver jitter buffer (NetEq) max capacity in number of packets.
- Settable<int> audio_jitter_buffer_max_packets;
+ rtc::Optional<int> audio_jitter_buffer_max_packets;
// Audio receiver jitter buffer (NetEq) fast accelerate mode.
- Settable<bool> audio_jitter_buffer_fast_accelerate;
+ rtc::Optional<bool> audio_jitter_buffer_fast_accelerate;
// Audio processing to detect typing.
- Settable<bool> typing_detection;
- Settable<bool> aecm_generate_comfort_noise;
- Settable<bool> conference_mode;
- Settable<int> adjust_agc_delta;
- Settable<bool> experimental_agc;
- Settable<bool> extended_filter_aec;
- Settable<bool> delay_agnostic_aec;
- Settable<bool> experimental_ns;
- Settable<bool> aec_dump;
+ rtc::Optional<bool> typing_detection;
+ rtc::Optional<bool> aecm_generate_comfort_noise;
+ rtc::Optional<bool> conference_mode;
+ rtc::Optional<int> adjust_agc_delta;
+ rtc::Optional<bool> experimental_agc;
+ rtc::Optional<bool> extended_filter_aec;
+ rtc::Optional<bool> delay_agnostic_aec;
+ rtc::Optional<bool> experimental_ns;
+ rtc::Optional<bool> aec_dump;
// Note that tx_agc_* only applies to non-experimental AGC.
- Settable<uint16_t> tx_agc_target_dbov;
- Settable<uint16_t> tx_agc_digital_compression_gain;
- Settable<bool> tx_agc_limiter;
- Settable<uint32_t> recording_sample_rate;
- Settable<uint32_t> playout_sample_rate;
+ rtc::Optional<uint16_t> tx_agc_target_dbov;
+ rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
+ rtc::Optional<bool> tx_agc_limiter;
+ rtc::Optional<uint32_t> recording_sample_rate;
+ rtc::Optional<uint32_t> playout_sample_rate;
// Set DSCP value for packet sent from audio channel.
- Settable<bool> dscp;
+ rtc::Optional<bool> dscp;
// Enable combined audio+bandwidth BWE.
- Settable<bool> combined_audio_video_bwe;
+ rtc::Optional<bool> combined_audio_video_bwe;
+
+ private:
+ template <typename T>
+ static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
+ if (o) {
+ *s = o;
+ }
+ }
};
// Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
@@ -287,38 +238,41 @@
// We are moving all of the setting of options to structs like this,
// but some things currently still use flags.
struct VideoOptions {
- VideoOptions() {
- process_adaptation_threshhold.Set(kProcessCpuThreshold);
- system_low_adaptation_threshhold.Set(kLowSystemCpuThreshold);
- system_high_adaptation_threshhold.Set(kHighSystemCpuThreshold);
- unsignalled_recv_stream_limit.Set(kNumDefaultUnsignalledVideoRecvStreams);
- }
+ VideoOptions()
+ : process_adaptation_threshhold(kProcessCpuThreshold),
+ system_low_adaptation_threshhold(kLowSystemCpuThreshold),
+ system_high_adaptation_threshhold(kHighSystemCpuThreshold),
+ unsignalled_recv_stream_limit(kNumDefaultUnsignalledVideoRecvStreams) {}
void SetAll(const VideoOptions& change) {
- adapt_input_to_cpu_usage.SetFrom(change.adapt_input_to_cpu_usage);
- adapt_cpu_with_smoothing.SetFrom(change.adapt_cpu_with_smoothing);
- video_adapt_third.SetFrom(change.video_adapt_third);
- video_noise_reduction.SetFrom(change.video_noise_reduction);
- video_start_bitrate.SetFrom(change.video_start_bitrate);
- cpu_overuse_detection.SetFrom(change.cpu_overuse_detection);
- cpu_underuse_threshold.SetFrom(change.cpu_underuse_threshold);
- cpu_overuse_threshold.SetFrom(change.cpu_overuse_threshold);
- cpu_underuse_encode_rsd_threshold.SetFrom(
- change.cpu_underuse_encode_rsd_threshold);
- cpu_overuse_encode_rsd_threshold.SetFrom(
- change.cpu_overuse_encode_rsd_threshold);
- cpu_overuse_encode_usage.SetFrom(change.cpu_overuse_encode_usage);
- conference_mode.SetFrom(change.conference_mode);
- process_adaptation_threshhold.SetFrom(change.process_adaptation_threshhold);
- system_low_adaptation_threshhold.SetFrom(
- change.system_low_adaptation_threshhold);
- system_high_adaptation_threshhold.SetFrom(
- change.system_high_adaptation_threshhold);
- dscp.SetFrom(change.dscp);
- suspend_below_min_bitrate.SetFrom(change.suspend_below_min_bitrate);
- unsignalled_recv_stream_limit.SetFrom(change.unsignalled_recv_stream_limit);
- use_simulcast_adapter.SetFrom(change.use_simulcast_adapter);
- screencast_min_bitrate.SetFrom(change.screencast_min_bitrate);
+ SetFrom(&adapt_input_to_cpu_usage, change.adapt_input_to_cpu_usage);
+ SetFrom(&adapt_cpu_with_smoothing, change.adapt_cpu_with_smoothing);
+ SetFrom(&video_adapt_third, change.video_adapt_third);
+ SetFrom(&video_noise_reduction, change.video_noise_reduction);
+ SetFrom(&video_start_bitrate, change.video_start_bitrate);
+ SetFrom(&cpu_overuse_detection, change.cpu_overuse_detection);
+ SetFrom(&cpu_underuse_threshold, change.cpu_underuse_threshold);
+ SetFrom(&cpu_overuse_threshold, change.cpu_overuse_threshold);
+ SetFrom(&cpu_underuse_encode_rsd_threshold,
+ change.cpu_underuse_encode_rsd_threshold);
+ SetFrom(&cpu_overuse_encode_rsd_threshold,
+ change.cpu_overuse_encode_rsd_threshold);
+ SetFrom(&cpu_overuse_encode_usage, change.cpu_overuse_encode_usage);
+ SetFrom(&conference_mode, change.conference_mode);
+ SetFrom(&process_adaptation_threshhold,
+ change.process_adaptation_threshhold);
+ SetFrom(&system_low_adaptation_threshhold,
+ change.system_low_adaptation_threshhold);
+ SetFrom(&system_high_adaptation_threshhold,
+ change.system_high_adaptation_threshhold);
+ SetFrom(&dscp, change.dscp);
+ SetFrom(&suspend_below_min_bitrate, change.suspend_below_min_bitrate);
+ SetFrom(&unsignalled_recv_stream_limit,
+ change.unsignalled_recv_stream_limit);
+ SetFrom(&use_simulcast_adapter, change.use_simulcast_adapter);
+ SetFrom(&screencast_min_bitrate, change.screencast_min_bitrate);
+ SetFrom(&disable_prerenderer_smoothing,
+ change.disable_prerenderer_smoothing);
}
bool operator==(const VideoOptions& o) const {
@@ -345,7 +299,8 @@
suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
unsignalled_recv_stream_limit == o.unsignalled_recv_stream_limit &&
use_simulcast_adapter == o.use_simulcast_adapter &&
- screencast_min_bitrate == o.screencast_min_bitrate;
+ screencast_min_bitrate == o.screencast_min_bitrate &&
+ disable_prerenderer_smoothing == o.disable_prerenderer_smoothing;
}
std::string ToString() const {
@@ -381,56 +336,71 @@
}
// Enable CPU adaptation?
- Settable<bool> adapt_input_to_cpu_usage;
+ rtc::Optional<bool> adapt_input_to_cpu_usage;
// Enable CPU adaptation smoothing?
- Settable<bool> adapt_cpu_with_smoothing;
+ rtc::Optional<bool> adapt_cpu_with_smoothing;
// Enable video adapt third?
- Settable<bool> video_adapt_third;
+ rtc::Optional<bool> video_adapt_third;
// Enable denoising?
- Settable<bool> video_noise_reduction;
+ rtc::Optional<bool> video_noise_reduction;
// Experimental: Enable WebRtc higher start bitrate?
- Settable<int> video_start_bitrate;
+ rtc::Optional<int> video_start_bitrate;
// Enable WebRTC Cpu Overuse Detection, which is a new version of the CPU
// adaptation algorithm. So this option will override the
// |adapt_input_to_cpu_usage|.
- Settable<bool> cpu_overuse_detection;
+ rtc::Optional<bool> cpu_overuse_detection;
// Low threshold (t1) for cpu overuse adaptation. (Adapt up)
// Metric: encode usage (m1). m1 < t1 => underuse.
- Settable<int> cpu_underuse_threshold;
+ rtc::Optional<int> cpu_underuse_threshold;
// High threshold (t1) for cpu overuse adaptation. (Adapt down)
// Metric: encode usage (m1). m1 > t1 => overuse.
- Settable<int> cpu_overuse_threshold;
+ rtc::Optional<int> cpu_overuse_threshold;
// Low threshold (t2) for cpu overuse adaptation. (Adapt up)
// Metric: relative standard deviation of encode time (m2).
// Optional threshold. If set, (m1 < t1 && m2 < t2) => underuse.
// Note: t2 will have no effect if t1 is not set.
- Settable<int> cpu_underuse_encode_rsd_threshold;
+ rtc::Optional<int> cpu_underuse_encode_rsd_threshold;
// High threshold (t2) for cpu overuse adaptation. (Adapt down)
// Metric: relative standard deviation of encode time (m2).
// Optional threshold. If set, (m1 > t1 || m2 > t2) => overuse.
// Note: t2 will have no effect if t1 is not set.
- Settable<int> cpu_overuse_encode_rsd_threshold;
+ rtc::Optional<int> cpu_overuse_encode_rsd_threshold;
// Use encode usage for cpu detection.
- Settable<bool> cpu_overuse_encode_usage;
+ rtc::Optional<bool> cpu_overuse_encode_usage;
// Use conference mode?
- Settable<bool> conference_mode;
+ rtc::Optional<bool> conference_mode;
// Threshhold for process cpu adaptation. (Process limit)
- Settable<float> process_adaptation_threshhold;
+ rtc::Optional<float> process_adaptation_threshhold;
// Low threshhold for cpu adaptation. (Adapt up)
- Settable<float> system_low_adaptation_threshhold;
+ rtc::Optional<float> system_low_adaptation_threshhold;
// High threshhold for cpu adaptation. (Adapt down)
- Settable<float> system_high_adaptation_threshhold;
+ rtc::Optional<float> system_high_adaptation_threshhold;
// Set DSCP value for packet sent from video channel.
- Settable<bool> dscp;
+ rtc::Optional<bool> dscp;
// Enable WebRTC suspension of video. No video frames will be sent when the
// bitrate is below the configured minimum bitrate.
- Settable<bool> suspend_below_min_bitrate;
+ rtc::Optional<bool> suspend_below_min_bitrate;
// Limit on the number of early receive channels that can be created.
- Settable<int> unsignalled_recv_stream_limit;
+ rtc::Optional<int> unsignalled_recv_stream_limit;
// Enable use of simulcast adapter.
- Settable<bool> use_simulcast_adapter;
+ rtc::Optional<bool> use_simulcast_adapter;
// Force screencast to use a minimum bitrate
- Settable<int> screencast_min_bitrate;
+ rtc::Optional<int> screencast_min_bitrate;
+ // Set to true if the renderer has an algorithm of frame selection.
+ // If the value is true, then WebRTC will hand over a frame as soon as
+ // possible without delay, and rendering smoothness is completely the duty
+ // of the renderer;
+ // If the value is false, then WebRTC is responsible to delay frame release
+ // in order to increase rendering smoothness.
+ rtc::Optional<bool> disable_prerenderer_smoothing;
+
+ private:
+ template <typename T>
+ static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
+ if (o) {
+ *s = o;
+ }
+ }
};
struct RtpHeaderExtension {
@@ -447,8 +417,8 @@
std::string ToString() const {
std::ostringstream ost;
ost << "{";
- ost << "id: , " << id;
ost << "uri: " << uri;
+ ost << ", id: " << id;
ost << "}";
return ost.str();
}
@@ -481,12 +451,6 @@
OPT_AGC_MINUS_10DB = 0x80000000
};
-// DTMF flags to control if a DTMF tone should be played and/or sent.
-enum DtmfFlags {
- DF_PLAY = 0x01,
- DF_SEND = 0x02,
-};
-
class MediaChannel : public sigslot::has_slots<> {
public:
class NetworkInterface {
@@ -593,7 +557,6 @@
enum SendFlags {
SEND_NOTHING,
- SEND_RINGBACKTONE,
SEND_MICROPHONE
};
@@ -820,6 +783,7 @@
}
std::vector<SsrcGroup> ssrc_groups;
+ std::string encoder_implementation_name;
int packets_cached;
int firs_rcvd;
int plis_rcvd;
@@ -865,6 +829,7 @@
}
std::vector<SsrcGroup> ssrc_groups;
+ std::string decoder_implementation_name;
int packets_concealed;
int firs_sent;
int plis_sent;
@@ -968,9 +933,13 @@
std::vector<DataReceiverInfo> receivers;
};
+struct RtcpParameters {
+ bool reduced_size = false;
+};
+
template <class Codec>
struct RtpParameters {
- virtual std::string ToString() {
+ virtual std::string ToString() const {
std::ostringstream ost;
ost << "{";
ost << "codecs: " << VectorToString(codecs) << ", ";
@@ -982,11 +951,12 @@
std::vector<Codec> codecs;
std::vector<RtpHeaderExtension> extensions;
// TODO(pthatcher): Add streams.
+ RtcpParameters rtcp;
};
template <class Codec, class Options>
struct RtpSendParameters : RtpParameters<Codec> {
- std::string ToString() override {
+ std::string ToString() const override {
std::ostringstream ost;
ost << "{";
ost << "codecs: " << VectorToString(this->codecs) << ", ";
@@ -1056,18 +1026,18 @@
// Set speaker output volume of the specified ssrc.
virtual bool SetOutputVolume(uint32_t ssrc, double volume) = 0;
// Returns if the telephone-event has been negotiated.
- virtual bool CanInsertDtmf() { return false; }
- // Send and/or play a DTMF |event| according to the |flags|.
- // The DTMF out-of-band signal will be used on sending.
+ virtual bool CanInsertDtmf() = 0;
+ // Send a DTMF |event|. The DTMF out-of-band signal will be used.
// The |ssrc| should be either 0 or a valid send stream ssrc.
// The valid value for the |event| are 0 to 15 which corresponding to
// DTMF event 0-9, *, #, A-D.
- virtual bool InsertDtmf(uint32_t ssrc,
- int event,
- int duration,
- int flags) = 0;
+ virtual bool InsertDtmf(uint32_t ssrc, int event, int duration) = 0;
// Gets quality stats for the channel.
virtual bool GetStats(VoiceMediaInfo* info) = 0;
+
+ virtual void SetRawAudioSink(
+ uint32_t ssrc,
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) = 0;
};
struct VideoSendParameters : RtpSendParameters<VideoCodec, VideoOptions> {
@@ -1194,13 +1164,13 @@
enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK };
struct DataOptions {
- std::string ToString() {
+ std::string ToString() const {
return "{}";
}
};
struct DataSendParameters : RtpSendParameters<DataCodec, DataOptions> {
- std::string ToString() {
+ std::string ToString() const {
std::ostringstream ost;
// Options and extensions aren't used.
ost << "{";
diff --git a/talk/media/base/mediaengine.h b/talk/media/base/mediaengine.h
index 1a992d7..467614b 100644
--- a/talk/media/base/mediaengine.h
+++ b/talk/media/base/mediaengine.h
@@ -28,7 +28,7 @@
#ifndef TALK_MEDIA_BASE_MEDIAENGINE_H_
#define TALK_MEDIA_BASE_MEDIAENGINE_H_
-#ifdef OSX
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
#include <CoreAudio/CoreAudio.h>
#endif
@@ -40,8 +40,8 @@
#include "talk/media/base/mediacommon.h"
#include "talk/media/base/videocapturer.h"
#include "talk/media/base/videocommon.h"
-#include "talk/media/base/voiceprocessor.h"
#include "talk/media/devices/devicemanager.h"
+#include "webrtc/audio_state.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/sigslotrepeater.h"
@@ -51,13 +51,16 @@
namespace webrtc {
class Call;
-class VoiceEngine;
}
namespace cricket {
class VideoCapturer;
+struct RtpCapabilities {
+ std::vector<RtpHeaderExtension> header_extensions;
+};
+
// MediaEngineInterface is an abstraction of a media engine which can be
// subclassed to support different media componentry backends.
// It supports voice and video operations in the same class to facilitate
@@ -72,7 +75,7 @@
// Shuts down the engine.
virtual void Terminate() = 0;
// TODO(solenberg): Remove once VoE API refactoring is done.
- virtual webrtc::VoiceEngine* GetVoE() = 0;
+ virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const = 0;
// MediaChannel creation
// Creates a voice media channel. Returns NULL on failure.
@@ -85,20 +88,6 @@
webrtc::Call* call,
const VideoOptions& options) = 0;
- // Configuration
- // Gets global audio options.
- virtual AudioOptions GetAudioOptions() const = 0;
- // Sets global audio options. "options" are from AudioOptions, above.
- virtual bool SetAudioOptions(const AudioOptions& options) = 0;
- // Sets the default (maximum) codec/resolution and encoder option to capture
- // and encode video.
- virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config)
- = 0;
-
- // Device selection
- virtual bool SetSoundDevices(const Device* in_device,
- const Device* out_device) = 0;
-
// Device configuration
// Gets the current speaker volume, as a value between 0 and 255.
virtual bool GetOutputVolume(int* level) = 0;
@@ -109,15 +98,9 @@
virtual int GetInputLevel() = 0;
virtual const std::vector<AudioCodec>& audio_codecs() = 0;
- virtual const std::vector<RtpHeaderExtension>&
- audio_rtp_header_extensions() = 0;
+ virtual RtpCapabilities GetAudioCapabilities() = 0;
virtual const std::vector<VideoCodec>& video_codecs() = 0;
- virtual const std::vector<RtpHeaderExtension>&
- video_rtp_header_extensions() = 0;
-
- // Logging control
- virtual void SetVoiceLogging(int min_sev, const char* filter) = 0;
- virtual void SetVideoLogging(int min_sev, const char* filter) = 0;
+ virtual RtpCapabilities GetVideoCapabilities() = 0;
// Starts AEC dump using existing file.
virtual bool StartAecDump(rtc::PlatformFile file) = 0;
@@ -167,8 +150,8 @@
voice_.Terminate();
}
- virtual webrtc::VoiceEngine* GetVoE() {
- return voice_.GetVoE();
+ virtual rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
+ return voice_.GetAudioState();
}
virtual VoiceMediaChannel* CreateChannel(webrtc::Call* call,
const AudioOptions& options) {
@@ -179,21 +162,6 @@
return video_.CreateChannel(call, options);
}
- virtual AudioOptions GetAudioOptions() const {
- return voice_.GetOptions();
- }
- virtual bool SetAudioOptions(const AudioOptions& options) {
- return voice_.SetOptions(options);
- }
- virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
- return video_.SetDefaultEncoderConfig(config);
- }
-
- virtual bool SetSoundDevices(const Device* in_device,
- const Device* out_device) {
- return voice_.SetDevices(in_device, out_device);
- }
-
virtual bool GetOutputVolume(int* level) {
return voice_.GetOutputVolume(level);
}
@@ -207,21 +175,14 @@
virtual const std::vector<AudioCodec>& audio_codecs() {
return voice_.codecs();
}
- virtual const std::vector<RtpHeaderExtension>& audio_rtp_header_extensions() {
- return voice_.rtp_header_extensions();
+ virtual RtpCapabilities GetAudioCapabilities() {
+ return voice_.GetCapabilities();
}
virtual const std::vector<VideoCodec>& video_codecs() {
return video_.codecs();
}
- virtual const std::vector<RtpHeaderExtension>& video_rtp_header_extensions() {
- return video_.rtp_header_extensions();
- }
-
- virtual void SetVoiceLogging(int min_sev, const char* filter) {
- voice_.SetLogging(min_sev, filter);
- }
- virtual void SetVideoLogging(int min_sev, const char* filter) {
- video_.SetLogging(min_sev, filter);
+ virtual RtpCapabilities GetVideoCapabilities() {
+ return video_.GetCapabilities();
}
virtual bool StartAecDump(rtc::PlatformFile file) {
@@ -243,70 +204,6 @@
VIDEO video_;
};
-// NullVoiceEngine can be used with CompositeMediaEngine in the case where only
-// a video engine is desired.
-class NullVoiceEngine {
- public:
- bool Init(rtc::Thread* worker_thread) { return true; }
- void Terminate() {}
- // If you need this to return an actual channel, use FakeMediaEngine instead.
- VoiceMediaChannel* CreateChannel(const AudioOptions& options) {
- return nullptr;
- }
- AudioOptions GetOptions() const { return AudioOptions(); }
- bool SetOptions(const AudioOptions& options) { return true; }
- bool SetDevices(const Device* in_device, const Device* out_device) {
- return true;
- }
- bool GetOutputVolume(int* level) {
- *level = 0;
- return true;
- }
- bool SetOutputVolume(int level) { return true; }
- int GetInputLevel() { return 0; }
- const std::vector<AudioCodec>& codecs() { return codecs_; }
- const std::vector<RtpHeaderExtension>& rtp_header_extensions() {
- return rtp_header_extensions_;
- }
- void SetLogging(int min_sev, const char* filter) {}
- bool StartAecDump(rtc::PlatformFile file) { return false; }
- bool StartRtcEventLog(rtc::PlatformFile file) { return false; }
- void StopRtcEventLog() {}
-
- private:
- std::vector<AudioCodec> codecs_;
- std::vector<RtpHeaderExtension> rtp_header_extensions_;
-};
-
-// NullVideoEngine can be used with CompositeMediaEngine in the case where only
-// a voice engine is desired.
-class NullVideoEngine {
- public:
- bool Init(rtc::Thread* worker_thread) { return true; }
- void Terminate() {}
- // If you need this to return an actual channel, use FakeMediaEngine instead.
- VideoMediaChannel* CreateChannel(
- const VideoOptions& options,
- VoiceMediaChannel* voice_media_channel) {
- return NULL;
- }
- bool SetOptions(const VideoOptions& options) { return true; }
- bool SetDefaultEncoderConfig(const VideoEncoderConfig& config) {
- return true;
- }
- const std::vector<VideoCodec>& codecs() { return codecs_; }
- const std::vector<RtpHeaderExtension>& rtp_header_extensions() {
- return rtp_header_extensions_;
- }
- void SetLogging(int min_sev, const char* filter) {}
-
- private:
- std::vector<VideoCodec> codecs_;
- std::vector<RtpHeaderExtension> rtp_header_extensions_;
-};
-
-typedef CompositeMediaEngine<NullVoiceEngine, NullVideoEngine> NullMediaEngine;
-
enum DataChannelType {
DCT_NONE = 0,
DCT_RTP = 1,
diff --git a/talk/media/base/streamparams_unittest.cc b/talk/media/base/streamparams_unittest.cc
index a9e1ce3..a016473 100644
--- a/talk/media/base/streamparams_unittest.cc
+++ b/talk/media/base/streamparams_unittest.cc
@@ -27,6 +27,7 @@
#include "talk/media/base/streamparams.h"
#include "talk/media/base/testutils.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/gunit.h"
static const uint32_t kSsrcs1[] = {1};
@@ -54,8 +55,8 @@
cricket::SsrcGroup("abc", MAKE_VECTOR(kSsrcs2)),
};
- for (size_t i = 0; i < ARRAY_SIZE(ssrc_groups); ++i) {
- for (size_t j = 0; j < ARRAY_SIZE(ssrc_groups); ++j) {
+ for (size_t i = 0; i < arraysize(ssrc_groups); ++i) {
+ for (size_t j = 0; j < arraysize(ssrc_groups); ++j) {
EXPECT_EQ((ssrc_groups[i] == ssrc_groups[j]), (i == j));
EXPECT_EQ((ssrc_groups[i] != ssrc_groups[j]), (i != j));
}
@@ -92,7 +93,7 @@
TEST(StreamParams, HasSsrcGroup) {
cricket::StreamParams sp =
- CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, ARRAY_SIZE(kSsrcs2));
+ CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, arraysize(kSsrcs2));
EXPECT_EQ(2U, sp.ssrcs.size());
EXPECT_EQ(kSsrcs2[0], sp.first_ssrc());
EXPECT_TRUE(sp.has_ssrcs());
@@ -107,7 +108,7 @@
TEST(StreamParams, GetSsrcGroup) {
cricket::StreamParams sp =
- CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, ARRAY_SIZE(kSsrcs2));
+ CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, arraysize(kSsrcs2));
EXPECT_EQ(NULL, sp.get_ssrc_group("xyz"));
EXPECT_EQ(&sp.ssrc_groups[0], sp.get_ssrc_group("XYZ"));
}
@@ -116,17 +117,17 @@
cricket::StreamParams l1 = cricket::StreamParams::CreateLegacy(1);
cricket::StreamParams l2 = cricket::StreamParams::CreateLegacy(2);
cricket::StreamParams sg1 =
- CreateStreamParamsWithSsrcGroup("ABC", kSsrcs1, ARRAY_SIZE(kSsrcs1));
+ CreateStreamParamsWithSsrcGroup("ABC", kSsrcs1, arraysize(kSsrcs1));
cricket::StreamParams sg2 =
- CreateStreamParamsWithSsrcGroup("ABC", kSsrcs2, ARRAY_SIZE(kSsrcs2));
+ CreateStreamParamsWithSsrcGroup("ABC", kSsrcs2, arraysize(kSsrcs2));
cricket::StreamParams sg3 =
- CreateStreamParamsWithSsrcGroup("Abc", kSsrcs2, ARRAY_SIZE(kSsrcs2));
+ CreateStreamParamsWithSsrcGroup("Abc", kSsrcs2, arraysize(kSsrcs2));
cricket::StreamParams sg4 =
- CreateStreamParamsWithSsrcGroup("abc", kSsrcs2, ARRAY_SIZE(kSsrcs2));
+ CreateStreamParamsWithSsrcGroup("abc", kSsrcs2, arraysize(kSsrcs2));
cricket::StreamParams sps[] = {l1, l2, sg1, sg2, sg3, sg4};
- for (size_t i = 0; i < ARRAY_SIZE(sps); ++i) {
- for (size_t j = 0; j < ARRAY_SIZE(sps); ++j) {
+ for (size_t i = 0; i < arraysize(sps); ++i) {
+ for (size_t j = 0; j < arraysize(sps); ++j) {
EXPECT_EQ((sps[i] == sps[j]), (i == j));
EXPECT_EQ((sps[i] != sps[j]), (i != j));
}
@@ -195,7 +196,7 @@
TEST(StreamParams, ToString) {
cricket::StreamParams sp =
- CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, ARRAY_SIZE(kSsrcs2));
+ CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, arraysize(kSsrcs2));
EXPECT_STREQ("{ssrcs:[1,2];ssrc_groups:{semantics:XYZ;ssrcs:[1,2]};}",
sp.ToString().c_str());
}
diff --git a/talk/media/base/testutils.cc b/talk/media/base/testutils.cc
index 3b1fcf0..49a78e6 100644
--- a/talk/media/base/testutils.cc
+++ b/talk/media/base/testutils.cc
@@ -132,8 +132,8 @@
};
size_t RtpTestUtility::GetTestPacketCount() {
- return std::min(ARRAY_SIZE(kTestRawRtpPackets),
- ARRAY_SIZE(kTestRawRtcpPackets));
+ return std::min(arraysize(kTestRawRtpPackets),
+ arraysize(kTestRawRtcpPackets));
}
bool RtpTestUtility::WriteTestPackets(size_t count,
diff --git a/talk/media/base/testutils.h b/talk/media/base/testutils.h
index cb4146d..20c0d62 100644
--- a/talk/media/base/testutils.h
+++ b/talk/media/base/testutils.h
@@ -35,6 +35,7 @@
#include "talk/media/base/mediachannel.h"
#include "talk/media/base/videocapturer.h"
#include "talk/media/base/videocommon.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/window.h"
@@ -54,7 +55,7 @@
template <class T> inline std::vector<T> MakeVector(const T a[], size_t s) {
return std::vector<T>(a, a + s);
}
-#define MAKE_VECTOR(a) cricket::MakeVector(a, ARRAY_SIZE(a))
+#define MAKE_VECTOR(a) cricket::MakeVector(a, arraysize(a))
struct RtpDumpPacket;
class RtpDumpWriter;
diff --git a/talk/media/base/videocapturer.cc b/talk/media/base/videocapturer.cc
index ca4b906..d525a41 100644
--- a/talk/media/base/videocapturer.cc
+++ b/talk/media/base/videocapturer.cc
@@ -59,7 +59,7 @@
};
static const int64_t kMaxDistance = ~(static_cast<int64_t>(1) << 63);
-#ifdef LINUX
+#ifdef WEBRTC_LINUX
static const int kYU12Penalty = 16; // Needs to be higher than MJPG index.
#endif
static const int kDefaultScreencastFps = 5;
@@ -82,7 +82,7 @@
pixel_height(0),
time_stamp(0),
data_size(0),
- rotation(0),
+ rotation(webrtc::kVideoRotation_0),
data(NULL) {}
// TODO(fbarchard): Remove this function once lmimediaengine stops using it.
@@ -94,11 +94,6 @@
return true;
}
-webrtc::VideoRotation CapturedFrame::GetRotation() const {
- ASSERT(rotation == 0 || rotation == 90 || rotation == 180 || rotation == 270);
- return static_cast<webrtc::VideoRotation>(rotation);
-}
-
/////////////////////////////////////////////////////////////////////
// Implementation of class VideoCapturer
/////////////////////////////////////////////////////////////////////
@@ -126,7 +121,6 @@
SignalFrameCaptured.connect(this, &VideoCapturer::OnFrameCaptured);
scaled_width_ = 0;
scaled_height_ = 0;
- screencast_max_pixels_ = 0;
muted_ = false;
black_frame_count_down_ = kNumBlackFramesOnMute;
enable_video_adapter_ = true;
@@ -365,16 +359,11 @@
if (IsScreencast()) {
int scaled_width, scaled_height;
- if (screencast_max_pixels_ > 0) {
- ComputeScaleMaxPixels(captured_frame->width, captured_frame->height,
- screencast_max_pixels_, &scaled_width, &scaled_height);
- } else {
- int desired_screencast_fps = capture_format_.get() ?
- VideoFormat::IntervalToFps(capture_format_->interval) :
- kDefaultScreencastFps;
- ComputeScale(captured_frame->width, captured_frame->height,
- desired_screencast_fps, &scaled_width, &scaled_height);
- }
+ int desired_screencast_fps = capture_format_.get() ?
+ VideoFormat::IntervalToFps(capture_format_->interval) :
+ kDefaultScreencastFps;
+ ComputeScale(captured_frame->width, captured_frame->height,
+ desired_screencast_fps, &scaled_width, &scaled_height);
if (FOURCC_ARGB == captured_frame->fourcc &&
(scaled_width != captured_frame->width ||
@@ -605,7 +594,7 @@
for (size_t i = 0; i < preferred_fourccs.size(); ++i) {
if (supported_fourcc == CanonicalFourCC(preferred_fourccs[i])) {
delta_fourcc = i;
-#ifdef LINUX
+#ifdef WEBRTC_LINUX
// For HD avoid YU12 which is a software conversion and has 2 bugs
// b/7326348 b/6960899. Reenable when fixed.
if (supported.height >= 720 && (supported_fourcc == FOURCC_YU12 ||
diff --git a/talk/media/base/videocapturer.h b/talk/media/base/videocapturer.h
index 0a11ed0..a13c201 100644
--- a/talk/media/base/videocapturer.h
+++ b/talk/media/base/videocapturer.h
@@ -78,10 +78,6 @@
// fourcc. Return true if succeeded.
bool GetDataSize(uint32_t* size) const;
- // TODO(guoweis): Change the type of |rotation| from int to
- // webrtc::VideoRotation once chromium gets the code.
- webrtc::VideoRotation GetRotation() const;
-
// The width and height of the captured frame could be different from those
// of VideoFormat. Once the first frame is captured, the width, height,
// fourcc, pixel_width, and pixel_height should keep the same over frames.
@@ -90,15 +86,11 @@
uint32_t fourcc; // compression
uint32_t pixel_width; // width of a pixel, default is 1
uint32_t pixel_height; // height of a pixel, default is 1
- // TODO(magjed): |elapsed_time| is deprecated - remove once not used anymore.
- int64_t elapsed_time;
int64_t time_stamp; // timestamp of when the frame was captured, in unix
// time with nanosecond units.
uint32_t data_size; // number of bytes of the frame data
- // TODO(guoweis): This can't be converted to VideoRotation yet as it's
- // used by chrome now.
- int rotation; // rotation in degrees of the frame (0, 90, 180, 270)
+ webrtc::VideoRotation rotation; // rotation in degrees of the frame.
void* data; // pointer to the frame data. This object allocates the
// memory or points to an existing memory.
@@ -270,17 +262,6 @@
sigslot::signal2<VideoCapturer*, const VideoFrame*,
sigslot::multi_threaded_local> SignalVideoFrame;
- // If 'screencast_max_pixels' is set greater than zero, screencasts will be
- // scaled to be no larger than this value.
- // If set to zero, the max pixels will be limited to
- // Retina MacBookPro 15" resolution of 2880 x 1800.
- // For high fps, maximum pixels limit is set based on common 24" monitor
- // resolution of 2048 x 1280.
- int screencast_max_pixels() const { return screencast_max_pixels_; }
- void set_screencast_max_pixels(int p) {
- screencast_max_pixels_ = std::max(0, p);
- }
-
// If true, run video adaptation. By default, video adaptation is enabled
// and users must call video_adapter()->OnOutputFormatRequest()
// to receive frames.
@@ -377,7 +358,6 @@
bool square_pixel_aspect_ratio_; // Enable scaling to square pixels.
int scaled_width_; // Current output size from ComputeScale.
int scaled_height_;
- int screencast_max_pixels_; // Downscale screencasts further if requested.
bool muted_;
int black_frame_count_down_;
diff --git a/talk/media/base/videocapturer_unittest.cc b/talk/media/base/videocapturer_unittest.cc
index 359fe95..6d1d8aa 100644
--- a/talk/media/base/videocapturer_unittest.cc
+++ b/talk/media/base/videocapturer_unittest.cc
@@ -196,39 +196,6 @@
EXPECT_EQ(33, video_frames_received());
}
-TEST_F(VideoCapturerTest, ScreencastScaledMaxPixels) {
- capturer_.SetScreencast(true);
-
- int kWidth = 1280;
- int kHeight = 720;
-
- // Screencasts usually have large weird dimensions and are ARGB.
- std::vector<cricket::VideoFormat> formats;
- formats.push_back(cricket::VideoFormat(kWidth, kHeight,
- cricket::VideoFormat::FpsToInterval(5), cricket::FOURCC_ARGB));
- formats.push_back(cricket::VideoFormat(2 * kWidth, 2 * kHeight,
- cricket::VideoFormat::FpsToInterval(5), cricket::FOURCC_ARGB));
- capturer_.ResetSupportedFormats(formats);
-
-
- EXPECT_EQ(0, capturer_.screencast_max_pixels());
- EXPECT_EQ(cricket::CS_RUNNING, capturer_.Start(cricket::VideoFormat(
- 2 * kWidth,
- 2 * kHeight,
- cricket::VideoFormat::FpsToInterval(30),
- cricket::FOURCC_ARGB)));
- EXPECT_TRUE(capturer_.IsRunning());
- EXPECT_EQ(0, renderer_.num_rendered_frames());
- renderer_.SetSize(2 * kWidth, 2 * kHeight, 0);
- EXPECT_TRUE(capturer_.CaptureFrame());
- EXPECT_EQ(1, renderer_.num_rendered_frames());
-
- capturer_.set_screencast_max_pixels(kWidth * kHeight);
- renderer_.SetSize(kWidth, kHeight, 0);
- EXPECT_TRUE(capturer_.CaptureFrame());
- EXPECT_EQ(2, renderer_.num_rendered_frames());
-}
-
TEST_F(VideoCapturerTest, ScreencastScaledOddWidth) {
capturer_.SetScreencast(true);
diff --git a/talk/media/base/videocommon.cc b/talk/media/base/videocommon.cc
index 7b6aac2..faf6450 100644
--- a/talk/media/base/videocommon.cc
+++ b/talk/media/base/videocommon.cc
@@ -31,6 +31,7 @@
#include <math.h>
#include <sstream>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/common.h"
namespace cricket {
@@ -58,7 +59,7 @@
};
uint32_t CanonicalFourCC(uint32_t fourcc) {
- for (int i = 0; i < ARRAY_SIZE(kFourCCAliases); ++i) {
+ for (int i = 0; i < arraysize(kFourCCAliases); ++i) {
if (kFourCCAliases[i].alias == fourcc) {
return kFourCCAliases[i].canonical;
}
@@ -75,7 +76,7 @@
1.f / 16.f // 1/16 scale.
};
-static const int kNumScaleFactors = ARRAY_SIZE(kScaleFactors);
+static const int kNumScaleFactors = arraysize(kScaleFactors);
// Finds the scale factor that, when applied to width and height, produces
// fewer than num_pixels.
@@ -106,9 +107,6 @@
ASSERT(scaled_width != NULL);
ASSERT(scaled_height != NULL);
ASSERT(max_pixels > 0);
- // For VP8 the values for max width and height can be found here
- // webrtc/src/video_engine/vie_defines.h (kViEMaxCodecWidth and
- // kViEMaxCodecHeight)
const int kMaxWidth = 4096;
const int kMaxHeight = 3072;
int new_frame_width = frame_width;
diff --git a/talk/media/base/videoengine_unittest.h b/talk/media/base/videoengine_unittest.h
index d89b3e6..d7fa00d 100644
--- a/talk/media/base/videoengine_unittest.h
+++ b/talk/media/base/videoengine_unittest.h
@@ -126,327 +126,6 @@
}
};
-template<class E>
-class VideoEngineTest : public testing::Test {
- protected:
- // Tests starting and stopping the engine, and creating a channel.
- void StartupShutdown() {
- EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
- cricket::VideoMediaChannel* channel = engine_.CreateChannel(NULL);
- EXPECT_TRUE(channel != NULL);
- delete channel;
- engine_.Terminate();
- }
-
- void ConstrainNewCodecBody() {
- cricket::VideoCodec empty, in, out;
- cricket::VideoCodec max_settings(engine_.codecs()[0].id,
- engine_.codecs()[0].name,
- 1280, 800, 30, 0);
-
- // set max settings of 1280x800x30
- EXPECT_TRUE(engine_.SetDefaultEncoderConfig(
- cricket::VideoEncoderConfig(max_settings)));
-
- // don't constrain the max resolution
- in = max_settings;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // constrain resolution greater than the max and wider aspect,
- // picking best aspect (16:10)
- in.width = 1380;
- in.height = 800;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 1280, 720, 30);
-
- // constrain resolution greater than the max and narrow aspect,
- // picking best aspect (16:9)
- in.width = 1280;
- in.height = 740;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 1280, 720, 30);
-
- // constrain resolution greater than the max, picking equal aspect (4:3)
- in.width = 1280;
- in.height = 960;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 1280, 800, 30);
-
- // constrain resolution greater than the max, picking equal aspect (16:10)
- in.width = 1280;
- in.height = 800;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 1280, 800, 30);
-
- // reduce max settings to 640x480x30
- max_settings.width = 640;
- max_settings.height = 480;
- EXPECT_TRUE(engine_.SetDefaultEncoderConfig(
- cricket::VideoEncoderConfig(max_settings)));
-
- // don't constrain the max resolution
- in = max_settings;
- in.width = 640;
- in.height = 480;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // keep 16:10 if they request it
- in.height = 400;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // don't constrain lesser 4:3 resolutions
- in.width = 320;
- in.height = 240;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // don't constrain lesser 16:10 resolutions
- in.width = 320;
- in.height = 200;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // requested resolution of 0x0 succeeds
- in.width = 0;
- in.height = 0;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // constrain resolution lesser than the max and wider aspect,
- // picking best aspect (16:9)
- in.width = 350;
- in.height = 201;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 320, 180, 30);
-
- // constrain resolution greater than the max and narrow aspect,
- // picking best aspect (4:3)
- in.width = 350;
- in.height = 300;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 320, 240, 30);
-
- // constrain resolution greater than the max and wider aspect,
- // picking best aspect (16:9)
- in.width = 1380;
- in.height = 800;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 360, 30);
-
- // constrain resolution greater than the max and narrow aspect,
- // picking best aspect (4:3)
- in.width = 1280;
- in.height = 900;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 480, 30);
-
- // constrain resolution greater than the max, picking equal aspect (4:3)
- in.width = 1280;
- in.height = 960;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 480, 30);
-
- // constrain resolution greater than the max, picking equal aspect (16:10)
- in.width = 1280;
- in.height = 800;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 400, 30);
-
- // constrain res & fps greater than the max
- in.framerate = 50;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 400, 30);
-
- // reduce max settings to 160x100x10
- max_settings.width = 160;
- max_settings.height = 100;
- max_settings.framerate = 10;
- EXPECT_TRUE(engine_.SetDefaultEncoderConfig(
- cricket::VideoEncoderConfig(max_settings)));
-
- // constrain res & fps to new max
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 160, 100, 10);
-
- // allow 4:3 "comparable" resolutions
- in.width = 160;
- in.height = 120;
- in.framerate = 10;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 160, 120, 10);
- }
-
- // This is the new way of constraining codec size, where we no longer maintain
- // a list of the supported formats. Instead, CanSendCodec will just downscale
- // the resolution by 2 until the width is below clamp.
- void ConstrainNewCodec2Body() {
- cricket::VideoCodec empty, in, out;
- cricket::VideoCodec max_settings(engine_.codecs()[0].id,
- engine_.codecs()[0].name,
- 1280, 800, 30, 0);
-
- // Set max settings of 1280x800x30
- EXPECT_TRUE(engine_.SetDefaultEncoderConfig(
- cricket::VideoEncoderConfig(max_settings)));
-
- // Don't constrain the max resolution
- in = max_settings;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // Constrain resolution greater than the max width.
- in.width = 1380;
- in.height = 800;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 690, 400, 30);
-
- // Don't constrain resolution when only the height is greater than max.
- in.width = 960;
- in.height = 1280;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 960, 1280, 30);
-
- // Don't constrain smaller format.
- in.width = 640;
- in.height = 480;
- EXPECT_TRUE(engine_.CanSendCodec(in, empty, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 480, 30);
- }
-
- void ConstrainRunningCodecBody() {
- cricket::VideoCodec in, out, current;
- cricket::VideoCodec max_settings(engine_.codecs()[0].id,
- engine_.codecs()[0].name,
- 1280, 800, 30, 0);
-
- // set max settings of 1280x960x30
- EXPECT_TRUE(engine_.SetDefaultEncoderConfig(
- cricket::VideoEncoderConfig(max_settings)));
-
- // establish current call at 1280x800x30 (16:10)
- current = max_settings;
- current.height = 800;
-
- // Don't constrain current resolution
- in = current;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // requested resolution of 0x0 succeeds
- in.width = 0;
- in.height = 0;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // Reduce an intermediate resolution down to the next lowest one, preserving
- // aspect ratio.
- in.width = 800;
- in.height = 600;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 400, 30);
-
- // Clamping by aspect ratio, but still never return a dimension higher than
- // requested.
- in.width = 1280;
- in.height = 720;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 1280, 720, 30);
-
- in.width = 1279;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 960, 600, 30);
-
- in.width = 1281;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 1280, 720, 30);
-
- // Clamp large resolutions down, always preserving aspect
- in.width = 1920;
- in.height = 1080;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 1280, 800, 30);
-
- in.width = 1921;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 1280, 800, 30);
-
- in.width = 1919;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 1280, 800, 30);
-
- // reduce max settings to 640x480x30
- max_settings.width = 640;
- max_settings.height = 480;
- EXPECT_TRUE(engine_.SetDefaultEncoderConfig(
- cricket::VideoEncoderConfig(max_settings)));
-
- // establish current call at 640x400x30 (16:10)
- current = max_settings;
- current.height = 400;
-
- // Don't constrain current resolution
- in = current;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // requested resolution of 0x0 succeeds
- in.width = 0;
- in.height = 0;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED2(IsEqualCodec, out, in);
-
- // Reduce an intermediate resolution down to the next lowest one, preserving
- // aspect ratio.
- in.width = 400;
- in.height = 300;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 320, 200, 30);
-
- // Clamping by aspect ratio, but still never return a dimension higher than
- // requested.
- in.width = 640;
- in.height = 360;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 360, 30);
-
- in.width = 639;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 480, 300, 30);
-
- in.width = 641;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 360, 30);
-
- // Clamp large resolutions down, always preserving aspect
- in.width = 1280;
- in.height = 800;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 400, 30);
-
- in.width = 1281;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 400, 30);
-
- in.width = 1279;
- EXPECT_TRUE(engine_.CanSendCodec(in, current, &out));
- EXPECT_PRED4(IsEqualRes, out, 640, 400, 30);
-
- // Should fail for any that are smaller than our supported formats
- in.width = 80;
- in.height = 80;
- EXPECT_FALSE(engine_.CanSendCodec(in, current, &out));
-
- in.height = 50;
- EXPECT_FALSE(engine_.CanSendCodec(in, current, &out));
- }
-
- VideoEngineOverride<E> engine_;
- rtc::scoped_ptr<cricket::FakeVideoCapturer> video_capturer_;
-};
-
template<class E, class C>
class VideoMediaChannelTest : public testing::Test,
public sigslot::has_slots<> {
@@ -875,7 +554,7 @@
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
- parameters.options.conference_mode.Set(true);
+ parameters.options.conference_mode = rtc::Optional<bool>(true);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->AddRecvStream(
@@ -926,7 +605,7 @@
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
- parameters.options.conference_mode.Set(true);
+ parameters.options.conference_mode = rtc::Optional<bool>(true);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kSsrc)));
@@ -1009,8 +688,10 @@
rtc::scoped_ptr<const rtc::Buffer> p(GetRtpPacket(0));
ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
EXPECT_EQ(kSsrc, ssrc);
- EXPECT_EQ(NumRtpPackets(), NumRtpPackets(ssrc));
- EXPECT_EQ(NumRtpBytes(), NumRtpBytes(ssrc));
+ // Packets are being paced out, so these can mismatch between the first and
+ // second call to NumRtpPackets until pending packets are paced out.
+ EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(ssrc), kTimeout);
+ EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(ssrc), kTimeout);
EXPECT_EQ(1, NumSentSsrcs());
EXPECT_EQ(0, NumRtpPackets(kSsrc - 1));
EXPECT_EQ(0, NumRtpBytes(kSsrc - 1));
@@ -1031,8 +712,10 @@
rtc::scoped_ptr<const rtc::Buffer> p(GetRtpPacket(0));
ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
EXPECT_EQ(999u, ssrc);
- EXPECT_EQ(NumRtpPackets(), NumRtpPackets(ssrc));
- EXPECT_EQ(NumRtpBytes(), NumRtpBytes(ssrc));
+ // Packets are being paced out, so these can mismatch between the first and
+ // second call to NumRtpPackets until pending packets are paced out.
+ EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(ssrc), kTimeout);
+ EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(ssrc), kTimeout);
EXPECT_EQ(1, NumSentSsrcs());
EXPECT_EQ(0, NumRtpPackets(kSsrc));
EXPECT_EQ(0, NumRtpBytes(kSsrc));
@@ -1236,7 +919,7 @@
EXPECT_TRUE(SetDefaultCodec());
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(DefaultCodec());
- parameters.options.conference_mode.Set(true);
+ parameters.options.conference_mode = rtc::Optional<bool>(true);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->AddRecvStream(
@@ -1746,8 +1429,8 @@
// Tests that we can send and receive frames with early receive.
void TwoStreamsSendAndUnsignalledRecv(const cricket::VideoCodec& codec) {
cricket::VideoSendParameters parameters;
- parameters.options.conference_mode.Set(true);
- parameters.options.unsignalled_recv_stream_limit.Set(1);
+ parameters.options.conference_mode = rtc::Optional<bool>(true);
+ parameters.options.unsignalled_recv_stream_limit = rtc::Optional<int>(1);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
SetUpSecondStreamWithNoRecv();
// Test sending and receiving on first stream.
@@ -1780,8 +1463,8 @@
void TwoStreamsAddAndRemoveUnsignalledRecv(
const cricket::VideoCodec& codec) {
cricket::VideoOptions vmo;
- vmo.conference_mode.Set(true);
- vmo.unsignalled_recv_stream_limit.Set(1);
+ vmo.conference_mode = rtc::Optional<bool>(true);
+ vmo.unsignalled_recv_stream_limit = rtc::Optional<int>(1);
EXPECT_TRUE(channel_->SetOptions(vmo));
SetUpSecondStreamWithNoRecv();
// Sending and receiving on first stream.
diff --git a/talk/media/base/videoframe.cc b/talk/media/base/videoframe.cc
index 2b604b0..3e4d60a 100644
--- a/talk/media/base/videoframe.cc
+++ b/talk/media/base/videoframe.cc
@@ -33,6 +33,7 @@
#include "libyuv/planar_functions.h"
#include "libyuv/scale.h"
#include "talk/media/base/videocommon.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
@@ -318,7 +319,7 @@
}
// TODO(fbarchard): Make function to dump information about frames.
uint8_t four_samples[4] = {0, 0, 0, 0};
- for (size_t i = 0; i < ARRAY_SIZE(four_samples) && i < sample_size; ++i) {
+ for (size_t i = 0; i < arraysize(four_samples) && i < sample_size; ++i) {
four_samples[i] = sample[i];
}
if (sample_size < expected_size) {
diff --git a/talk/media/base/videoframe.h b/talk/media/base/videoframe.h
index 217732f..f81c678 100644
--- a/talk/media/base/videoframe.h
+++ b/talk/media/base/videoframe.h
@@ -30,7 +30,7 @@
#include "webrtc/base/basictypes.h"
#include "webrtc/base/stream.h"
-#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
#include "webrtc/common_video/rotation.h"
namespace cricket {
diff --git a/talk/media/base/videoframefactory.cc b/talk/media/base/videoframefactory.cc
index dfd97c6..fb81096 100644
--- a/talk/media/base/videoframefactory.cc
+++ b/talk/media/base/videoframefactory.cc
@@ -51,8 +51,8 @@
// If the frame is rotated, we need to switch the width and height.
if (apply_rotation_ &&
- (input_frame->GetRotation() == webrtc::kVideoRotation_90 ||
- input_frame->GetRotation() == webrtc::kVideoRotation_270)) {
+ (input_frame->rotation == webrtc::kVideoRotation_90 ||
+ input_frame->rotation == webrtc::kVideoRotation_270)) {
std::swap(output_width, output_height);
}
diff --git a/talk/media/base/videorenderer.h b/talk/media/base/videorenderer.h
index 0a0ee51..a18c4e3 100644
--- a/talk/media/base/videorenderer.h
+++ b/talk/media/base/videorenderer.h
@@ -42,11 +42,12 @@
class VideoRenderer {
public:
virtual ~VideoRenderer() {}
- // Called when the video has changed size. This is also used as an
- // initialization method to set the UI size before any video frame
- // rendered. webrtc::ExternalRenderer's FrameSizeChange will invoke this when
- // it's called or later when a VideoRenderer is attached.
- virtual bool SetSize(int width, int height, int reserved) = 0;
+ // Called when the video has changed size.
+ // TODO(nisse): This method is not really used, and should be
+ // deleted. Provide a default do-nothing implementation, to easy the
+ // transition as the method is deleted in subclasses, in particular,
+ // chrome's MockVideoRenderer class.
+ virtual bool SetSize(int width, int height, int reserved) { return true; };
// Called when a new frame is available for display.
virtual bool RenderFrame(const VideoFrame *frame) = 0;
diff --git a/talk/media/base/voiceprocessor.h b/talk/media/base/voiceprocessor.h
deleted file mode 100755
index 8de2678..0000000
--- a/talk/media/base/voiceprocessor.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * libjingle
- * Copyright 2004 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-// TODO(solenberg): Remove this file once Chromium's libjingle.gyp/.gn are
-// updated.
diff --git a/talk/media/devices/carbonvideorenderer.cc b/talk/media/devices/carbonvideorenderer.cc
index 846135d..b711ae4 100644
--- a/talk/media/devices/carbonvideorenderer.cc
+++ b/talk/media/devices/carbonvideorenderer.cc
@@ -40,7 +40,6 @@
image_height_(0),
x_(x),
y_(y),
- image_ref_(NULL),
window_ref_(NULL) {
}
diff --git a/talk/media/devices/carbonvideorenderer.h b/talk/media/devices/carbonvideorenderer.h
index 52c9740..e8329ea 100644
--- a/talk/media/devices/carbonvideorenderer.h
+++ b/talk/media/devices/carbonvideorenderer.h
@@ -65,7 +65,6 @@
int image_height_;
int x_;
int y_;
- CGImageRef image_ref_;
WindowRef window_ref_;
};
diff --git a/talk/media/devices/devicemanager.cc b/talk/media/devices/devicemanager.cc
index 1d7ac5b..eca14a5 100644
--- a/talk/media/devices/devicemanager.cc
+++ b/talk/media/devices/devicemanager.cc
@@ -123,7 +123,7 @@
bool DeviceManager::GetVideoCaptureDevices(std::vector<Device>* devices) {
devices->clear();
-#if defined(ANDROID) || defined(IOS)
+#if defined(ANDROID) || defined(WEBRTC_IOS)
// On Android and iOS, we treat the camera(s) as a single device. Even if
// there are multiple cameras, that's abstracted away at a higher level.
Device dev("camera", "1"); // name and ID
diff --git a/talk/media/devices/devicemanager_unittest.cc b/talk/media/devices/devicemanager_unittest.cc
index f259c7d..606a05e 100644
--- a/talk/media/devices/devicemanager_unittest.cc
+++ b/talk/media/devices/devicemanager_unittest.cc
@@ -39,6 +39,7 @@
#include "talk/media/base/videocapturerfactory.h"
#include "talk/media/devices/filevideocapturer.h"
#include "talk/media/devices/v4llookup.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
@@ -47,10 +48,10 @@
#include "webrtc/base/stream.h"
#include "webrtc/base/windowpickerfactory.h"
-#ifdef LINUX
+#ifdef WEBRTC_LINUX
// TODO(juberti): Figure out why this doesn't compile on Windows.
#include "webrtc/base/fileutils_mock.h"
-#endif // LINUX
+#endif // WEBRTC_LINUX
using rtc::Pathname;
using rtc::FileTimeType;
@@ -269,22 +270,22 @@
"device5",
};
std::vector<Device> devices;
- for (int i = 0; i < ARRAY_SIZE(kTotalDevicesName); ++i) {
+ for (int i = 0; i < arraysize(kTotalDevicesName); ++i) {
devices.push_back(Device(kTotalDevicesName[i], i));
}
EXPECT_TRUE(CompareDeviceList(devices, kTotalDevicesName,
- ARRAY_SIZE(kTotalDevicesName)));
+ arraysize(kTotalDevicesName)));
// Return false if given NULL as the exclusion list.
EXPECT_TRUE(DeviceManager::FilterDevices(&devices, NULL));
// The devices should not change.
EXPECT_TRUE(CompareDeviceList(devices, kTotalDevicesName,
- ARRAY_SIZE(kTotalDevicesName)));
+ arraysize(kTotalDevicesName)));
EXPECT_TRUE(DeviceManager::FilterDevices(&devices, kFilteredDevicesName));
EXPECT_TRUE(CompareDeviceList(devices, kDevicesName,
- ARRAY_SIZE(kDevicesName)));
+ arraysize(kDevicesName)));
}
-#ifdef LINUX
+#ifdef WEBRTC_LINUX
class FakeV4LLookup : public cricket::V4LLookup {
public:
explicit FakeV4LLookup(std::vector<std::string> device_paths)
@@ -376,7 +377,7 @@
EXPECT_EQ("/dev/video0", video_ins.at(0).name);
EXPECT_EQ("/dev/video5", video_ins.at(1).name);
}
-#endif // LINUX
+#endif // WEBRTC_LINUX
// TODO(noahric): These are flaky on windows on headless machines.
#ifndef WIN32
diff --git a/talk/media/devices/fakedevicemanager.h b/talk/media/devices/fakedevicemanager.h
index a4b2b86..77a8342 100644
--- a/talk/media/devices/fakedevicemanager.h
+++ b/talk/media/devices/fakedevicemanager.h
@@ -156,7 +156,7 @@
return true;
}
-#ifdef OSX
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
bool QtKitToSgDevice(const std::string& qtkit_name, Device* out) {
out->name = qtkit_name;
out->id = "sg:" + qtkit_name;
diff --git a/talk/media/devices/mobiledevicemanager.cc b/talk/media/devices/mobiledevicemanager.cc
index 2a886a3..5739c7e 100644
--- a/talk/media/devices/mobiledevicemanager.cc
+++ b/talk/media/devices/mobiledevicemanager.cc
@@ -27,7 +27,7 @@
#include "talk/media/devices/devicemanager.h"
#include "webrtc/base/arraysize.h"
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
+#include "webrtc/modules/video_capture/video_capture_factory.h"
namespace cricket {
diff --git a/talk/media/devices/v4llookup.h b/talk/media/devices/v4llookup.h
index 1bed90b6..5c53ede 100644
--- a/talk/media/devices/v4llookup.h
+++ b/talk/media/devices/v4llookup.h
@@ -37,7 +37,7 @@
#include <string>
-#ifdef LINUX
+#ifdef WEBRTC_LINUX
namespace cricket {
class V4LLookup {
public:
@@ -66,5 +66,5 @@
} // namespace cricket
-#endif // LINUX
+#endif // WEBRTC_LINUX
#endif // TALK_MEDIA_DEVICES_V4LLOOKUP_H_
diff --git a/talk/media/devices/videorendererfactory.h b/talk/media/devices/videorendererfactory.h
index 416f05b..b7128f6 100644
--- a/talk/media/devices/videorendererfactory.h
+++ b/talk/media/devices/videorendererfactory.h
@@ -32,9 +32,9 @@
#define TALK_MEDIA_DEVICES_VIDEORENDERERFACTORY_H_
#include "talk/media/base/videorenderer.h"
-#if defined(LINUX) && defined(HAVE_GTK)
+#if defined(WEBRTC_LINUX) && defined(HAVE_GTK)
#include "talk/media/devices/gtkvideorenderer.h"
-#elif defined(OSX) && !defined(CARBON_DEPRECATED)
+#elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) && !defined(CARBON_DEPRECATED)
#include "talk/media/devices/carbonvideorenderer.h"
#elif defined(WIN32)
#include "talk/media/devices/gdivideorenderer.h"
@@ -45,9 +45,10 @@
class VideoRendererFactory {
public:
static VideoRenderer* CreateGuiVideoRenderer(int x, int y) {
- #if defined(LINUX) && defined(HAVE_GTK)
+ #if defined(WEBRTC_LINUX) && defined(HAVE_GTK)
return new GtkVideoRenderer(x, y);
- #elif defined(OSX) && !defined(CARBON_DEPRECATED)
+ #elif defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) && \
+ !defined(CARBON_DEPRECATED)
CarbonVideoRenderer* renderer = new CarbonVideoRenderer(x, y);
// Needs to be initialized on the main thread.
if (renderer->Initialize()) {
diff --git a/talk/media/devices/win32devicemanager.cc b/talk/media/devices/win32devicemanager.cc
index 1b9e9d8..f34e3c4 100644
--- a/talk/media/devices/win32devicemanager.cc
+++ b/talk/media/devices/win32devicemanager.cc
@@ -48,6 +48,7 @@
} }, 4
};
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/thread.h"
@@ -148,7 +149,7 @@
*device = devices[0];
for (size_t i = 0; i < devices.size(); ++i) {
if (strnicmp(devices[i].id.c_str(), kUsbDevicePathPrefix,
- ARRAY_SIZE(kUsbDevicePathPrefix) - 1) == 0) {
+ arraysize(kUsbDevicePathPrefix) - 1) == 0) {
*device = devices[i];
break;
}
diff --git a/talk/media/sctp/sctpdataengine.cc b/talk/media/sctp/sctpdataengine.cc
index c88882d..3753cd2 100644
--- a/talk/media/sctp/sctpdataengine.cc
+++ b/talk/media/sctp/sctpdataengine.cc
@@ -36,6 +36,7 @@
#include "talk/media/base/constants.h"
#include "talk/media/base/streamparams.h"
#include "usrsctplib/usrsctp.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
@@ -76,7 +77,7 @@
MAKEFLAG(SCTP_STREAM_CHANGE_DENIED)
};
#undef MAKEFLAG
- for (int i = 0; i < ARRAY_SIZE(flaginfo); ++i) {
+ for (int i = 0; i < arraysize(flaginfo); ++i) {
if (flags & flaginfo[i].value) {
if (!first) result << " | ";
result << flaginfo[i].name;
@@ -473,7 +474,7 @@
struct sctp_event event = {0};
event.se_assoc_id = SCTP_ALL_ASSOC;
event.se_on = 1;
- for (size_t i = 0; i < ARRAY_SIZE(event_types); i++) {
+ for (size_t i = 0; i < arraysize(event_types); i++) {
event.se_type = event_types[i];
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EVENT, &event,
sizeof(event)) < 0) {
@@ -728,7 +729,13 @@
}
const uint32_t ssrc = stream.first_ssrc();
- if (open_streams_.find(ssrc) != open_streams_.end()) {
+ if (ssrc >= cricket::kMaxSctpSid) {
+ LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): "
+ << "Not adding data stream '" << stream.id
+ << "' with ssrc=" << ssrc
+ << " because stream ssrc is too high.";
+ return false;
+ } else if (open_streams_.find(ssrc) != open_streams_.end()) {
LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): "
<< "Not adding data stream '" << stream.id
<< "' with ssrc=" << ssrc
diff --git a/talk/media/sctp/sctpdataengine_unittest.cc b/talk/media/sctp/sctpdataengine_unittest.cc
index 4706368..d673c69 100644
--- a/talk/media/sctp/sctpdataengine_unittest.cc
+++ b/talk/media/sctp/sctpdataengine_unittest.cc
@@ -270,12 +270,14 @@
ProcessMessagesUntilIdle();
}
- void AddStream(int ssrc) {
+ bool AddStream(int ssrc) {
+ bool ret = true;
cricket::StreamParams p(cricket::StreamParams::CreateLegacy(ssrc));
- chan1_->AddSendStream(p);
- chan1_->AddRecvStream(p);
- chan2_->AddSendStream(p);
- chan2_->AddRecvStream(p);
+ ret = ret && chan1_->AddSendStream(p);
+ ret = ret && chan1_->AddRecvStream(p);
+ ret = ret && chan2_->AddSendStream(p);
+ ret = ret && chan2_->AddRecvStream(p);
+ return ret;
}
cricket::SctpDataMediaChannel* CreateChannel(
@@ -504,6 +506,12 @@
EXPECT_GT(channel1_ready_to_send_count(), prior_count);
}
+TEST_F(SctpDataMediaChannelTest, RefusesHighNumberedChannels) {
+ SetupConnectedChannels();
+ EXPECT_TRUE(AddStream(1022));
+ EXPECT_FALSE(AddStream(1023));
+}
+
// Flaky on Linux and Windows. See webrtc:4453.
#if defined(WEBRTC_WIN) || defined(WEBRTC_LINUX)
#define MAYBE_ReusesAStream DISABLED_ReusesAStream
diff --git a/talk/media/webrtc/fakewebrtccall.cc b/talk/media/webrtc/fakewebrtccall.cc
index d86bfb5..d50a53c 100644
--- a/talk/media/webrtc/fakewebrtccall.cc
+++ b/talk/media/webrtc/fakewebrtccall.cc
@@ -28,10 +28,12 @@
#include "talk/media/webrtc/fakewebrtccall.h"
#include <algorithm>
+#include <utility>
#include "talk/media/base/rtputils.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/gunit.h"
+#include "webrtc/audio/audio_sink.h"
namespace cricket {
FakeAudioSendStream::FakeAudioSendStream(
@@ -39,14 +41,27 @@
RTC_DCHECK(config.voe_channel_id != -1);
}
+const webrtc::AudioSendStream::Config&
+ FakeAudioSendStream::GetConfig() const {
+ return config_;
+}
+
void FakeAudioSendStream::SetStats(
const webrtc::AudioSendStream::Stats& stats) {
stats_ = stats;
}
-const webrtc::AudioSendStream::Config&
- FakeAudioSendStream::GetConfig() const {
- return config_;
+FakeAudioSendStream::TelephoneEvent
+ FakeAudioSendStream::GetLatestTelephoneEvent() const {
+ return latest_telephone_event_;
+}
+
+bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
+ uint32_t duration_ms) {
+ latest_telephone_event_.payload_type = payload_type;
+ latest_telephone_event_.event_code = event;
+ latest_telephone_event_.duration_ms = duration_ms;
+ return true;
}
webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
@@ -77,6 +92,11 @@
return stats_;
}
+void FakeAudioReceiveStream::SetSink(
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+ sink_ = std::move(sink);
+}
+
FakeVideoSendStream::FakeVideoSendStream(
const webrtc::VideoSendStream::Config& config,
const webrtc::VideoEncoderConfig& encoder_config)
diff --git a/talk/media/webrtc/fakewebrtccall.h b/talk/media/webrtc/fakewebrtccall.h
index 88edc60..3528c7a 100644
--- a/talk/media/webrtc/fakewebrtccall.h
+++ b/talk/media/webrtc/fakewebrtccall.h
@@ -47,14 +47,19 @@
#include "webrtc/video_send_stream.h"
namespace cricket {
-
-class FakeAudioSendStream : public webrtc::AudioSendStream {
+class FakeAudioSendStream final : public webrtc::AudioSendStream {
public:
- explicit FakeAudioSendStream(
- const webrtc::AudioSendStream::Config& config);
+ struct TelephoneEvent {
+ int payload_type = -1;
+ uint8_t event_code = 0;
+ uint32_t duration_ms = 0;
+ };
+
+ explicit FakeAudioSendStream(const webrtc::AudioSendStream::Config& config);
const webrtc::AudioSendStream::Config& GetConfig() const;
void SetStats(const webrtc::AudioSendStream::Stats& stats);
+ TelephoneEvent GetLatestTelephoneEvent() const;
private:
// webrtc::SendStream implementation.
@@ -66,13 +71,16 @@
}
// webrtc::AudioSendStream implementation.
+ bool SendTelephoneEvent(int payload_type, uint8_t event,
+ uint32_t duration_ms) override;
webrtc::AudioSendStream::Stats GetStats() const override;
+ TelephoneEvent latest_telephone_event_;
webrtc::AudioSendStream::Config config_;
webrtc::AudioSendStream::Stats stats_;
};
-class FakeAudioReceiveStream : public webrtc::AudioReceiveStream {
+class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
public:
explicit FakeAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config);
@@ -98,14 +106,16 @@
// webrtc::AudioReceiveStream implementation.
webrtc::AudioReceiveStream::Stats GetStats() const override;
+ void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
webrtc::AudioReceiveStream::Config config_;
webrtc::AudioReceiveStream::Stats stats_;
int received_packets_;
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;
};
-class FakeVideoSendStream : public webrtc::VideoSendStream,
- public webrtc::VideoCaptureInput {
+class FakeVideoSendStream final : public webrtc::VideoSendStream,
+ public webrtc::VideoCaptureInput {
public:
FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
const webrtc::VideoEncoderConfig& encoder_config);
@@ -153,7 +163,7 @@
webrtc::VideoSendStream::Stats stats_;
};
-class FakeVideoReceiveStream : public webrtc::VideoReceiveStream {
+class FakeVideoReceiveStream final : public webrtc::VideoReceiveStream {
public:
explicit FakeVideoReceiveStream(
const webrtc::VideoReceiveStream::Config& config);
@@ -188,7 +198,7 @@
webrtc::VideoReceiveStream::Stats stats_;
};
-class FakeCall : public webrtc::Call, public webrtc::PacketReceiver {
+class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
public:
explicit FakeCall(const webrtc::Call::Config& config);
~FakeCall() override;
diff --git a/talk/media/webrtc/fakewebrtcvideoengine.h b/talk/media/webrtc/fakewebrtcvideoengine.h
index 8e4c7c8..e0d4db5 100644
--- a/talk/media/webrtc/fakewebrtcvideoengine.h
+++ b/talk/media/webrtc/fakewebrtcvideoengine.h
@@ -41,7 +41,7 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/video_coding/codecs/interface/video_error_codes.h"
+#include "webrtc/modules/video_coding/include/video_error_codes.h"
#include "webrtc/video_decoder.h"
#include "webrtc/video_encoder.h"
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 2405e07..65ba927 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -41,19 +41,11 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/config.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace cricket {
-static const char kFakeDefaultDeviceName[] = "Fake Default";
-static const int kFakeDefaultDeviceId = -1;
-static const char kFakeDeviceName[] = "Fake Device";
-#ifdef WIN32
-static const int kFakeDeviceId = 0;
-#else
-static const int kFakeDeviceId = 1;
-#endif
-
static const int kOpusBandwidthNb = 4000;
static const int kOpusBandwidthMb = 6000;
static const int kOpusBandwidthWb = 8000;
@@ -63,18 +55,6 @@
#define WEBRTC_CHECK_CHANNEL(channel) \
if (channels_.find(channel) == channels_.end()) return -1;
-#define WEBRTC_ASSERT_CHANNEL(channel) \
- RTC_DCHECK(channels_.find(channel) != channels_.end());
-
-// Verify the header extension ID, if enabled, is within the bounds specified in
-// [RFC5285]: 1-14 inclusive.
-#define WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id) \
- do { \
- if (enable && (id < 1 || id > 14)) { \
- return -1; \
- } \
- } while (0);
-
class FakeAudioProcessing : public webrtc::AudioProcessing {
public:
FakeAudioProcessing() : experimental_ns_enabled_(false) {}
@@ -94,11 +74,13 @@
experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
}
+ WEBRTC_STUB_CONST(input_sample_rate_hz, ());
WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
- WEBRTC_STUB_CONST(num_input_channels, ());
- WEBRTC_STUB_CONST(num_output_channels, ());
- WEBRTC_STUB_CONST(num_reverse_channels, ());
+ size_t num_input_channels() const override { return 0; }
+ size_t num_proc_channels() const override { return 0; }
+ size_t num_output_channels() const override { return 0; }
+ size_t num_reverse_channels() const override { return 0; }
WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted));
WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
WEBRTC_STUB(ProcessStream, (
@@ -156,20 +138,11 @@
class FakeWebRtcVoiceEngine
: public webrtc::VoEAudioProcessing,
- public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
+ public webrtc::VoEBase, public webrtc::VoECodec,
public webrtc::VoEHardware,
public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
public webrtc::VoEVolumeControl {
public:
- struct DtmfInfo {
- DtmfInfo()
- : dtmf_event_code(-1),
- dtmf_out_of_band(false),
- dtmf_length_ms(-1) {}
- int dtmf_event_code;
- bool dtmf_out_of_band;
- int dtmf_length_ms;
- };
struct Channel {
explicit Channel()
: external_transport(false),
@@ -184,15 +157,11 @@
nack(false),
cn8_type(13),
cn16_type(105),
- dtmf_type(106),
red_type(117),
nack_max_packets(0),
send_ssrc(0),
- send_audio_level_ext_(-1),
- receive_audio_level_ext_(-1),
- send_absolute_sender_time_ext_(-1),
- receive_absolute_sender_time_ext_(-1),
associate_send_channel(-1),
+ recv_codecs(),
neteq_capacity(-1),
neteq_fast_accelerate(false) {
memset(&send_codec, 0, sizeof(send_codec));
@@ -209,16 +178,10 @@
bool nack;
int cn8_type;
int cn16_type;
- int dtmf_type;
int red_type;
int nack_max_packets;
uint32_t send_ssrc;
- int send_audio_level_ext_;
- int receive_audio_level_ext_;
- int send_absolute_sender_time_ext_;
- int receive_absolute_sender_time_ext_;
int associate_send_channel;
- DtmfInfo dtmf_info;
std::vector<webrtc::CodecInst> recv_codecs;
webrtc::CodecInst send_codec;
webrtc::PacketTime last_rtp_packet_time;
@@ -227,13 +190,10 @@
bool neteq_fast_accelerate;
};
- FakeWebRtcVoiceEngine(const cricket::AudioCodec* const* codecs,
- int num_codecs)
+ FakeWebRtcVoiceEngine()
: inited_(false),
last_channel_(-1),
fail_create_channel_(false),
- codecs_(codecs),
- num_codecs_(num_codecs),
num_set_send_codecs_(0),
ec_enabled_(false),
ec_metrics_enabled_(false),
@@ -255,26 +215,13 @@
memset(&agc_config_, 0, sizeof(agc_config_));
}
~FakeWebRtcVoiceEngine() {
- // Ought to have all been deleted by the WebRtcVoiceMediaChannel
- // destructors, but just in case ...
- for (std::map<int, Channel*>::const_iterator i = channels_.begin();
- i != channels_.end(); ++i) {
- delete i->second;
- }
+ RTC_CHECK(channels_.empty());
}
bool ec_metrics_enabled() const { return ec_metrics_enabled_; }
bool IsInited() const { return inited_; }
int GetLastChannel() const { return last_channel_; }
- int GetChannelFromLocalSsrc(uint32_t local_ssrc) const {
- for (std::map<int, Channel*>::const_iterator iter = channels_.begin();
- iter != channels_.end(); ++iter) {
- if (local_ssrc == iter->second->send_ssrc)
- return iter->first;
- }
- return -1;
- }
int GetNumChannels() const { return static_cast<int>(channels_.size()); }
uint32_t GetLocalSSRC(int channel) {
return channels_[channel]->send_ssrc;
@@ -307,7 +254,7 @@
return channels_[channel]->nack_max_packets;
}
const webrtc::PacketTime& GetLastRtpPacketTime(int channel) {
- WEBRTC_ASSERT_CHANNEL(channel);
+ RTC_DCHECK(channels_.find(channel) != channels_.end());
return channels_[channel]->last_rtp_packet_time;
}
int GetSendCNPayloadType(int channel, bool wideband) {
@@ -315,9 +262,6 @@
channels_[channel]->cn16_type :
channels_[channel]->cn8_type;
}
- int GetSendTelephoneEventPayloadType(int channel) {
- return channels_[channel]->dtmf_type;
- }
int GetSendREDPayloadType(int channel) {
return channels_[channel]->red_type;
}
@@ -351,11 +295,8 @@
return -1;
}
Channel* ch = new Channel();
- for (int i = 0; i < NumOfCodecs(); ++i) {
- webrtc::CodecInst codec;
- GetCodec(i, codec);
- ch->recv_codecs.push_back(codec);
- }
+ auto db = webrtc::acm2::RentACodec::Database();
+ ch->recv_codecs.assign(db.begin(), db.end());
if (config.Get<webrtc::NetEqCapacityConfig>().enabled) {
ch->neteq_capacity = config.Get<webrtc::NetEqCapacityConfig>().capacity;
}
@@ -364,24 +305,6 @@
channels_[++last_channel_] = ch;
return last_channel_;
}
- int GetSendRtpExtensionId(int channel, const std::string& extension) {
- WEBRTC_ASSERT_CHANNEL(channel);
- if (extension == kRtpAudioLevelHeaderExtension) {
- return channels_[channel]->send_audio_level_ext_;
- } else if (extension == kRtpAbsoluteSenderTimeHeaderExtension) {
- return channels_[channel]->send_absolute_sender_time_ext_;
- }
- return -1;
- }
- int GetReceiveRtpExtensionId(int channel, const std::string& extension) {
- WEBRTC_ASSERT_CHANNEL(channel);
- if (extension == kRtpAudioLevelHeaderExtension) {
- return channels_[channel]->receive_audio_level_ext_;
- } else if (extension == kRtpAbsoluteSenderTimeHeaderExtension) {
- return channels_[channel]->receive_absolute_sender_time_ext_;
- }
- return -1;
- }
int GetNumSetSendCodecs() const { return num_set_send_codecs_; }
@@ -473,22 +396,8 @@
webrtc::RtcEventLog* GetEventLog() { return nullptr; }
// webrtc::VoECodec
- WEBRTC_FUNC(NumOfCodecs, ()) {
- return num_codecs_;
- }
- WEBRTC_FUNC(GetCodec, (int index, webrtc::CodecInst& codec)) {
- if (index < 0 || index >= NumOfCodecs()) {
- return -1;
- }
- const cricket::AudioCodec& c(*codecs_[index]);
- codec.pltype = c.id;
- rtc::strcpyn(codec.plname, sizeof(codec.plname), c.name.c_str());
- codec.plfreq = c.clockrate;
- codec.pacsize = 0;
- codec.channels = c.channels;
- codec.rate = c.bitrate;
- return 0;
- }
+ WEBRTC_STUB(NumOfCodecs, ());
+ WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) {
WEBRTC_CHECK_CHANNEL(channel);
// To match the behavior of the real implementation.
@@ -526,16 +435,17 @@
}
}
// Otherwise try to find this codec and update its payload type.
+ int result = -1; // not found
for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
it != ch->recv_codecs.end(); ++it) {
if (strcmp(it->plname, codec.plname) == 0 &&
- it->plfreq == codec.plfreq) {
+ it->plfreq == codec.plfreq &&
+ it->channels == codec.channels) {
it->pltype = codec.pltype;
- it->channels = codec.channels;
- return 0;
+ result = 0;
}
}
- return -1; // not found
+ return result;
}
WEBRTC_FUNC(SetSendCNPayloadType, (int channel, int type,
webrtc::PayloadFrequencies frequency)) {
@@ -620,46 +530,11 @@
return 0;
}
- // webrtc::VoEDtmf
- WEBRTC_FUNC(SendTelephoneEvent, (int channel, int event_code,
- bool out_of_band = true, int length_ms = 160, int attenuation_db = 10)) {
- channels_[channel]->dtmf_info.dtmf_event_code = event_code;
- channels_[channel]->dtmf_info.dtmf_out_of_band = out_of_band;
- channels_[channel]->dtmf_info.dtmf_length_ms = length_ms;
- return 0;
- }
-
- WEBRTC_FUNC(SetSendTelephoneEventPayloadType,
- (int channel, unsigned char type)) {
- channels_[channel]->dtmf_type = type;
- return 0;
- };
- WEBRTC_STUB(GetSendTelephoneEventPayloadType,
- (int channel, unsigned char& type));
-
- WEBRTC_STUB(SetDtmfFeedbackStatus, (bool enable, bool directFeedback));
- WEBRTC_STUB(GetDtmfFeedbackStatus, (bool& enabled, bool& directFeedback));
-
- WEBRTC_FUNC(PlayDtmfTone,
- (int event_code, int length_ms = 200, int attenuation_db = 10)) {
- dtmf_info_.dtmf_event_code = event_code;
- dtmf_info_.dtmf_length_ms = length_ms;
- return 0;
- }
-
// webrtc::VoEHardware
- WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) {
- return GetNumDevices(num);
- }
- WEBRTC_FUNC(GetNumOfPlayoutDevices, (int& num)) {
- return GetNumDevices(num);
- }
- WEBRTC_FUNC(GetRecordingDeviceName, (int i, char* name, char* guid)) {
- return GetDeviceName(i, name, guid);
- }
- WEBRTC_FUNC(GetPlayoutDeviceName, (int i, char* name, char* guid)) {
- return GetDeviceName(i, name, guid);
- }
+ WEBRTC_STUB(GetNumOfRecordingDevices, (int& num));
+ WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num));
+ WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid));
+ WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid));
WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel));
WEBRTC_STUB(SetPlayoutDevice, (int));
WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
@@ -729,35 +604,14 @@
}
WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc));
WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc));
- WEBRTC_FUNC(SetSendAudioLevelIndicationStatus, (int channel, bool enable,
- unsigned char id)) {
- WEBRTC_CHECK_CHANNEL(channel);
- WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
- channels_[channel]->send_audio_level_ext_ = (enable) ? id : -1;
- return 0;
- }
- WEBRTC_FUNC(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable,
- unsigned char id)) {
- WEBRTC_CHECK_CHANNEL(channel);
- WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
- channels_[channel]->receive_audio_level_ext_ = (enable) ? id : -1;
- return 0;
- }
- WEBRTC_FUNC(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
- unsigned char id)) {
- WEBRTC_CHECK_CHANNEL(channel);
- WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
- channels_[channel]->send_absolute_sender_time_ext_ = (enable) ? id : -1;
- return 0;
- }
- WEBRTC_FUNC(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable,
- unsigned char id)) {
- WEBRTC_CHECK_CHANNEL(channel);
- WEBRTC_CHECK_HEADER_EXTENSION_ID(enable, id);
- channels_[channel]->receive_absolute_sender_time_ext_ = (enable) ? id : -1;
- return 0;
- }
-
+ WEBRTC_STUB(SetSendAudioLevelIndicationStatus, (int channel, bool enable,
+ unsigned char id));
+ WEBRTC_STUB(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable,
+ unsigned char id));
+ WEBRTC_STUB(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
+ unsigned char id));
+ WEBRTC_STUB(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable,
+ unsigned char id));
WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable));
WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled));
WEBRTC_STUB(SetRTCP_CNAME, (int channel, const char cname[256]));
@@ -776,22 +630,12 @@
unsigned int& discardedPackets));
WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats));
WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) {
- return SetFECStatus(channel, enable, redPayloadtype);
- }
- // TODO(minyue): remove the below function when transition to SetREDStatus
- // is finished.
- WEBRTC_FUNC(SetFECStatus, (int channel, bool enable, int redPayloadtype)) {
WEBRTC_CHECK_CHANNEL(channel);
channels_[channel]->red = enable;
channels_[channel]->red_type = redPayloadtype;
return 0;
}
WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) {
- return GetFECStatus(channel, enable, redPayloadtype);
- }
- // TODO(minyue): remove the below function when transition to GetREDStatus
- // is finished.
- WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable, int& redPayloadtype)) {
WEBRTC_CHECK_CHANNEL(channel);
enable = channels_[channel]->red;
redPayloadtype = channels_[channel]->red_type;
@@ -937,15 +781,6 @@
void EnableStereoChannelSwapping(bool enable) {
stereo_swapping_enabled_ = enable;
}
- bool WasSendTelephoneEventCalled(int channel, int event_code, int length_ms) {
- return (channels_[channel]->dtmf_info.dtmf_event_code == event_code &&
- channels_[channel]->dtmf_info.dtmf_out_of_band == true &&
- channels_[channel]->dtmf_info.dtmf_length_ms == length_ms);
- }
- bool WasPlayDtmfToneCalled(int event_code, int length_ms) {
- return (dtmf_info_.dtmf_event_code == event_code &&
- dtmf_info_.dtmf_length_ms == length_ms);
- }
int GetNetEqCapacity() const {
auto ch = channels_.find(last_channel_);
ASSERT(ch != channels_.end());
@@ -958,47 +793,10 @@
}
private:
- int GetNumDevices(int& num) {
-#ifdef WIN32
- num = 1;
-#else
- // On non-Windows platforms VE adds a special entry for the default device,
- // so if there is one physical device then there are two entries in the
- // list.
- num = 2;
-#endif
- return 0;
- }
-
- int GetDeviceName(int i, char* name, char* guid) {
- const char *s;
-#ifdef WIN32
- if (0 == i) {
- s = kFakeDeviceName;
- } else {
- return -1;
- }
-#else
- // See comment above.
- if (0 == i) {
- s = kFakeDefaultDeviceName;
- } else if (1 == i) {
- s = kFakeDeviceName;
- } else {
- return -1;
- }
-#endif
- strcpy(name, s);
- guid[0] = '\0';
- return 0;
- }
-
bool inited_;
int last_channel_;
std::map<int, Channel*> channels_;
bool fail_create_channel_;
- const cricket::AudioCodec* const* codecs_;
- int num_codecs_;
int num_set_send_codecs_; // how many times we call SetSendCodec().
bool ec_enabled_;
bool ec_metrics_enabled_;
@@ -1018,12 +816,9 @@
int send_fail_channel_;
int recording_sample_rate_;
int playout_sample_rate_;
- DtmfInfo dtmf_info_;
FakeAudioProcessing audio_processing_;
};
-#undef WEBRTC_CHECK_HEADER_EXTENSION_ID
-
} // namespace cricket
#endif // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_
diff --git a/talk/media/webrtc/simulcast.cc b/talk/media/webrtc/simulcast.cc
index f55d960..b67a363 100755
--- a/talk/media/webrtc/simulcast.cc
+++ b/talk/media/webrtc/simulcast.cc
@@ -29,9 +29,11 @@
#include "talk/media/base/streamparams.h"
#include "talk/media/webrtc/simulcast.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/system_wrappers/include/field_trial.h"
+
namespace cricket {
struct SimulcastFormat {
@@ -93,7 +95,7 @@
int FindSimulcastFormatIndex(int width, int height) {
MaybeExchangeWidthHeight(&width, &height);
- for (int i = 0; i < ARRAY_SIZE(kSimulcastFormats); ++i) {
+ for (int i = 0; i < arraysize(kSimulcastFormats); ++i) {
if (width >= kSimulcastFormats[i].width &&
height >= kSimulcastFormats[i].height) {
return i;
@@ -105,7 +107,7 @@
int FindSimulcastFormatIndex(int width, int height, size_t max_layers) {
MaybeExchangeWidthHeight(&width, &height);
- for (int i = 0; i < ARRAY_SIZE(kSimulcastFormats); ++i) {
+ for (int i = 0; i < arraysize(kSimulcastFormats); ++i) {
if (width >= kSimulcastFormats[i].width &&
height >= kSimulcastFormats[i].height &&
max_layers == kSimulcastFormats[i].max_layers) {
diff --git a/talk/media/webrtc/webrtcmediaengine.cc b/talk/media/webrtc/webrtcmediaengine.cc
index af202bd..31e5025 100644
--- a/talk/media/webrtc/webrtcmediaengine.cc
+++ b/talk/media/webrtc/webrtcmediaengine.cc
@@ -26,6 +26,9 @@
*/
#include "talk/media/webrtc/webrtcmediaengine.h"
+
+#include <algorithm>
+
#include "talk/media/webrtc/webrtcvideoengine2.h"
#include "talk/media/webrtc/webrtcvoiceengine.h"
@@ -68,44 +71,85 @@
return CreateWebRtcMediaEngine(adm, encoder_factory, decoder_factory);
}
-const char* kBweExtensionPriorities[] = {
- kRtpTransportSequenceNumberHeaderExtension,
- kRtpAbsoluteSenderTimeHeaderExtension, kRtpTimestampOffsetHeaderExtension};
-
-const size_t kBweExtensionPrioritiesLength =
- ARRAY_SIZE(kBweExtensionPriorities);
-
-int GetPriority(const RtpHeaderExtension& extension,
- const char* extension_prios[],
- size_t extension_prios_length) {
- for (size_t i = 0; i < extension_prios_length; ++i) {
- if (extension.uri == extension_prios[i])
- return static_cast<int>(i);
- }
- return -1;
-}
-
-std::vector<RtpHeaderExtension> FilterRedundantRtpExtensions(
- const std::vector<RtpHeaderExtension>& extensions,
- const char* extension_prios[],
- size_t extension_prios_length) {
- if (extensions.empty())
- return std::vector<RtpHeaderExtension>();
- std::vector<RtpHeaderExtension> filtered;
- std::map<int, const RtpHeaderExtension*> sorted;
- for (auto& extension : extensions) {
- int priority =
- GetPriority(extension, extension_prios, extension_prios_length);
- if (priority == -1) {
- filtered.push_back(extension);
- continue;
- } else {
- sorted[priority] = &extension;
+namespace {
+// Remove mutually exclusive extensions with lower priority.
+void DiscardRedundantExtensions(
+ std::vector<webrtc::RtpExtension>* extensions,
+ rtc::ArrayView<const char*> extensions_decreasing_prio) {
+ RTC_DCHECK(extensions);
+ bool found = false;
+ for (const char* name : extensions_decreasing_prio) {
+ auto it = std::find_if(extensions->begin(), extensions->end(),
+ [name](const webrtc::RtpExtension& rhs) {
+ return rhs.name == name;
+ });
+ if (it != extensions->end()) {
+ if (found) {
+ extensions->erase(it);
+ }
+ found = true;
}
}
- if (!sorted.empty())
- filtered.push_back(*sorted.begin()->second);
- return filtered;
+}
+} // namespace
+
+bool ValidateRtpExtensions(const std::vector<RtpHeaderExtension>& extensions) {
+ bool id_used[14] = {false};
+ for (const auto& extension : extensions) {
+ if (extension.id <= 0 || extension.id >= 15) {
+ LOG(LS_ERROR) << "Bad RTP extension ID: " << extension.ToString();
+ return false;
+ }
+ if (id_used[extension.id - 1]) {
+ LOG(LS_ERROR) << "Duplicate RTP extension ID: " << extension.ToString();
+ return false;
+ }
+ id_used[extension.id - 1] = true;
+ }
+ return true;
}
+std::vector<webrtc::RtpExtension> FilterRtpExtensions(
+ const std::vector<RtpHeaderExtension>& extensions,
+ bool (*supported)(const std::string&),
+ bool filter_redundant_extensions) {
+ RTC_DCHECK(ValidateRtpExtensions(extensions));
+ RTC_DCHECK(supported);
+ std::vector<webrtc::RtpExtension> result;
+
+ // Ignore any extensions that we don't recognize.
+ for (const auto& extension : extensions) {
+ if (supported(extension.uri)) {
+ result.push_back({extension.uri, extension.id});
+ } else {
+ LOG(LS_WARNING) << "Unsupported RTP extension: " << extension.ToString();
+ }
+ }
+
+ // Sort by name, ascending, so that we don't reset extensions if they were
+ // specified in a different order (also allows us to use std::unique below).
+ std::sort(result.begin(), result.end(),
+ [](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) {
+ return rhs.name < lhs.name;
+ });
+
+ // Remove unnecessary extensions (used on send side).
+ if (filter_redundant_extensions) {
+ auto it = std::unique(result.begin(), result.end(),
+ [](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) {
+ return rhs.name == lhs.name;
+ });
+ result.erase(it, result.end());
+
+ // Keep just the highest priority extension of any in the following list.
+ static const char* kBweExtensionPriorities[] = {
+ kRtpTransportSequenceNumberHeaderExtension,
+ kRtpAbsoluteSenderTimeHeaderExtension,
+ kRtpTimestampOffsetHeaderExtension
+ };
+ DiscardRedundantExtensions(&result, kBweExtensionPriorities);
+ }
+
+ return result;
+}
} // namespace cricket
diff --git a/talk/media/webrtc/webrtcmediaengine.h b/talk/media/webrtc/webrtcmediaengine.h
index 8d754040..831d072 100644
--- a/talk/media/webrtc/webrtcmediaengine.h
+++ b/talk/media/webrtc/webrtcmediaengine.h
@@ -28,7 +28,11 @@
#ifndef TALK_MEDIA_WEBRTCMEDIAENGINE_H_
#define TALK_MEDIA_WEBRTCMEDIAENGINE_H_
+#include <string>
+#include <vector>
+
#include "talk/media/base/mediaengine.h"
+#include "webrtc/config.h"
namespace webrtc {
class AudioDeviceModule;
@@ -48,13 +52,18 @@
WebRtcVideoDecoderFactory* decoder_factory);
};
-extern const char* kBweExtensionPriorities[];
-extern const size_t kBweExtensionPrioritiesLength;
+// Verify that extension IDs are within 1-byte extension range and are not
+// overlapping.
+bool ValidateRtpExtensions(const std::vector<RtpHeaderExtension>& extensions);
-std::vector<RtpHeaderExtension> FilterRedundantRtpExtensions(
+// Convert cricket::RtpHeaderExtension:s to webrtc::RtpExtension:s, discarding
+// any extensions not validated by the 'supported' predicate. Duplicate
+// extensions are removed if 'filter_redundant_extensions' is set, and also any
+// mutually exclusive extensions (see implementation for details).
+std::vector<webrtc::RtpExtension> FilterRtpExtensions(
const std::vector<RtpHeaderExtension>& extensions,
- const char* extension_prios[],
- size_t extension_prios_length);
+ bool (*supported)(const std::string&),
+ bool filter_redundant_extensions);
} // namespace cricket
diff --git a/talk/media/webrtc/webrtcmediaengine_unittest.cc b/talk/media/webrtc/webrtcmediaengine_unittest.cc
new file mode 100644
index 0000000..7c80e77
--- /dev/null
+++ b/talk/media/webrtc/webrtcmediaengine_unittest.cc
@@ -0,0 +1,205 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "talk/media/webrtc/webrtcmediaengine.h"
+
+namespace cricket {
+namespace {
+
+std::vector<RtpHeaderExtension> MakeUniqueExtensions() {
+ std::vector<RtpHeaderExtension> result;
+ char name[] = "a";
+ for (int i = 0; i < 7; ++i) {
+ result.push_back(RtpHeaderExtension(name, 1 + i));
+ name[0]++;
+ result.push_back(RtpHeaderExtension(name, 14 - i));
+ name[0]++;
+ }
+ return result;
+}
+
+std::vector<RtpHeaderExtension> MakeRedundantExtensions() {
+ std::vector<RtpHeaderExtension> result;
+ char name[] = "a";
+ for (int i = 0; i < 7; ++i) {
+ result.push_back(RtpHeaderExtension(name, 1 + i));
+ result.push_back(RtpHeaderExtension(name, 14 - i));
+ name[0]++;
+ }
+ return result;
+}
+
+bool SupportedExtensions1(const std::string& name) {
+ return name == "c" || name == "i";
+}
+
+bool SupportedExtensions2(const std::string& name) {
+ return name != "a" && name != "n";
+}
+
+bool IsSorted(const std::vector<webrtc::RtpExtension>& extensions) {
+ const std::string* last = nullptr;
+ for (const auto& extension : extensions) {
+ if (last && *last > extension.name) {
+ return false;
+ }
+ last = &extension.name;
+ }
+ return true;
+}
+} // namespace
+
+TEST(WebRtcMediaEngineTest, ValidateRtpExtensions_EmptyList) {
+ std::vector<RtpHeaderExtension> extensions;
+ EXPECT_TRUE(ValidateRtpExtensions(extensions));
+}
+
+TEST(WebRtcMediaEngineTest, ValidateRtpExtensions_AllGood) {
+ std::vector<RtpHeaderExtension> extensions = MakeUniqueExtensions();
+ EXPECT_TRUE(ValidateRtpExtensions(extensions));
+}
+
+TEST(WebRtcMediaEngineTest, ValidateRtpExtensions_OutOfRangeId_Low) {
+ std::vector<RtpHeaderExtension> extensions = MakeUniqueExtensions();
+ extensions.push_back(RtpHeaderExtension("foo", 0));
+ EXPECT_FALSE(ValidateRtpExtensions(extensions));
+}
+
+TEST(WebRtcMediaEngineTest, ValidateRtpExtensions_OutOfRangeId_High) {
+ std::vector<RtpHeaderExtension> extensions = MakeUniqueExtensions();
+ extensions.push_back(RtpHeaderExtension("foo", 15));
+ EXPECT_FALSE(ValidateRtpExtensions(extensions));
+}
+
+TEST(WebRtcMediaEngineTest, ValidateRtpExtensions_OverlappingIds_StartOfSet) {
+ std::vector<RtpHeaderExtension> extensions = MakeUniqueExtensions();
+ extensions.push_back(RtpHeaderExtension("foo", 1));
+ EXPECT_FALSE(ValidateRtpExtensions(extensions));
+}
+
+TEST(WebRtcMediaEngineTest, ValidateRtpExtensions_OverlappingIds_EndOfSet) {
+ std::vector<RtpHeaderExtension> extensions = MakeUniqueExtensions();
+ extensions.push_back(RtpHeaderExtension("foo", 14));
+ EXPECT_FALSE(ValidateRtpExtensions(extensions));
+}
+
+TEST(WebRtcMediaEngineTest, FilterRtpExtensions_EmptyList) {
+ std::vector<RtpHeaderExtension> extensions;
+ std::vector<webrtc::RtpExtension> filtered =
+ FilterRtpExtensions(extensions, SupportedExtensions1, true);
+ EXPECT_EQ(0, filtered.size());
+}
+
+TEST(WebRtcMediaEngineTest, FilterRtpExtensions_IncludeOnlySupported) {
+ std::vector<RtpHeaderExtension> extensions = MakeUniqueExtensions();
+ std::vector<webrtc::RtpExtension> filtered =
+ FilterRtpExtensions(extensions, SupportedExtensions1, false);
+ EXPECT_EQ(2, filtered.size());
+ EXPECT_EQ("c", filtered[0].name);
+ EXPECT_EQ("i", filtered[1].name);
+}
+
+TEST(WebRtcMediaEngineTest, FilterRtpExtensions_SortedByName_1) {
+ std::vector<RtpHeaderExtension> extensions = MakeUniqueExtensions();
+ std::vector<webrtc::RtpExtension> filtered =
+ FilterRtpExtensions(extensions, SupportedExtensions2, false);
+ EXPECT_EQ(12, filtered.size());
+ EXPECT_TRUE(IsSorted(filtered));
+}
+
+TEST(WebRtcMediaEngineTest, FilterRtpExtensions_SortedByName_2) {
+ std::vector<RtpHeaderExtension> extensions = MakeUniqueExtensions();
+ std::vector<webrtc::RtpExtension> filtered =
+ FilterRtpExtensions(extensions, SupportedExtensions2, true);
+ EXPECT_EQ(12, filtered.size());
+ EXPECT_TRUE(IsSorted(filtered));
+}
+
+TEST(WebRtcMediaEngineTest, FilterRtpExtensions_DontRemoveRedundant) {
+ std::vector<RtpHeaderExtension> extensions = MakeRedundantExtensions();
+ std::vector<webrtc::RtpExtension> filtered =
+ FilterRtpExtensions(extensions, SupportedExtensions2, false);
+ EXPECT_EQ(12, filtered.size());
+ EXPECT_TRUE(IsSorted(filtered));
+ EXPECT_EQ(filtered[0].name, filtered[1].name);
+}
+
+TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundant) {
+ std::vector<RtpHeaderExtension> extensions = MakeRedundantExtensions();
+ std::vector<webrtc::RtpExtension> filtered =
+ FilterRtpExtensions(extensions, SupportedExtensions2, true);
+ EXPECT_EQ(6, filtered.size());
+ EXPECT_TRUE(IsSorted(filtered));
+ EXPECT_NE(filtered[0].name, filtered[1].name);
+}
+
+TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_1) {
+ std::vector<RtpHeaderExtension> extensions;
+ extensions.push_back(
+ RtpHeaderExtension(kRtpTransportSequenceNumberHeaderExtension, 3));
+ extensions.push_back(
+ RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension, 9));
+ extensions.push_back(
+ RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, 6));
+ extensions.push_back(
+ RtpHeaderExtension(kRtpTransportSequenceNumberHeaderExtension, 1));
+ extensions.push_back(
+ RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension, 14));
+ std::vector<webrtc::RtpExtension> filtered =
+ FilterRtpExtensions(extensions, SupportedExtensions2, true);
+ EXPECT_EQ(1, filtered.size());
+ EXPECT_EQ(kRtpTransportSequenceNumberHeaderExtension, filtered[0].name);
+}
+
+TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_2) {
+ std::vector<RtpHeaderExtension> extensions;
+ extensions.push_back(
+ RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension, 1));
+ extensions.push_back(
+ RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension, 14));
+ extensions.push_back(
+ RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension, 7));
+ std::vector<webrtc::RtpExtension> filtered =
+ FilterRtpExtensions(extensions, SupportedExtensions2, true);
+ EXPECT_EQ(1, filtered.size());
+ EXPECT_EQ(kRtpAbsoluteSenderTimeHeaderExtension, filtered[0].name);
+}
+
+TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_3) {
+ std::vector<RtpHeaderExtension> extensions;
+ extensions.push_back(
+ RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension, 2));
+ extensions.push_back(
+ RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension, 14));
+ std::vector<webrtc::RtpExtension> filtered =
+ FilterRtpExtensions(extensions, SupportedExtensions2, true);
+ EXPECT_EQ(1, filtered.size());
+ EXPECT_EQ(kRtpTimestampOffsetHeaderExtension, filtered[0].name);
+}
+} // namespace cricket
diff --git a/talk/media/webrtc/webrtcvideocapturer.cc b/talk/media/webrtc/webrtcvideocapturer.cc
index 7d72128..ee4db5b 100644
--- a/talk/media/webrtc/webrtcvideocapturer.cc
+++ b/talk/media/webrtc/webrtcvideocapturer.cc
@@ -34,6 +34,7 @@
#ifdef HAVE_WEBRTC_VIDEO
#include "talk/media/webrtc/webrtcvideoframe.h"
#include "talk/media/webrtc/webrtcvideoframefactory.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/criticalsection.h"
@@ -43,7 +44,7 @@
#include "webrtc/base/timeutils.h"
#include "webrtc/base/win32.h" // Need this to #include the impl files.
-#include "webrtc/modules/video_capture/include/video_capture_factory.h"
+#include "webrtc/modules/video_capture/video_capture_factory.h"
#include "webrtc/system_wrappers/include/field_trial.h"
namespace cricket {
@@ -83,7 +84,7 @@
static bool CapabilityToFormat(const webrtc::VideoCaptureCapability& cap,
VideoFormat* format) {
uint32_t fourcc = 0;
- for (size_t i = 0; i < ARRAY_SIZE(kSupportedFourCCs); ++i) {
+ for (size_t i = 0; i < arraysize(kSupportedFourCCs); ++i) {
if (kSupportedFourCCs[i].webrtc_type == cap.rawType) {
fourcc = kSupportedFourCCs[i].fourcc;
break;
@@ -103,7 +104,7 @@
static bool FormatToCapability(const VideoFormat& format,
webrtc::VideoCaptureCapability* cap) {
webrtc::RawVideoType webrtc_type = webrtc::kVideoUnknown;
- for (size_t i = 0; i < ARRAY_SIZE(kSupportedFourCCs); ++i) {
+ for (size_t i = 0; i < arraysize(kSupportedFourCCs); ++i) {
if (kSupportedFourCCs[i].fourcc == format.fourcc) {
webrtc_type = kSupportedFourCCs[i].webrtc_type;
break;
@@ -171,8 +172,8 @@
bool found = false;
for (int index = 0; index < num_cams; ++index) {
char vcm_name[256];
- if (info->GetDeviceName(index, vcm_name, ARRAY_SIZE(vcm_name),
- vcm_id, ARRAY_SIZE(vcm_id)) != -1) {
+ if (info->GetDeviceName(index, vcm_name, arraysize(vcm_name), vcm_id,
+ arraysize(vcm_id)) != -1) {
if (device.name == reinterpret_cast<char*>(vcm_name)) {
found = true;
break;
@@ -349,6 +350,7 @@
SetCaptureFormat(NULL);
start_thread_ = nullptr;
+ SetCaptureState(CS_STOPPED);
}
bool WebRtcVideoCapturer::IsRunning() {
@@ -361,7 +363,7 @@
}
fourccs->clear();
- for (size_t i = 0; i < ARRAY_SIZE(kSupportedFourCCs); ++i) {
+ for (size_t i = 0; i < arraysize(kSupportedFourCCs); ++i) {
fourccs->push_back(kSupportedFourCCs[i].fourcc);
}
return true;
diff --git a/talk/media/webrtc/webrtcvideocapturer.h b/talk/media/webrtc/webrtcvideocapturer.h
index 0a99884..591e46f 100644
--- a/talk/media/webrtc/webrtcvideocapturer.h
+++ b/talk/media/webrtc/webrtcvideocapturer.h
@@ -39,7 +39,7 @@
#include "webrtc/base/messagehandler.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/video_capture/include/video_capture.h"
+#include "webrtc/modules/video_capture/video_capture.h"
namespace cricket {
diff --git a/talk/media/webrtc/webrtcvideocapturer_unittest.cc b/talk/media/webrtc/webrtcvideocapturer_unittest.cc
index d560fc5..85db32e 100644
--- a/talk/media/webrtc/webrtcvideocapturer_unittest.cc
+++ b/talk/media/webrtc/webrtcvideocapturer_unittest.cc
@@ -111,6 +111,7 @@
capturer_->Stop();
EXPECT_FALSE(capturer_->IsRunning());
EXPECT_TRUE(capturer_->GetCaptureFormat() == NULL);
+ EXPECT_EQ_WAIT(cricket::CS_STOPPED, listener_.last_capture_state(), 1000);
}
TEST_F(WebRtcVideoCapturerTest, TestCaptureVcm) {
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index bcd513e..55c0742 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -152,9 +152,7 @@
return true;
}
if (CodecNamesEq(codec_name, kVp9CodecName)) {
- const std::string group_name =
- webrtc::field_trial::FindFullName("WebRTC-SupportVP9");
- return group_name == "Enabled" || group_name == "EnabledByFlag";
+ return true;
}
if (CodecNamesEq(codec_name, kH264CodecName)) {
return webrtc::H264Encoder::IsSupported() &&
@@ -168,6 +166,8 @@
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kParamValueEmpty));
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli));
codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
+ codec->AddFeedbackParam(
+ FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
}
static VideoCodec MakeVideoCodecWithDefaultFeedbackParams(int payload_type,
@@ -243,20 +243,6 @@
return true;
}
-static std::string RtpExtensionsToString(
- const std::vector<RtpHeaderExtension>& extensions) {
- std::stringstream out;
- out << '{';
- for (size_t i = 0; i < extensions.size(); ++i) {
- out << "{" << extensions[i].uri << ": " << extensions[i].id << "}";
- if (i != extensions.size() - 1) {
- out << ", ";
- }
- }
- out << '}';
- return out.str();
-}
-
inline const webrtc::RtpExtension* FindHeaderExtension(
const std::vector<webrtc::RtpExtension>& extensions,
const std::string& name) {
@@ -303,7 +289,8 @@
// Returns true if the given codec is disallowed from doing simulcast.
bool IsCodecBlacklistedForSimulcast(const std::string& codec_name) {
- return CodecNamesEq(codec_name, kH264CodecName);
+ return CodecNamesEq(codec_name, kH264CodecName) ||
+ CodecNamesEq(codec_name, kVp9CodecName);
}
// The selected thresholds for QVGA and VGA corresponded to a QP around 10.
@@ -339,13 +326,13 @@
std::vector<VideoCodec> DefaultVideoCodecList() {
std::vector<VideoCodec> codecs;
+ codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType,
+ kVp8CodecName));
if (CodecIsInternallySupported(kVp9CodecName)) {
codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp9PlType,
kVp9CodecName));
// TODO(andresp): Add rtx codec for vp9 and verify it works.
}
- codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType,
- kVp8CodecName));
if (CodecIsInternallySupported(kH264CodecName)) {
codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultH264PlType,
kH264CodecName));
@@ -357,72 +344,6 @@
return codecs;
}
-static bool FindFirstMatchingCodec(const std::vector<VideoCodec>& codecs,
- const VideoCodec& requested_codec,
- VideoCodec* matching_codec) {
- for (size_t i = 0; i < codecs.size(); ++i) {
- if (requested_codec.Matches(codecs[i])) {
- *matching_codec = codecs[i];
- return true;
- }
- }
- return false;
-}
-
-static bool ValidateRtpHeaderExtensionIds(
- const std::vector<RtpHeaderExtension>& extensions) {
- std::set<int> extensions_used;
- for (size_t i = 0; i < extensions.size(); ++i) {
- if (extensions[i].id <= 0 || extensions[i].id >= 15 ||
- !extensions_used.insert(extensions[i].id).second) {
- LOG(LS_ERROR) << "RTP extensions are with incorrect or duplicate ids.";
- return false;
- }
- }
- return true;
-}
-
-static bool CompareRtpHeaderExtensionIds(
- const webrtc::RtpExtension& extension1,
- const webrtc::RtpExtension& extension2) {
- // Sorting on ID is sufficient, more than one extension per ID is unsupported.
- return extension1.id > extension2.id;
-}
-
-static std::vector<webrtc::RtpExtension> FilterRtpExtensions(
- const std::vector<RtpHeaderExtension>& extensions) {
- std::vector<webrtc::RtpExtension> webrtc_extensions;
- for (size_t i = 0; i < extensions.size(); ++i) {
- // Unsupported extensions will be ignored.
- if (webrtc::RtpExtension::IsSupportedForVideo(extensions[i].uri)) {
- webrtc_extensions.push_back(webrtc::RtpExtension(
- extensions[i].uri, extensions[i].id));
- } else {
- LOG(LS_WARNING) << "Unsupported RTP extension: " << extensions[i].uri;
- }
- }
-
- // Sort filtered headers to make sure that they can later be compared
- // regardless of in which order they were entered.
- std::sort(webrtc_extensions.begin(), webrtc_extensions.end(),
- CompareRtpHeaderExtensionIds);
- return webrtc_extensions;
-}
-
-static bool RtpExtensionsHaveChanged(
- const std::vector<webrtc::RtpExtension>& before,
- const std::vector<webrtc::RtpExtension>& after) {
- if (before.size() != after.size())
- return true;
- for (size_t i = 0; i < before.size(); ++i) {
- if (before[i].id != after[i].id)
- return true;
- if (before[i].name != after[i].name)
- return true;
- }
- return false;
-}
-
std::vector<webrtc::VideoStream>
WebRtcVideoChannel2::WebRtcVideoSendStream::CreateSimulcastVideoStreams(
const VideoCodec& codec,
@@ -489,7 +410,8 @@
denoising = false;
} else {
// Use codec default if video_noise_reduction is unset.
- codec_default_denoising = !options.video_noise_reduction.Get(&denoising);
+ codec_default_denoising = !options.video_noise_reduction;
+ denoising = options.video_noise_reduction.value_or(false);
}
if (CodecNamesEq(codec.name, kVp8CodecName)) {
@@ -554,20 +476,6 @@
external_encoder_factory_(NULL) {
LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
video_codecs_ = GetSupportedCodecs();
- rtp_header_extensions_.push_back(
- RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
- kRtpTimestampOffsetHeaderExtensionDefaultId));
- rtp_header_extensions_.push_back(
- RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
- kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
- rtp_header_extensions_.push_back(
- RtpHeaderExtension(kRtpVideoRotationHeaderExtension,
- kRtpVideoRotationHeaderExtensionDefaultId));
- if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
- rtp_header_extensions_.push_back(RtpHeaderExtension(
- kRtpTransportSequenceNumberHeaderExtension,
- kRtpTransportSequenceNumberHeaderExtensionDefaultId));
- }
}
WebRtcVideoEngine2::~WebRtcVideoEngine2() {
@@ -579,29 +487,6 @@
initialized_ = true;
}
-bool WebRtcVideoEngine2::SetDefaultEncoderConfig(
- const VideoEncoderConfig& config) {
- const VideoCodec& codec = config.max_codec;
- bool supports_codec = false;
- for (size_t i = 0; i < video_codecs_.size(); ++i) {
- if (CodecNamesEq(video_codecs_[i].name, codec.name)) {
- video_codecs_[i].width = codec.width;
- video_codecs_[i].height = codec.height;
- video_codecs_[i].framerate = codec.framerate;
- supports_codec = true;
- break;
- }
- }
-
- if (!supports_codec) {
- LOG(LS_ERROR) << "SetDefaultEncoderConfig, codec not supported: "
- << codec.ToString();
- return false;
- }
-
- return true;
-}
-
WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel(
webrtc::Call* call,
const VideoOptions& options) {
@@ -615,19 +500,23 @@
return video_codecs_;
}
-const std::vector<RtpHeaderExtension>&
-WebRtcVideoEngine2::rtp_header_extensions() const {
- return rtp_header_extensions_;
-}
-
-void WebRtcVideoEngine2::SetLogging(int min_sev, const char* filter) {
- // TODO(pbos): Set up logging.
- LOG(LS_VERBOSE) << "SetLogging: " << min_sev << '"' << filter << '"';
- // if min_sev == -1, we keep the current log level.
- if (min_sev < 0) {
- RTC_DCHECK(min_sev == -1);
- return;
+RtpCapabilities WebRtcVideoEngine2::GetCapabilities() const {
+ RtpCapabilities capabilities;
+ capabilities.header_extensions.push_back(
+ RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
+ kRtpTimestampOffsetHeaderExtensionDefaultId));
+ capabilities.header_extensions.push_back(
+ RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
+ kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
+ capabilities.header_extensions.push_back(
+ RtpHeaderExtension(kRtpVideoRotationHeaderExtension,
+ kRtpVideoRotationHeaderExtensionDefaultId));
+ if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
+ capabilities.header_extensions.push_back(RtpHeaderExtension(
+ kRtpTransportSequenceNumberHeaderExtension,
+ kRtpTransportSequenceNumberHeaderExtensionDefaultId));
}
+ return capabilities;
}
void WebRtcVideoEngine2::SetExternalDecoderFactory(
@@ -677,48 +566,6 @@
return false;
}
-// Tells whether the |requested| codec can be transmitted or not. If it can be
-// transmitted |out| is set with the best settings supported. Aspect ratio will
-// be set as close to |current|'s as possible. If not set |requested|'s
-// dimensions will be used for aspect ratio matching.
-bool WebRtcVideoEngine2::CanSendCodec(const VideoCodec& requested,
- const VideoCodec& current,
- VideoCodec* out) {
- RTC_DCHECK(out != NULL);
-
- if (requested.width != requested.height &&
- (requested.height == 0 || requested.width == 0)) {
- // 0xn and nx0 are invalid resolutions.
- return false;
- }
-
- VideoCodec matching_codec;
- if (!FindFirstMatchingCodec(video_codecs_, requested, &matching_codec)) {
- // Codec not supported.
- return false;
- }
-
- out->id = requested.id;
- out->name = requested.name;
- out->preference = requested.preference;
- out->params = requested.params;
- out->framerate = std::min(requested.framerate, matching_codec.framerate);
- out->params = requested.params;
- out->feedback_params = requested.feedback_params;
- out->width = requested.width;
- out->height = requested.height;
- if (requested.width == 0 && requested.height == 0) {
- return true;
- }
-
- while (out->width > matching_codec.width) {
- out->width /= 2;
- out->height /= 2;
- }
-
- return out->width > 0 && out->height > 0;
-}
-
// Ignore spammy trace messages, mostly from the stats API when we haven't
// gotten RTCP info yet from the remote side.
bool WebRtcVideoEngine2::ShouldIgnoreTrace(const std::string& trace) {
@@ -777,7 +624,8 @@
RTC_DCHECK(thread_checker_.CalledOnValidThread());
SetDefaultOptions();
options_.SetAll(options);
- options_.cpu_overuse_detection.Get(&signal_cpu_adaptation_);
+ if (options_.cpu_overuse_detection)
+ signal_cpu_adaptation_ = *options_.cpu_overuse_detection;
rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
sending_ = false;
default_send_ssrc_ = 0;
@@ -785,10 +633,10 @@
}
void WebRtcVideoChannel2::SetDefaultOptions() {
- options_.cpu_overuse_detection.Set(true);
- options_.dscp.Set(false);
- options_.suspend_below_min_bitrate.Set(false);
- options_.screencast_min_bitrate.Set(0);
+ options_.cpu_overuse_detection = rtc::Optional<bool>(true);
+ options_.dscp = rtc::Optional<bool>(false);
+ options_.suspend_below_min_bitrate = rtc::Optional<bool>(false);
+ options_.screencast_min_bitrate = rtc::Optional<int>(0);
}
WebRtcVideoChannel2::~WebRtcVideoChannel2() {
@@ -863,19 +711,43 @@
}
bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
+ TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendParameters");
+ LOG(LS_INFO) << "SetSendParameters: " << params.ToString();
// TODO(pbos): Refactor this to only recreate the send streams once
// instead of 4 times.
- return (SetSendCodecs(params.codecs) &&
- SetSendRtpHeaderExtensions(params.extensions) &&
- SetMaxSendBandwidth(params.max_bandwidth_bps) &&
- SetOptions(params.options));
+ if (!SetSendCodecs(params.codecs) ||
+ !SetSendRtpHeaderExtensions(params.extensions) ||
+ !SetMaxSendBandwidth(params.max_bandwidth_bps) ||
+ !SetOptions(params.options)) {
+ return false;
+ }
+ if (send_params_.rtcp.reduced_size != params.rtcp.reduced_size) {
+ rtc::CritScope stream_lock(&stream_crit_);
+ for (auto& kv : send_streams_) {
+ kv.second->SetSendParameters(params);
+ }
+ }
+ send_params_ = params;
+ return true;
}
bool WebRtcVideoChannel2::SetRecvParameters(const VideoRecvParameters& params) {
+ TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvParameters");
+ LOG(LS_INFO) << "SetRecvParameters: " << params.ToString();
// TODO(pbos): Refactor this to only recreate the recv streams once
// instead of twice.
- return (SetRecvCodecs(params.codecs) &&
- SetRecvRtpHeaderExtensions(params.extensions));
+ if (!SetRecvCodecs(params.codecs) ||
+ !SetRecvRtpHeaderExtensions(params.extensions)) {
+ return false;
+ }
+ if (recv_params_.rtcp.reduced_size != params.rtcp.reduced_size) {
+ rtc::CritScope stream_lock(&stream_crit_);
+ for (auto& kv : receive_streams_) {
+ kv.second->SetRecvParameters(params);
+ }
+ }
+ recv_params_ = params;
+ return true;
}
std::string WebRtcVideoChannel2::CodecSettingsVectorToString(
@@ -952,15 +824,15 @@
LOG(LS_INFO) << "Using codec: " << supported_codecs.front().codec.ToString();
- VideoCodecSettings old_codec;
- if (send_codec_.Get(&old_codec) && supported_codecs.front() == old_codec) {
+ if (send_codec_ && supported_codecs.front() == *send_codec_) {
LOG(LS_INFO) << "Ignore call to SetSendCodecs because first supported "
"codec hasn't changed.";
// Using same codec, avoid reconfiguring.
return true;
}
- send_codec_.Set(supported_codecs.front());
+ send_codec_ = rtc::Optional<WebRtcVideoChannel2::VideoCodecSettings>(
+ supported_codecs.front());
rtc::CritScope stream_lock(&stream_crit_);
LOG(LS_INFO) << "Change the send codec because SetSendCodecs has a different "
@@ -969,12 +841,15 @@
RTC_DCHECK(kv.second != nullptr);
kv.second->SetCodec(supported_codecs.front());
}
- LOG(LS_INFO) << "SetNackAndRemb on all the receive streams because the send "
- "codec has changed.";
+ LOG(LS_INFO)
+ << "SetFeedbackOptions on all the receive streams because the send "
+ "codec has changed.";
for (auto& kv : receive_streams_) {
RTC_DCHECK(kv.second != nullptr);
- kv.second->SetNackAndRemb(HasNack(supported_codecs.front().codec),
- HasRemb(supported_codecs.front().codec));
+ kv.second->SetFeedbackParameters(
+ HasNack(supported_codecs.front().codec),
+ HasRemb(supported_codecs.front().codec),
+ HasTransportCc(supported_codecs.front().codec));
}
// TODO(holmer): Changing the codec parameters shouldn't necessarily mean that
@@ -1006,12 +881,11 @@
}
bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) {
- VideoCodecSettings codec_settings;
- if (!send_codec_.Get(&codec_settings)) {
+ if (!send_codec_) {
LOG(LS_VERBOSE) << "GetSendCodec: No send codec set.";
return false;
}
- *codec = codec_settings.codec;
+ *codec = send_codec_->codec;
return true;
}
@@ -1028,7 +902,7 @@
bool WebRtcVideoChannel2::SetSend(bool send) {
LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
- if (send && !send_codec_.IsSet()) {
+ if (send && !send_codec_) {
LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
return false;
}
@@ -1094,15 +968,10 @@
webrtc::VideoSendStream::Config config(this);
config.overuse_callback = this;
- WebRtcVideoSendStream* stream =
- new WebRtcVideoSendStream(call_,
- sp,
- config,
- external_encoder_factory_,
- options_,
- bitrate_config_.max_bitrate_bps,
- send_codec_,
- send_rtp_extensions_);
+ WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
+ call_, sp, config, external_encoder_factory_, options_,
+ bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_,
+ send_params_);
uint32_t ssrc = sp.first_ssrc();
RTC_DCHECK(ssrc != 0);
@@ -1224,15 +1093,13 @@
// Set up A/V sync group based on sync label.
config.sync_group = sp.sync_label;
- config.rtp.remb = false;
- VideoCodecSettings send_codec;
- if (send_codec_.Get(&send_codec)) {
- config.rtp.remb = HasRemb(send_codec.codec);
- }
+ config.rtp.remb = send_codec_ ? HasRemb(send_codec_->codec) : false;
+ config.rtp.transport_cc =
+ send_codec_ ? HasTransportCc(send_codec_->codec) : false;
receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
call_, sp, config, external_decoder_factory_, default_stream,
- recv_codecs_);
+ recv_codecs_, options_.disable_prerenderer_smoothing.value_or(false));
return true;
}
@@ -1246,6 +1113,9 @@
config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
config->rtp.extensions = recv_rtp_extensions_;
+ config->rtp.rtcp_mode = recv_params_.rtcp.reduced_size
+ ? webrtc::RtcpMode::kReducedSize
+ : webrtc::RtcpMode::kCompound;
// TODO(pbos): This protection is against setting the same local ssrc as
// remote which is not permitted by the lower-level API. RTCP requires a
@@ -1482,12 +1352,14 @@
const rtc::PacketTime& packet_time) {
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
packet_time.not_before);
- if (call_->Receiver()->DeliverPacket(
- webrtc::MediaType::VIDEO,
- reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
- webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) {
- LOG(LS_WARNING) << "Failed to deliver RTCP packet.";
- }
+ // TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver
+ // for both audio and video on the same path. Since BundleFilter doesn't
+ // filter RTCP anymore incoming RTCP packets could've been going to audio (so
+ // logging failures spam the log).
+ call_->Receiver()->DeliverPacket(
+ webrtc::MediaType::VIDEO,
+ reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+ webrtc_packet_time);
}
void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
@@ -1512,20 +1384,17 @@
bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvRtpHeaderExtensions");
- LOG(LS_INFO) << "SetRecvRtpHeaderExtensions: "
- << RtpExtensionsToString(extensions);
- if (!ValidateRtpHeaderExtensionIds(extensions))
+ if (!ValidateRtpExtensions(extensions)) {
return false;
-
- std::vector<webrtc::RtpExtension> filtered_extensions =
- FilterRtpExtensions(extensions);
- if (!RtpExtensionsHaveChanged(recv_rtp_extensions_, filtered_extensions)) {
+ }
+ std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
+ extensions, webrtc::RtpExtension::IsSupportedForVideo, false);
+ if (recv_rtp_extensions_ == filtered_extensions) {
LOG(LS_INFO) << "Ignoring call to SetRecvRtpHeaderExtensions because "
"header extensions haven't changed.";
return true;
}
-
- recv_rtp_extensions_ = filtered_extensions;
+ recv_rtp_extensions_.swap(filtered_extensions);
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
@@ -1539,21 +1408,17 @@
bool WebRtcVideoChannel2::SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendRtpHeaderExtensions");
- LOG(LS_INFO) << "SetSendRtpHeaderExtensions: "
- << RtpExtensionsToString(extensions);
- if (!ValidateRtpHeaderExtensionIds(extensions))
+ if (!ValidateRtpExtensions(extensions)) {
return false;
-
- std::vector<webrtc::RtpExtension> filtered_extensions =
- FilterRtpExtensions(FilterRedundantRtpExtensions(
- extensions, kBweExtensionPriorities, kBweExtensionPrioritiesLength));
- if (!RtpExtensionsHaveChanged(send_rtp_extensions_, filtered_extensions)) {
- LOG(LS_INFO) << "Ignoring call to SetSendRtpHeaderExtensions because "
+ }
+ std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
+ extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
+ if (send_rtp_extensions_ == filtered_extensions) {
+ LOG(LS_INFO) << "Ignoring call to SetRecvRtpHeaderExtensions because "
"header extensions haven't changed.";
return true;
}
-
- send_rtp_extensions_ = filtered_extensions;
+ send_rtp_extensions_.swap(filtered_extensions);
const webrtc::RtpExtension* cvo_extension = FindHeaderExtension(
send_rtp_extensions_, kRtpVideoRotationHeaderExtension);
@@ -1612,11 +1477,11 @@
}
{
rtc::CritScope lock(&capturer_crit_);
- options_.cpu_overuse_detection.Get(&signal_cpu_adaptation_);
+ if (options_.cpu_overuse_detection)
+ signal_cpu_adaptation_ = *options_.cpu_overuse_detection;
}
- rtc::DiffServCodePoint dscp = options_.dscp.GetWithDefaultIfUnset(false)
- ? rtc::DSCP_AF41
- : rtc::DSCP_DEFAULT;
+ rtc::DiffServCodePoint dscp =
+ options_.dscp.value_or(false) ? rtc::DSCP_AF41 : rtc::DSCP_DEFAULT;
MediaChannel::SetDscp(dscp);
rtc::CritScope stream_lock(&stream_crit_);
for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
@@ -1708,12 +1573,11 @@
const webrtc::VideoSendStream::Config& config,
const VideoOptions& options,
int max_bitrate_bps,
- const Settable<VideoCodecSettings>& codec_settings)
+ const rtc::Optional<VideoCodecSettings>& codec_settings)
: config(config),
options(options),
max_bitrate_bps(max_bitrate_bps),
- codec_settings(codec_settings) {
-}
+ codec_settings(codec_settings) {}
WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder::AllocatedEncoder(
webrtc::VideoEncoder* encoder,
@@ -1737,8 +1601,11 @@
WebRtcVideoEncoderFactory* external_encoder_factory,
const VideoOptions& options,
int max_bitrate_bps,
- const Settable<VideoCodecSettings>& codec_settings,
- const std::vector<webrtc::RtpExtension>& rtp_extensions)
+ const rtc::Optional<VideoCodecSettings>& codec_settings,
+ const std::vector<webrtc::RtpExtension>& rtp_extensions,
+ // TODO(deadbeef): Don't duplicate information between send_params,
+ // rtp_extensions, options, etc.
+ const VideoSendParameters& send_params)
: ssrcs_(sp.ssrcs),
ssrc_groups_(sp.ssrc_groups),
call_(call),
@@ -1759,10 +1626,12 @@
¶meters_.config.rtp.rtx.ssrcs);
parameters_.config.rtp.c_name = sp.cname;
parameters_.config.rtp.extensions = rtp_extensions;
+ parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
+ ? webrtc::RtcpMode::kReducedSize
+ : webrtc::RtcpMode::kCompound;
- VideoCodecSettings params;
- if (codec_settings.Get(¶ms)) {
- SetCodec(params);
+ if (codec_settings) {
+ SetCodec(*codec_settings);
}
}
@@ -1940,11 +1809,10 @@
void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
const VideoOptions& options) {
rtc::CritScope cs(&lock_);
- VideoCodecSettings codec_settings;
- if (parameters_.codec_settings.Get(&codec_settings)) {
+ if (parameters_.codec_settings) {
LOG(LS_INFO) << "SetCodecAndOptions because of SetOptions; options="
<< options.ToString();
- SetCodecAndOptions(codec_settings, options);
+ SetCodecAndOptions(*parameters_.codec_settings, options);
} else {
parameters_.options = options;
}
@@ -2049,10 +1917,12 @@
parameters_.config.rtp.nack.rtp_history_ms =
HasNack(codec_settings.codec) ? kNackHistoryMs : 0;
- options.suspend_below_min_bitrate.Get(
- ¶meters_.config.suspend_below_min_bitrate);
+ RTC_CHECK(options.suspend_below_min_bitrate);
+ parameters_.config.suspend_below_min_bitrate =
+ *options.suspend_below_min_bitrate;
- parameters_.codec_settings.Set(codec_settings);
+ parameters_.codec_settings =
+ rtc::Optional<WebRtcVideoChannel2::VideoCodecSettings>(codec_settings);
parameters_.options = options;
LOG(LS_INFO)
@@ -2075,17 +1945,27 @@
}
}
+void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
+ const VideoSendParameters& send_params) {
+ rtc::CritScope cs(&lock_);
+ parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
+ ? webrtc::RtcpMode::kReducedSize
+ : webrtc::RtcpMode::kCompound;
+ if (stream_ != nullptr) {
+ LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters";
+ RecreateWebRtcStream();
+ }
+}
+
webrtc::VideoEncoderConfig
WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
const Dimensions& dimensions,
const VideoCodec& codec) const {
webrtc::VideoEncoderConfig encoder_config;
if (dimensions.is_screencast) {
- int screencast_min_bitrate_kbps;
- parameters_.options.screencast_min_bitrate.Get(
- &screencast_min_bitrate_kbps);
+ RTC_CHECK(parameters_.options.screencast_min_bitrate);
encoder_config.min_transmit_bitrate_bps =
- screencast_min_bitrate_kbps * 1000;
+ *parameters_.options.screencast_min_bitrate * 1000;
encoder_config.content_type =
webrtc::VideoEncoderConfig::ContentType::kScreen;
} else {
@@ -2121,7 +2001,7 @@
parameters_.max_bitrate_bps, stream_count);
// Conference mode screencast uses 2 temporal layers split at 100kbit.
- if (parameters_.options.conference_mode.GetWithDefaultIfUnset(false) &&
+ if (parameters_.options.conference_mode.value_or(false) &&
dimensions.is_screencast && encoder_config.streams.size() == 1) {
ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
@@ -2156,8 +2036,8 @@
RTC_DCHECK(!parameters_.encoder_config.streams.empty());
- VideoCodecSettings codec_settings;
- parameters_.codec_settings.Get(&codec_settings);
+ RTC_CHECK(parameters_.codec_settings);
+ VideoCodecSettings codec_settings = *parameters_.codec_settings;
webrtc::VideoEncoderConfig encoder_config =
CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
@@ -2202,9 +2082,8 @@
for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
info.add_ssrc(ssrc);
- VideoCodecSettings codec_settings;
- if (parameters_.codec_settings.Get(&codec_settings))
- info.codec_name = codec_settings.codec.name;
+ if (parameters_.codec_settings)
+ info.codec_name = parameters_.codec_settings->codec.name;
for (size_t i = 0; i < parameters_.encoder_config.streams.size(); ++i) {
if (i == parameters_.encoder_config.streams.size() - 1) {
info.preferred_bitrate +=
@@ -2238,6 +2117,15 @@
}
}
}
+
+ // Get bandwidth limitation info from stream_->GetStats().
+ // Input resolution (output from video_adapter) can be further scaled down or
+ // higher video layer(s) can be dropped due to bitrate constraints.
+ // Note, adapt_changes only include changes from the video_adapter.
+ if (stats.bw_limited_resolution)
+ info.adapt_reason |= CoordinatedVideoAdapter::ADAPTREASON_BANDWIDTH;
+
+ info.encoder_implementation_name = stats.encoder_implementation_name;
info.ssrc_groups = ssrc_groups_;
info.framerate_input = stats.input_frame_rate;
info.framerate_sent = stats.encode_frame_rate;
@@ -2316,11 +2204,10 @@
call_->DestroyVideoSendStream(stream_);
}
- VideoCodecSettings codec_settings;
- parameters_.codec_settings.Get(&codec_settings);
+ RTC_CHECK(parameters_.codec_settings);
parameters_.encoder_config.encoder_specific_settings =
ConfigureVideoEncoderSettings(
- codec_settings.codec, parameters_.options,
+ parameters_.codec_settings->codec, parameters_.options,
parameters_.encoder_config.content_type ==
webrtc::VideoEncoderConfig::ContentType::kScreen);
@@ -2345,7 +2232,8 @@
const webrtc::VideoReceiveStream::Config& config,
WebRtcVideoDecoderFactory* external_decoder_factory,
bool default_stream,
- const std::vector<VideoCodecSettings>& recv_codecs)
+ const std::vector<VideoCodecSettings>& recv_codecs,
+ bool disable_prerenderer_smoothing)
: call_(call),
ssrcs_(sp.ssrcs),
ssrc_groups_(sp.ssrc_groups),
@@ -2353,6 +2241,7 @@
default_stream_(default_stream),
config_(config),
external_decoder_factory_(external_decoder_factory),
+ disable_prerenderer_smoothing_(disable_prerenderer_smoothing),
renderer_(NULL),
last_width_(-1),
last_height_(-1),
@@ -2457,10 +2346,10 @@
config_.rtp.nack.rtp_history_ms =
HasNack(recv_codecs.begin()->codec) ? kNackHistoryMs : 0;
- ClearDecoders(&old_decoders);
LOG(LS_INFO) << "RecreateWebRtcStream (recv) because of SetRecvCodecs: "
<< CodecSettingsVectorToString(recv_codecs);
RecreateWebRtcStream();
+ ClearDecoders(&old_decoders);
}
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetLocalSsrc(
@@ -2482,20 +2371,28 @@
RecreateWebRtcStream();
}
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetNackAndRemb(
- bool nack_enabled, bool remb_enabled) {
+void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetFeedbackParameters(
+ bool nack_enabled,
+ bool remb_enabled,
+ bool transport_cc_enabled) {
int nack_history_ms = nack_enabled ? kNackHistoryMs : 0;
if (config_.rtp.nack.rtp_history_ms == nack_history_ms &&
- config_.rtp.remb == remb_enabled) {
- LOG(LS_INFO) << "Ignoring call to SetNackAndRemb because parameters are "
- "unchanged; nack=" << nack_enabled
- << ", remb=" << remb_enabled;
+ config_.rtp.remb == remb_enabled &&
+ config_.rtp.transport_cc == transport_cc_enabled) {
+ LOG(LS_INFO)
+ << "Ignoring call to SetFeedbackParameters because parameters are "
+ "unchanged; nack="
+ << nack_enabled << ", remb=" << remb_enabled
+ << ", transport_cc=" << transport_cc_enabled;
return;
}
config_.rtp.remb = remb_enabled;
config_.rtp.nack.rtp_history_ms = nack_history_ms;
- LOG(LS_INFO) << "RecreateWebRtcStream (recv) because of SetNackAndRemb; nack="
- << nack_enabled << ", remb=" << remb_enabled;
+ config_.rtp.transport_cc = transport_cc_enabled;
+ LOG(LS_INFO)
+ << "RecreateWebRtcStream (recv) because of SetFeedbackParameters; nack="
+ << nack_enabled << ", remb=" << remb_enabled
+ << ", transport_cc=" << transport_cc_enabled;
RecreateWebRtcStream();
}
@@ -2506,6 +2403,15 @@
RecreateWebRtcStream();
}
+void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvParameters(
+ const VideoRecvParameters& recv_params) {
+ config_.rtp.rtcp_mode = recv_params.rtcp.reduced_size
+ ? webrtc::RtcpMode::kReducedSize
+ : webrtc::RtcpMode::kCompound;
+ LOG(LS_INFO) << "RecreateWebRtcStream (recv) because of SetRecvParameters";
+ RecreateWebRtcStream();
+}
+
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() {
if (stream_ != NULL) {
call_->DestroyVideoReceiveStream(stream_);
@@ -2560,6 +2466,11 @@
return true;
}
+bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::SmoothsRenderedFrames()
+ const {
+ return disable_prerenderer_smoothing_;
+}
+
bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsDefaultStream() const {
return default_stream_;
}
@@ -2607,6 +2518,7 @@
info.ssrc_groups = ssrc_groups_;
info.add_ssrc(config_.rtp.remote_ssrc);
webrtc::VideoReceiveStream::Stats stats = stream_->GetStats();
+ info.decoder_implementation_name = stats.decoder_implementation_name;
info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes +
stats.rtp_stats.transmitted.header_bytes +
stats.rtp_stats.transmitted.padding_bytes;
diff --git a/talk/media/webrtc/webrtcvideoengine2.h b/talk/media/webrtc/webrtcvideoengine2.h
index 7096135..1b8da16 100644
--- a/talk/media/webrtc/webrtcvideoengine2.h
+++ b/talk/media/webrtc/webrtcvideoengine2.h
@@ -112,14 +112,11 @@
// Basic video engine implementation.
void Init();
- bool SetDefaultEncoderConfig(const VideoEncoderConfig& config);
-
WebRtcVideoChannel2* CreateChannel(webrtc::Call* call,
const VideoOptions& options);
const std::vector<VideoCodec>& codecs() const;
- const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
- void SetLogging(int min_sev, const char* filter);
+ RtpCapabilities GetCapabilities() const;
// Set a WebRtcVideoDecoderFactory for external decoding. Video engine does
// not take the ownership of |decoder_factory|. The caller needs to make sure
@@ -134,9 +131,6 @@
bool EnableTimedRender();
bool FindCodec(const VideoCodec& in);
- bool CanSendCodec(const VideoCodec& in,
- const VideoCodec& current,
- VideoCodec* out);
// Check whether the supplied trace should be ignored.
bool ShouldIgnoreTrace(const std::string& trace);
@@ -144,7 +138,6 @@
std::vector<VideoCodec> GetSupportedCodecs() const;
std::vector<VideoCodec> video_codecs_;
- std::vector<RtpHeaderExtension> rtp_header_extensions_;
bool initialized_;
@@ -250,14 +243,18 @@
WebRtcVideoEncoderFactory* external_encoder_factory,
const VideoOptions& options,
int max_bitrate_bps,
- const Settable<VideoCodecSettings>& codec_settings,
- const std::vector<webrtc::RtpExtension>& rtp_extensions);
+ const rtc::Optional<VideoCodecSettings>& codec_settings,
+ const std::vector<webrtc::RtpExtension>& rtp_extensions,
+ const VideoSendParameters& send_params);
~WebRtcVideoSendStream();
void SetOptions(const VideoOptions& options);
void SetCodec(const VideoCodecSettings& codec);
void SetRtpExtensions(
const std::vector<webrtc::RtpExtension>& rtp_extensions);
+ // TODO(deadbeef): Move logic from SetCodec/SetRtpExtensions/etc.
+ // into this method. Currently this method only sets the RTCP mode.
+ void SetSendParameters(const VideoSendParameters& send_params);
void InputFrame(VideoCapturer* capturer, const VideoFrame* frame);
bool SetCapturer(VideoCapturer* capturer);
@@ -286,11 +283,11 @@
const webrtc::VideoSendStream::Config& config,
const VideoOptions& options,
int max_bitrate_bps,
- const Settable<VideoCodecSettings>& codec_settings);
+ const rtc::Optional<VideoCodecSettings>& codec_settings);
webrtc::VideoSendStream::Config config;
VideoOptions options;
int max_bitrate_bps;
- Settable<VideoCodecSettings> codec_settings;
+ rtc::Optional<VideoCodecSettings> codec_settings;
// Sent resolutions + bitrates etc. by the underlying VideoSendStream,
// typically changes when setting a new resolution or reconfiguring
// bitrates.
@@ -395,19 +392,26 @@
const webrtc::VideoReceiveStream::Config& config,
WebRtcVideoDecoderFactory* external_decoder_factory,
bool default_stream,
- const std::vector<VideoCodecSettings>& recv_codecs);
+ const std::vector<VideoCodecSettings>& recv_codecs,
+ bool disable_prerenderer_smoothing);
~WebRtcVideoReceiveStream();
const std::vector<uint32_t>& GetSsrcs() const;
void SetLocalSsrc(uint32_t local_ssrc);
- void SetNackAndRemb(bool nack_enabled, bool remb_enabled);
+ void SetFeedbackParameters(bool nack_enabled,
+ bool remb_enabled,
+ bool transport_cc_enabled);
void SetRecvCodecs(const std::vector<VideoCodecSettings>& recv_codecs);
void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions);
+ // TODO(deadbeef): Move logic from SetRecvCodecs/SetRtpExtensions/etc.
+ // into this method. Currently this method only sets the RTCP mode.
+ void SetRecvParameters(const VideoRecvParameters& recv_params);
void RenderFrame(const webrtc::VideoFrame& frame,
int time_to_render_ms) override;
bool IsTextureSupported() const override;
+ bool SmoothsRenderedFrames() const override;
bool IsDefaultStream() const;
void SetRenderer(cricket::VideoRenderer* renderer);
@@ -448,6 +452,8 @@
WebRtcVideoDecoderFactory* const external_decoder_factory_;
std::vector<AllocatedDecoder> allocated_decoders_;
+ const bool disable_prerenderer_smoothing_;
+
rtc::CriticalSection renderer_lock_;
cricket::VideoRenderer* renderer_ GUARDED_BY(renderer_lock_);
int last_width_ GUARDED_BY(renderer_lock_);
@@ -512,7 +518,7 @@
std::set<uint32_t> send_ssrcs_ GUARDED_BY(stream_crit_);
std::set<uint32_t> receive_ssrcs_ GUARDED_BY(stream_crit_);
- Settable<VideoCodecSettings> send_codec_;
+ rtc::Optional<VideoCodecSettings> send_codec_;
std::vector<webrtc::RtpExtension> send_rtp_extensions_;
WebRtcVideoEncoderFactory* const external_encoder_factory_;
@@ -521,6 +527,10 @@
std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
webrtc::Call::Config::BitrateConfig bitrate_config_;
VideoOptions options_;
+ // TODO(deadbeef): Don't duplicate information between
+ // send_params/recv_params, rtp_extensions, options, etc.
+ VideoSendParameters send_params_;
+ VideoRecvParameters recv_params_;
};
} // namespace cricket
diff --git a/talk/media/webrtc/webrtcvideoengine2_unittest.cc b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
index c0cd2ff..41e04a9 100644
--- a/talk/media/webrtc/webrtcvideoengine2_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine2_unittest.cc
@@ -75,6 +75,8 @@
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamRemb, cricket::kParamValueEmpty)));
EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
+ cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty)));
+ EXPECT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir)));
}
@@ -205,26 +207,6 @@
EXPECT_TRUE(engine_.FindCodec(rtx));
}
-TEST_F(WebRtcVideoEngine2Test, SetDefaultEncoderConfigPreservesFeedbackParams) {
- cricket::VideoCodec max_settings(
- engine_.codecs()[0].id, engine_.codecs()[0].name,
- engine_.codecs()[0].width / 2, engine_.codecs()[0].height / 2, 30, 0);
- // This codec shouldn't have NACK by default or the test is pointless.
- EXPECT_FALSE(max_settings.HasFeedbackParam(
- FeedbackParam(kRtcpFbParamNack, kParamValueEmpty)));
- // The engine should by default have it however.
- EXPECT_TRUE(engine_.codecs()[0].HasFeedbackParam(
- FeedbackParam(kRtcpFbParamNack, kParamValueEmpty)));
-
- // Set constrained max codec settings.
- EXPECT_TRUE(engine_.SetDefaultEncoderConfig(
- cricket::VideoEncoderConfig(max_settings)));
-
- // Verify that feedback parameters are retained.
- EXPECT_TRUE(engine_.codecs()[0].HasFeedbackParam(
- FeedbackParam(kRtcpFbParamNack, kParamValueEmpty)));
-}
-
TEST_F(WebRtcVideoEngine2Test, DefaultRtxCodecHasAssociatedPayloadTypeSet) {
std::vector<VideoCodec> engine_codecs = engine_.codecs();
for (size_t i = 0; i < engine_codecs.size(); ++i) {
@@ -240,11 +222,11 @@
}
TEST_F(WebRtcVideoEngine2Test, SupportsTimestampOffsetHeaderExtension) {
- std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
- ASSERT_FALSE(extensions.empty());
- for (size_t i = 0; i < extensions.size(); ++i) {
- if (extensions[i].uri == kRtpTimestampOffsetHeaderExtension) {
- EXPECT_EQ(kRtpTimestampOffsetHeaderExtensionDefaultId, extensions[i].id);
+ RtpCapabilities capabilities = engine_.GetCapabilities();
+ ASSERT_FALSE(capabilities.header_extensions.empty());
+ for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
+ if (extension.uri == kRtpTimestampOffsetHeaderExtension) {
+ EXPECT_EQ(kRtpTimestampOffsetHeaderExtensionDefaultId, extension.id);
return;
}
}
@@ -252,12 +234,11 @@
}
TEST_F(WebRtcVideoEngine2Test, SupportsAbsoluteSenderTimeHeaderExtension) {
- std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
- ASSERT_FALSE(extensions.empty());
- for (size_t i = 0; i < extensions.size(); ++i) {
- if (extensions[i].uri == kRtpAbsoluteSenderTimeHeaderExtension) {
- EXPECT_EQ(kRtpAbsoluteSenderTimeHeaderExtensionDefaultId,
- extensions[i].id);
+ RtpCapabilities capabilities = engine_.GetCapabilities();
+ ASSERT_FALSE(capabilities.header_extensions.empty());
+ for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
+ if (extension.uri == kRtpAbsoluteSenderTimeHeaderExtension) {
+ EXPECT_EQ(kRtpAbsoluteSenderTimeHeaderExtensionDefaultId, extension.id);
return;
}
}
@@ -272,12 +253,12 @@
TEST_F(WebRtcVideoEngine2WithSendSideBweTest,
SupportsTransportSequenceNumberHeaderExtension) {
- std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
- ASSERT_FALSE(extensions.empty());
- for (size_t i = 0; i < extensions.size(); ++i) {
- if (extensions[i].uri == kRtpTransportSequenceNumberHeaderExtension) {
+ RtpCapabilities capabilities = engine_.GetCapabilities();
+ ASSERT_FALSE(capabilities.header_extensions.empty());
+ for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
+ if (extension.uri == kRtpTransportSequenceNumberHeaderExtension) {
EXPECT_EQ(kRtpTransportSequenceNumberHeaderExtensionDefaultId,
- extensions[i].id);
+ extension.id);
return;
}
}
@@ -285,11 +266,11 @@
}
TEST_F(WebRtcVideoEngine2Test, SupportsVideoRotationHeaderExtension) {
- std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
- ASSERT_FALSE(extensions.empty());
- for (size_t i = 0; i < extensions.size(); ++i) {
- if (extensions[i].uri == kRtpVideoRotationHeaderExtension) {
- EXPECT_EQ(kRtpVideoRotationHeaderExtensionDefaultId, extensions[i].id);
+ RtpCapabilities capabilities = engine_.GetCapabilities();
+ ASSERT_FALSE(capabilities.header_extensions.empty());
+ for (const RtpHeaderExtension& extension : capabilities.header_extensions) {
+ if (extension.uri == kRtpVideoRotationHeaderExtension) {
+ EXPECT_EQ(kRtpVideoRotationHeaderExtensionDefaultId, extension.id);
return;
}
}
@@ -794,17 +775,6 @@
ASSERT_EQ(1u, decoder_factory.decoders().size());
}
-class WebRtcVideoEngine2BaseTest
- : public VideoEngineTest<cricket::WebRtcVideoEngine2> {
- protected:
- typedef VideoEngineTest<cricket::WebRtcVideoEngine2> Base;
-};
-
-#define WEBRTC_ENGINE_BASE_TEST(test) \
- TEST_F(WebRtcVideoEngine2BaseTest, test) { Base::test##Body(); }
-
-WEBRTC_ENGINE_BASE_TEST(ConstrainNewCodec2);
-
class WebRtcVideoChannel2BaseTest
: public VideoMediaChannelTest<WebRtcVideoEngine2, WebRtcVideoChannel2> {
protected:
@@ -894,7 +864,10 @@
Base::TwoStreamsReUseFirstStream(kVp8Codec);
}
+//Disabled for TSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=4963
+#if !defined(THREAD_SANITIZER)
WEBRTC_BASE_TEST(SendManyResizeOnce);
+#endif // THREAD_SANITIZER
// TODO(pbos): Enable and figure out why this fails (or should work).
TEST_F(WebRtcVideoChannel2BaseTest, DISABLED_SendVp8HdAndReceiveAdaptedVp8Vga) {
@@ -1097,7 +1070,7 @@
FakeVideoSendStream* SetDenoisingOption(
const cricket::VideoSendParameters& parameters, bool enabled) {
cricket::VideoSendParameters params = parameters;
- params.options.video_noise_reduction.Set(enabled);
+ params.options.video_noise_reduction = rtc::Optional<bool>(enabled);
channel_->SetSendParameters(params);
return fake_call_->GetVideoSendStreams().back();
}
@@ -1148,7 +1121,7 @@
parameters.codecs = engine_.codecs();
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_TRUE(channel_->SetSend(true));
- parameters.options.conference_mode.Set(true);
+ parameters.options.conference_mode = rtc::Optional<bool>(true);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
// Send side.
@@ -1451,6 +1424,11 @@
EXPECT_TRUE(stream->GetConfig().rtp.remb);
}
+TEST_F(WebRtcVideoChannel2Test, TransportCcIsEnabledByDefault) {
+ FakeVideoReceiveStream* stream = AddRecvStream();
+ EXPECT_TRUE(stream->GetConfig().rtp.transport_cc);
+}
+
TEST_F(WebRtcVideoChannel2Test, RembCanBeEnabledAndDisabled) {
FakeVideoReceiveStream* stream = AddRecvStream();
EXPECT_TRUE(stream->GetConfig().rtp.remb);
@@ -1471,6 +1449,27 @@
EXPECT_TRUE(stream->GetConfig().rtp.remb);
}
+TEST_F(WebRtcVideoChannel2Test, TransportCcCanBeEnabledAndDisabled) {
+ FakeVideoReceiveStream* stream = AddRecvStream();
+ EXPECT_TRUE(stream->GetConfig().rtp.transport_cc);
+
+ // Verify that transport cc feedback is turned off when send(!) codecs without
+ // transport cc feedback are set.
+ cricket::VideoSendParameters parameters;
+ parameters.codecs.push_back(kVp8Codec);
+ EXPECT_TRUE(parameters.codecs[0].feedback_params.params().empty());
+ EXPECT_TRUE(channel_->SetSendParameters(parameters));
+ stream = fake_call_->GetVideoReceiveStreams()[0];
+ EXPECT_FALSE(stream->GetConfig().rtp.transport_cc);
+
+ // Verify that transport cc feedback is turned on when setting default codecs
+ // since the default codecs have transport cc feedback enabled.
+ parameters.codecs = engine_.codecs();
+ EXPECT_TRUE(channel_->SetSendParameters(parameters));
+ stream = fake_call_->GetVideoReceiveStreams()[0];
+ EXPECT_TRUE(stream->GetConfig().rtp.transport_cc);
+}
+
TEST_F(WebRtcVideoChannel2Test, NackIsEnabledByDefault) {
VerifyCodecHasDefaultFeedbackParams(default_codec_);
@@ -1558,7 +1557,8 @@
cricket::VideoCodec codec = kVp8Codec360p;
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
- parameters.options.screencast_min_bitrate.Set(kScreenshareMinBitrateKbps);
+ parameters.options.screencast_min_bitrate =
+ rtc::Optional<int>(kScreenshareMinBitrateKbps);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
AddSendStream();
@@ -1612,7 +1612,7 @@
ConferenceModeScreencastConfiguresTemporalLayer) {
static const int kConferenceScreencastTemporalBitrateBps =
ScreenshareLayerConfig::GetDefault().tl0_bitrate_kbps * 1000;
- send_parameters_.options.conference_mode.Set(true);
+ send_parameters_.options.conference_mode = rtc::Optional<bool>(true);
channel_->SetSendParameters(send_parameters_);
AddSendStream();
@@ -1659,13 +1659,15 @@
}
TEST_F(WebRtcVideoChannel2Test, SetOptionsWithSuspendBelowMinBitrate) {
- send_parameters_.options.suspend_below_min_bitrate.Set(true);
+ send_parameters_.options.suspend_below_min_bitrate =
+ rtc::Optional<bool>(true);
channel_->SetSendParameters(send_parameters_);
FakeVideoSendStream* stream = AddSendStream();
EXPECT_TRUE(stream->GetConfig().suspend_below_min_bitrate);
- send_parameters_.options.suspend_below_min_bitrate.Set(false);
+ send_parameters_.options.suspend_below_min_bitrate =
+ rtc::Optional<bool>(false);
channel_->SetSendParameters(send_parameters_);
stream = fake_call_->GetVideoSendStreams()[0];
@@ -1853,7 +1855,7 @@
cricket::VideoSendParameters parameters;
parameters.codecs.push_back(codec);
if (!enable_overuse) {
- parameters.options.cpu_overuse_detection.Set(false);
+ parameters.options.cpu_overuse_detection = rtc::Optional<bool>(false);
}
EXPECT_TRUE(channel_->SetSendParameters(parameters));
@@ -2375,19 +2377,55 @@
cricket::VideoSendParameters parameters = send_parameters_;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_EQ(rtc::DSCP_NO_CHANGE, network_interface->dscp());
- parameters.options.dscp.Set(true);
+ parameters.options.dscp = rtc::Optional<bool>(true);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
EXPECT_EQ(rtc::DSCP_AF41, network_interface->dscp());
// Verify previous value is not modified if dscp option is not set.
cricket::VideoSendParameters parameters1 = send_parameters_;
EXPECT_TRUE(channel_->SetSendParameters(parameters1));
EXPECT_EQ(rtc::DSCP_AF41, network_interface->dscp());
- parameters1.options.dscp.Set(false);
+ parameters1.options.dscp = rtc::Optional<bool>(false);
EXPECT_TRUE(channel_->SetSendParameters(parameters1));
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
channel_->SetInterface(NULL);
}
+// This test verifies that the RTCP reduced size mode is properly applied to
+// send video streams.
+TEST_F(WebRtcVideoChannel2Test, TestSetSendRtcpReducedSize) {
+ // Create stream, expecting that default mode is "compound".
+ FakeVideoSendStream* stream1 = AddSendStream();
+ EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode);
+
+ // Now enable reduced size mode.
+ send_parameters_.rtcp.reduced_size = true;
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+ stream1 = fake_call_->GetVideoSendStreams()[0];
+ EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode);
+
+ // Create a new stream and ensure it picks up the reduced size mode.
+ FakeVideoSendStream* stream2 = AddSendStream();
+ EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream2->GetConfig().rtp.rtcp_mode);
+}
+
+// This test verifies that the RTCP reduced size mode is properly applied to
+// receive video streams.
+TEST_F(WebRtcVideoChannel2Test, TestSetRecvRtcpReducedSize) {
+ // Create stream, expecting that default mode is "compound".
+ FakeVideoReceiveStream* stream1 = AddRecvStream();
+ EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode);
+
+ // Now enable reduced size mode.
+ recv_parameters_.rtcp.reduced_size = true;
+ EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
+ stream1 = fake_call_->GetVideoReceiveStreams()[0];
+ EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode);
+
+ // Create a new stream and ensure it picks up the reduced size mode.
+ FakeVideoReceiveStream* stream2 = AddRecvStream();
+ EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream2->GetConfig().rtp.rtcp_mode);
+}
+
TEST_F(WebRtcVideoChannel2Test, OnReadyToSendSignalsNetworkState) {
EXPECT_EQ(webrtc::kNetworkUp, fake_call_->GetNetworkState());
@@ -2410,6 +2448,18 @@
EXPECT_EQ(kVp8Codec.name, info.senders[0].codec_name);
}
+TEST_F(WebRtcVideoChannel2Test, GetStatsReportsEncoderImplementationName) {
+ FakeVideoSendStream* stream = AddSendStream();
+ webrtc::VideoSendStream::Stats stats;
+ stats.encoder_implementation_name = "encoder_implementation_name";
+ stream->SetStats(stats);
+
+ cricket::VideoMediaInfo info;
+ ASSERT_TRUE(channel_->GetStats(&info));
+ EXPECT_EQ(stats.encoder_implementation_name,
+ info.senders[0].encoder_implementation_name);
+}
+
TEST_F(WebRtcVideoChannel2Test, GetStatsReportsCpuOveruseMetrics) {
FakeVideoSendStream* stream = AddSendStream();
webrtc::VideoSendStream::Stats stats;
@@ -2460,7 +2510,7 @@
EXPECT_TRUE(channel_->SetSend(true));
// Verify that the CpuOveruseObserver is registered and trigger downgrade.
- parameters.options.cpu_overuse_detection.Set(true);
+ parameters.options.cpu_overuse_detection = rtc::Optional<bool>(true);
EXPECT_TRUE(channel_->SetSendParameters(parameters));
// Trigger overuse.
@@ -2518,6 +2568,87 @@
EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], NULL));
}
+TEST_F(WebRtcVideoChannel2Test, GetStatsTracksAdaptationAndBandwidthStats) {
+ AddSendStream(cricket::CreateSimStreamParams("cname", MAKE_VECTOR(kSsrcs3)));
+
+ // Capture format VGA.
+ cricket::FakeVideoCapturer video_capturer_vga;
+ const std::vector<cricket::VideoFormat>* formats =
+ video_capturer_vga.GetSupportedFormats();
+ cricket::VideoFormat capture_format_vga = (*formats)[1];
+ EXPECT_EQ(cricket::CS_RUNNING, video_capturer_vga.Start(capture_format_vga));
+ EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], &video_capturer_vga));
+ EXPECT_TRUE(video_capturer_vga.CaptureFrame());
+
+ cricket::VideoCodec send_codec(100, "VP8", 640, 480, 30, 0);
+ cricket::VideoSendParameters parameters;
+ parameters.codecs.push_back(send_codec);
+ EXPECT_TRUE(channel_->SetSendParameters(parameters));
+ EXPECT_TRUE(channel_->SetSend(true));
+
+ // Verify that the CpuOveruseObserver is registered and trigger downgrade.
+ parameters.options.cpu_overuse_detection = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(parameters));
+
+ // Trigger overuse -> adapt CPU.
+ ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
+ webrtc::LoadObserver* overuse_callback =
+ fake_call_->GetVideoSendStreams().front()->GetConfig().overuse_callback;
+ ASSERT_TRUE(overuse_callback != NULL);
+ overuse_callback->OnLoadUpdate(webrtc::LoadObserver::kOveruse);
+ EXPECT_TRUE(video_capturer_vga.CaptureFrame());
+ cricket::VideoMediaInfo info;
+ EXPECT_TRUE(channel_->GetStats(&info));
+ ASSERT_EQ(1U, info.senders.size());
+ EXPECT_EQ(CoordinatedVideoAdapter::ADAPTREASON_CPU,
+ info.senders[0].adapt_reason);
+
+ // Set bandwidth limitation stats for the stream -> adapt CPU + BW.
+ webrtc::VideoSendStream::Stats stats;
+ stats.bw_limited_resolution = true;
+ fake_call_->GetVideoSendStreams().front()->SetStats(stats);
+ info.Clear();
+ EXPECT_TRUE(channel_->GetStats(&info));
+ ASSERT_EQ(1U, info.senders.size());
+ EXPECT_EQ(CoordinatedVideoAdapter::ADAPTREASON_CPU +
+ CoordinatedVideoAdapter::ADAPTREASON_BANDWIDTH,
+ info.senders[0].adapt_reason);
+
+ // Trigger upgrade -> adapt BW.
+ overuse_callback->OnLoadUpdate(webrtc::LoadObserver::kUnderuse);
+ EXPECT_TRUE(video_capturer_vga.CaptureFrame());
+ info.Clear();
+ EXPECT_TRUE(channel_->GetStats(&info));
+ ASSERT_EQ(1U, info.senders.size());
+ EXPECT_EQ(CoordinatedVideoAdapter::ADAPTREASON_BANDWIDTH,
+ info.senders[0].adapt_reason);
+
+ // Reset bandwidth limitation state -> adapt NONE.
+ stats.bw_limited_resolution = false;
+ fake_call_->GetVideoSendStreams().front()->SetStats(stats);
+ info.Clear();
+ EXPECT_TRUE(channel_->GetStats(&info));
+ ASSERT_EQ(1U, info.senders.size());
+ EXPECT_EQ(CoordinatedVideoAdapter::ADAPTREASON_NONE,
+ info.senders[0].adapt_reason);
+
+ EXPECT_TRUE(channel_->SetCapturer(kSsrcs3[0], NULL));
+}
+
+TEST_F(WebRtcVideoChannel2Test,
+ GetStatsTranslatesBandwidthLimitedResolutionCorrectly) {
+ FakeVideoSendStream* stream = AddSendStream();
+ webrtc::VideoSendStream::Stats stats;
+ stats.bw_limited_resolution = true;
+ stream->SetStats(stats);
+
+ cricket::VideoMediaInfo info;
+ EXPECT_TRUE(channel_->GetStats(&info));
+ ASSERT_EQ(1U, info.senders.size());
+ EXPECT_EQ(CoordinatedVideoAdapter::ADAPTREASON_BANDWIDTH,
+ info.senders[0].adapt_reason);
+}
+
TEST_F(WebRtcVideoChannel2Test,
GetStatsTranslatesSendRtcpPacketTypesCorrectly) {
FakeVideoSendStream* stream = AddSendStream();
@@ -2561,6 +2692,7 @@
TEST_F(WebRtcVideoChannel2Test, GetStatsTranslatesDecodeStatsCorrectly) {
FakeVideoReceiveStream* stream = AddRecvStream();
webrtc::VideoReceiveStream::Stats stats;
+ stats.decoder_implementation_name = "decoder_implementation_name";
stats.decode_ms = 2;
stats.max_decode_ms = 3;
stats.current_delay_ms = 4;
@@ -2572,6 +2704,8 @@
cricket::VideoMediaInfo info;
ASSERT_TRUE(channel_->GetStats(&info));
+ EXPECT_EQ(stats.decoder_implementation_name,
+ info.receivers[0].decoder_implementation_name);
EXPECT_EQ(stats.decode_ms, info.receivers[0].decode_ms);
EXPECT_EQ(stats.max_decode_ms, info.receivers[0].max_decode_ms);
EXPECT_EQ(stats.current_delay_ms, info.receivers[0].current_delay_ms);
diff --git a/talk/media/webrtc/webrtcvideoframe.cc b/talk/media/webrtc/webrtcvideoframe.cc
index 7da7e3b..fcc991c 100644
--- a/talk/media/webrtc/webrtcvideoframe.cc
+++ b/talk/media/webrtc/webrtcvideoframe.cc
@@ -56,17 +56,6 @@
rotation_(rotation) {
}
-WebRtcVideoFrame::WebRtcVideoFrame(
- const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
- int64_t elapsed_time_ns,
- int64_t time_stamp_ns)
- : video_frame_buffer_(buffer),
- pixel_width_(1),
- pixel_height_(1),
- time_stamp_ns_(time_stamp_ns),
- rotation_(webrtc::kVideoRotation_0) {
-}
-
WebRtcVideoFrame::~WebRtcVideoFrame() {}
bool WebRtcVideoFrame::Init(uint32_t format,
@@ -90,13 +79,7 @@
return Reset(frame->fourcc, frame->width, frame->height, dw, dh,
static_cast<uint8_t*>(frame->data), frame->data_size,
frame->pixel_width, frame->pixel_height, frame->time_stamp,
- frame->GetRotation(), apply_rotation);
-}
-
-bool WebRtcVideoFrame::InitToBlack(int w, int h, size_t pixel_width,
- size_t pixel_height, int64_t,
- int64_t time_stamp_ns) {
- return InitToBlack(w, h, pixel_width, pixel_height, time_stamp_ns);
+ frame->rotation, apply_rotation);
}
bool WebRtcVideoFrame::InitToBlack(int w, int h, size_t pixel_width,
diff --git a/talk/media/webrtc/webrtcvideoframe.h b/talk/media/webrtc/webrtcvideoframe.h
index 0928c59..827cf28 100644
--- a/talk/media/webrtc/webrtcvideoframe.h
+++ b/talk/media/webrtc/webrtcvideoframe.h
@@ -33,7 +33,7 @@
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/common_types.h"
-#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
namespace cricket {
@@ -46,11 +46,6 @@
int64_t time_stamp_ns,
webrtc::VideoRotation rotation);
- // TODO(guoweis): Remove this when chrome code base is updated.
- WebRtcVideoFrame(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& buffer,
- int64_t elapsed_time_ns,
- int64_t time_stamp_ns);
-
~WebRtcVideoFrame();
// Creates a frame from a raw sample with FourCC "format" and size "w" x "h".
@@ -74,10 +69,6 @@
void InitToEmptyBuffer(int w, int h, size_t pixel_width, size_t pixel_height,
int64_t time_stamp_ns);
- // TODO(magjed): Remove once Chromium is updated.
- bool InitToBlack(int w, int h, size_t pixel_width, size_t pixel_height,
- int64_t elapsed_time_ns, int64_t time_stamp_ns);
-
bool InitToBlack(int w, int h, size_t pixel_width, size_t pixel_height,
int64_t time_stamp_ns) override;
diff --git a/talk/media/webrtc/webrtcvoe.h b/talk/media/webrtc/webrtcvoe.h
index db6a64a..aa705a0 100644
--- a/talk/media/webrtc/webrtcvoe.h
+++ b/talk/media/webrtc/webrtcvoe.h
@@ -36,7 +36,6 @@
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_dtmf.h"
#include "webrtc/voice_engine/include/voe_errors.h"
#include "webrtc/voice_engine/include/voe_hardware.h"
#include "webrtc/voice_engine/include/voe_network.h"
@@ -91,14 +90,13 @@
public:
VoEWrapper()
: engine_(webrtc::VoiceEngine::Create()), processing_(engine_),
- base_(engine_), codec_(engine_), dtmf_(engine_),
+ base_(engine_), codec_(engine_),
hw_(engine_), network_(engine_),
rtp_(engine_), volume_(engine_) {
}
VoEWrapper(webrtc::VoEAudioProcessing* processing,
webrtc::VoEBase* base,
webrtc::VoECodec* codec,
- webrtc::VoEDtmf* dtmf,
webrtc::VoEHardware* hw,
webrtc::VoENetwork* network,
webrtc::VoERTP_RTCP* rtp,
@@ -107,7 +105,6 @@
processing_(processing),
base_(base),
codec_(codec),
- dtmf_(dtmf),
hw_(hw),
network_(network),
rtp_(rtp),
@@ -118,7 +115,6 @@
webrtc::VoEAudioProcessing* processing() const { return processing_.get(); }
webrtc::VoEBase* base() const { return base_.get(); }
webrtc::VoECodec* codec() const { return codec_.get(); }
- webrtc::VoEDtmf* dtmf() const { return dtmf_.get(); }
webrtc::VoEHardware* hw() const { return hw_.get(); }
webrtc::VoENetwork* network() const { return network_.get(); }
webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); }
@@ -130,29 +126,11 @@
scoped_voe_ptr<webrtc::VoEAudioProcessing> processing_;
scoped_voe_ptr<webrtc::VoEBase> base_;
scoped_voe_ptr<webrtc::VoECodec> codec_;
- scoped_voe_ptr<webrtc::VoEDtmf> dtmf_;
scoped_voe_ptr<webrtc::VoEHardware> hw_;
scoped_voe_ptr<webrtc::VoENetwork> network_;
scoped_voe_ptr<webrtc::VoERTP_RTCP> rtp_;
scoped_voe_ptr<webrtc::VoEVolumeControl> volume_;
};
-
-// Adds indirection to static WebRtc functions, allowing them to be mocked.
-class VoETraceWrapper {
- public:
- virtual ~VoETraceWrapper() {}
-
- virtual int SetTraceFilter(const unsigned int filter) {
- return webrtc::VoiceEngine::SetTraceFilter(filter);
- }
- virtual int SetTraceFile(const char* fileNameUTF8) {
- return webrtc::VoiceEngine::SetTraceFile(fileNameUTF8);
- }
- virtual int SetTraceCallback(webrtc::TraceCallback* callback) {
- return webrtc::VoiceEngine::SetTraceCallback(callback);
- }
-};
-
} // namespace cricket
#endif // TALK_MEDIA_WEBRTCVOE_H_
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 27ca1de..9192b72 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -42,7 +42,10 @@
#include "talk/media/base/audiorenderer.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/streamparams.h"
+#include "talk/media/webrtc/webrtcmediaengine.h"
#include "talk/media/webrtc/webrtcvoe.h"
+#include "webrtc/audio/audio_sink.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/common.h"
@@ -52,53 +55,26 @@
#include "webrtc/base/stringutils.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/include/field_trial.h"
+#include "webrtc/system_wrappers/include/trace.h"
namespace cricket {
namespace {
-const int kMaxNumPacketSize = 6;
-struct CodecPref {
- const char* name;
- int clockrate;
- int channels;
- int payload_type;
- bool is_multi_rate;
- int packet_sizes_ms[kMaxNumPacketSize];
-};
-// Note: keep the supported packet sizes in ascending order.
-const CodecPref kCodecPrefs[] = {
- { kOpusCodecName, 48000, 2, 111, true, { 10, 20, 40, 60 } },
- { kIsacCodecName, 16000, 1, 103, true, { 30, 60 } },
- { kIsacCodecName, 32000, 1, 104, true, { 30 } },
- // G722 should be advertised as 8000 Hz because of the RFC "bug".
- { kG722CodecName, 8000, 1, 9, false, { 10, 20, 30, 40, 50, 60 } },
- { kIlbcCodecName, 8000, 1, 102, false, { 20, 30, 40, 60 } },
- { kPcmuCodecName, 8000, 1, 0, false, { 10, 20, 30, 40, 50, 60 } },
- { kPcmaCodecName, 8000, 1, 8, false, { 10, 20, 30, 40, 50, 60 } },
- { kCnCodecName, 32000, 1, 106, false, { } },
- { kCnCodecName, 16000, 1, 105, false, { } },
- { kCnCodecName, 8000, 1, 13, false, { } },
- { kRedCodecName, 8000, 1, 127, false, { } },
- { kDtmfCodecName, 8000, 1, 126, false, { } },
-};
+const int kDefaultTraceFilter = webrtc::kTraceNone | webrtc::kTraceTerseInfo |
+ webrtc::kTraceWarning | webrtc::kTraceError |
+ webrtc::kTraceCritical;
+const int kElevatedTraceFilter = kDefaultTraceFilter | webrtc::kTraceStateInfo |
+ webrtc::kTraceInfo;
-// For Linux/Mac, using the default device is done by specifying index 0 for
-// VoE 4.0 and not -1 (which was the case for VoE 3.5).
-//
// On Windows Vista and newer, Microsoft introduced the concept of "Default
// Communications Device". This means that there are two types of default
// devices (old Wave Audio style default and Default Communications Device).
//
// On Windows systems which only support Wave Audio style default, uses either
// -1 or 0 to select the default device.
-//
-// On Windows systems which support both "Default Communication Device" and
-// old Wave Audio style default, use -1 for Default Communications Device and
-// -2 for Wave Audio style default, which is what we want to use for clips.
-// It's not clear yet whether the -2 index is handled properly on other OSes.
-
#ifdef WIN32
const int kDefaultAudioDeviceId = -1;
#else
@@ -150,6 +126,12 @@
const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
#endif
+// Constants from voice_engine_defines.h.
+const int kMinTelephoneEventCode = 0; // RFC4733 (Section 2.3.1)
+const int kMaxTelephoneEventCode = 255;
+const int kMinTelephoneEventDuration = 100;
+const int kMaxTelephoneEventDuration = 60000; // Actual limit is 2^16
+
bool ValidateStreamParams(const StreamParams& sp) {
if (sp.ssrcs.empty()) {
LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
@@ -177,32 +159,6 @@
return ss.str();
}
-void LogMultiline(rtc::LoggingSeverity sev, char* text) {
- const char* delim = "\r\n";
- for (char* tok = strtok(text, delim); tok; tok = strtok(NULL, delim)) {
- LOG_V(sev) << tok;
- }
-}
-
-// Severity is an integer because it comes is assumed to be from command line.
-int SeverityToFilter(int severity) {
- int filter = webrtc::kTraceNone;
- switch (severity) {
- case rtc::LS_VERBOSE:
- filter |= webrtc::kTraceAll;
- FALLTHROUGH();
- case rtc::LS_INFO:
- filter |= (webrtc::kTraceStateInfo | webrtc::kTraceInfo);
- FALLTHROUGH();
- case rtc::LS_WARNING:
- filter |= (webrtc::kTraceTerseInfo | webrtc::kTraceWarning);
- FALLTHROUGH();
- case rtc::LS_ERROR:
- filter |= (webrtc::kTraceError | webrtc::kTraceCritical);
- }
- return filter;
-}
-
bool IsCodec(const AudioCodec& codec, const char* ref_name) {
return (_stricmp(codec.name.c_str(), ref_name) == 0);
}
@@ -211,19 +167,9 @@
return (_stricmp(codec.plname, ref_name) == 0);
}
-bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
- for (size_t i = 0; i < ARRAY_SIZE(kCodecPrefs); ++i) {
- if (IsCodec(codec, kCodecPrefs[i].name) &&
- kCodecPrefs[i].clockrate == codec.plfreq) {
- return kCodecPrefs[i].is_multi_rate;
- }
- }
- return false;
-}
-
bool FindCodec(const std::vector<AudioCodec>& codecs,
- const AudioCodec& codec,
- AudioCodec* found_codec) {
+ const AudioCodec& codec,
+ AudioCodec* found_codec) {
for (const AudioCodec& c : codecs) {
if (c.Matches(codec)) {
if (found_codec != NULL) {
@@ -253,38 +199,8 @@
kParamValueEmpty));
}
-int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
- int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
- for (int packet_size_ms : codec_pref.packet_sizes_ms) {
- if (packet_size_ms && packet_size_ms <= ptime_ms) {
- selected_packet_size_ms = packet_size_ms;
- }
- }
- return selected_packet_size_ms;
-}
-
-// If the AudioCodec param kCodecParamPTime is set, then we will set it to codec
-// pacsize if it's valid, or we will pick the next smallest value we support.
-// TODO(Brave): Query supported packet sizes from ACM when the API is ready.
-bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
- for (const CodecPref& codec_pref : kCodecPrefs) {
- if ((IsCodec(*codec, codec_pref.name) &&
- codec_pref.clockrate == codec->plfreq) ||
- IsCodec(*codec, kG722CodecName)) {
- int packet_size_ms = SelectPacketSize(codec_pref, ptime_ms);
- if (packet_size_ms) {
- // Convert unit from milli-seconds to samples.
- codec->pacsize = (codec->plfreq / 1000) * packet_size_ms;
- return true;
- }
- }
- }
- return false;
-}
-
// Return true if codec.params[feature] == "1", false otherwise.
-bool IsCodecFeatureEnabled(const AudioCodec& codec,
- const char* feature) {
+bool IsCodecFeatureEnabled(const AudioCodec& codec, const char* feature) {
int value;
return codec.GetParam(feature, &value) && value == 1;
}
@@ -351,109 +267,29 @@
voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
}
-// Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
-// which says that G722 should be advertised as 8 kHz although it is a 16 kHz
-// codec.
-void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
- if (IsCodec(*voe_codec, kG722CodecName)) {
- // If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
- // has changed, and this special case is no longer needed.
- RTC_DCHECK(voe_codec->plfreq != new_plfreq);
- voe_codec->plfreq = new_plfreq;
- }
+webrtc::AudioState::Config MakeAudioStateConfig(VoEWrapper* voe_wrapper) {
+ webrtc::AudioState::Config config;
+ config.voice_engine = voe_wrapper->engine();
+ return config;
}
-// Gets the default set of options applied to the engine. Historically, these
-// were supplied as a combination of flags from the channel manager (ec, agc,
-// ns, and highpass) and the rest hardcoded in InitInternal.
-AudioOptions GetDefaultEngineOptions() {
- AudioOptions options;
- options.echo_cancellation.Set(true);
- options.auto_gain_control.Set(true);
- options.noise_suppression.Set(true);
- options.highpass_filter.Set(true);
- options.stereo_swapping.Set(false);
- options.audio_jitter_buffer_max_packets.Set(50);
- options.audio_jitter_buffer_fast_accelerate.Set(false);
- options.typing_detection.Set(true);
- options.adjust_agc_delta.Set(0);
- options.experimental_agc.Set(false);
- options.extended_filter_aec.Set(false);
- options.delay_agnostic_aec.Set(false);
- options.experimental_ns.Set(false);
- options.aec_dump.Set(false);
- return options;
-}
-
-std::string GetEnableString(bool enable) {
- return enable ? "enable" : "disable";
-}
-} // namespace {
-
-WebRtcVoiceEngine::WebRtcVoiceEngine()
- : voe_wrapper_(new VoEWrapper()),
- tracing_(new VoETraceWrapper()),
- adm_(NULL),
- log_filter_(SeverityToFilter(kDefaultLogSeverity)),
- is_dumping_aec_(false) {
- Construct();
-}
-
-WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
- VoETraceWrapper* tracing)
- : voe_wrapper_(voe_wrapper),
- tracing_(tracing),
- adm_(NULL),
- log_filter_(SeverityToFilter(kDefaultLogSeverity)),
- is_dumping_aec_(false) {
- Construct();
-}
-
-void WebRtcVoiceEngine::Construct() {
- SetTraceFilter(log_filter_);
- initialized_ = false;
- LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
- SetTraceOptions("");
- if (tracing_->SetTraceCallback(this) == -1) {
- LOG_RTCERR0(SetTraceCallback);
- }
- if (voe_wrapper_->base()->RegisterVoiceEngineObserver(*this) == -1) {
- LOG_RTCERR0(RegisterVoiceEngineObserver);
- }
- // Clear the default agc state.
- memset(&default_agc_config_, 0, sizeof(default_agc_config_));
-
- // Load our audio codec list.
- ConstructCodecs();
-
- // Load our RTP Header extensions.
- rtp_header_extensions_.push_back(
- RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
- kRtpAudioLevelHeaderExtensionDefaultId));
- rtp_header_extensions_.push_back(
- RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
- kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
- if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
- rtp_header_extensions_.push_back(RtpHeaderExtension(
- kRtpTransportSequenceNumberHeaderExtension,
- kRtpTransportSequenceNumberHeaderExtensionDefaultId));
- }
- options_ = GetDefaultEngineOptions();
-}
-
-void WebRtcVoiceEngine::ConstructCodecs() {
- LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
- int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
- for (int i = 0; i < ncodecs; ++i) {
- webrtc::CodecInst voe_codec;
- if (GetVoeCodec(i, &voe_codec)) {
+class WebRtcVoiceCodecs final {
+ public:
+ // TODO(solenberg): Do this filtering once off-line, add a simple AudioCodec
+ // list and add a test which verifies VoE supports the listed codecs.
+ static std::vector<AudioCodec> SupportedCodecs() {
+ LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
+ std::vector<AudioCodec> result;
+ for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
+ // Change the sample rate of G722 to 8000 to match SDP.
+ MaybeFixupG722(&voe_codec, 8000);
// Skip uncompressed formats.
if (IsCodec(voe_codec, kL16CodecName)) {
continue;
}
const CodecPref* pref = NULL;
- for (size_t j = 0; j < ARRAY_SIZE(kCodecPrefs); ++j) {
+ for (size_t j = 0; j < arraysize(kCodecPrefs); ++j) {
if (IsCodec(voe_codec, kCodecPrefs[j].name) &&
kCodecPrefs[j].clockrate == voe_codec.plfreq &&
kCodecPrefs[j].channels == voe_codec.channels) {
@@ -465,9 +301,10 @@
if (pref) {
// Use the payload type that we've configured in our pref table;
// use the offset in our pref table to determine the sort order.
- AudioCodec codec(pref->payload_type, voe_codec.plname, voe_codec.plfreq,
- voe_codec.rate, voe_codec.channels,
- ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs));
+ AudioCodec codec(
+ pref->payload_type, voe_codec.plname, voe_codec.plfreq,
+ voe_codec.rate, voe_codec.channels,
+ static_cast<int>(arraysize(kCodecPrefs)) - (pref - kCodecPrefs));
LOG(LS_INFO) << ToString(codec);
if (IsCodec(codec, kIsacCodecName)) {
// Indicate auto-bitrate in signaling.
@@ -488,588 +325,21 @@
// TODO(hellner): Add ptime, sprop-stereo, and stereo
// when they can be set to values other than the default.
}
- codecs_.push_back(codec);
+ result.push_back(codec);
} else {
LOG(LS_WARNING) << "Unexpected codec: " << ToString(voe_codec);
}
}
- }
- // Make sure they are in local preference order.
- std::sort(codecs_.begin(), codecs_.end(), &AudioCodec::Preferable);
-}
-
-bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst* codec) {
- if (voe_wrapper_->codec()->GetCodec(index, *codec) == -1) {
- return false;
- }
- // Change the sample rate of G722 to 8000 to match SDP.
- MaybeFixupG722(codec, 8000);
- return true;
-}
-
-WebRtcVoiceEngine::~WebRtcVoiceEngine() {
- LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
- if (voe_wrapper_->base()->DeRegisterVoiceEngineObserver() == -1) {
- LOG_RTCERR0(DeRegisterVoiceEngineObserver);
- }
- if (adm_) {
- voe_wrapper_.reset();
- adm_->Release();
- adm_ = NULL;
+ // Make sure they are in local preference order.
+ std::sort(result.begin(), result.end(), &AudioCodec::Preferable);
+ return result;
}
- tracing_->SetTraceCallback(NULL);
-}
-
-bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
- RTC_DCHECK(worker_thread == rtc::Thread::Current());
- LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
- bool res = InitInternal();
- if (res) {
- LOG(LS_INFO) << "WebRtcVoiceEngine::Init Done!";
- } else {
- LOG(LS_ERROR) << "WebRtcVoiceEngine::Init failed";
- Terminate();
- }
- return res;
-}
-
-bool WebRtcVoiceEngine::InitInternal() {
- // Temporarily turn logging level up for the Init call
- int old_filter = log_filter_;
- int extended_filter = log_filter_ | SeverityToFilter(rtc::LS_INFO);
- SetTraceFilter(extended_filter);
- SetTraceOptions("");
-
- // Init WebRtc VoiceEngine.
- if (voe_wrapper_->base()->Init(adm_) == -1) {
- LOG_RTCERR0_EX(Init, voe_wrapper_->error());
- SetTraceFilter(old_filter);
- return false;
- }
-
- SetTraceFilter(old_filter);
- SetTraceOptions(log_options_);
-
- // Log the VoiceEngine version info
- char buffer[1024] = "";
- voe_wrapper_->base()->GetVersion(buffer);
- LOG(LS_INFO) << "WebRtc VoiceEngine Version:";
- LogMultiline(rtc::LS_INFO, buffer);
-
- // Save the default AGC configuration settings. This must happen before
- // calling SetOptions or the default will be overwritten.
- if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) {
- LOG_RTCERR0(GetAgcConfig);
- return false;
- }
-
- // Set defaults for options, so that ApplyOptions applies them explicitly
- // when we clear option (channel) overrides. External clients can still
- // modify the defaults via SetOptions (on the media engine).
- if (!SetOptions(GetDefaultEngineOptions())) {
- return false;
- }
-
- // Print our codec list again for the call diagnostic log
- LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
- for (const AudioCodec& codec : codecs_) {
- LOG(LS_INFO) << ToString(codec);
- }
-
- // Disable the DTMF playout when a tone is sent.
- // PlayDtmfTone will be used if local playout is needed.
- if (voe_wrapper_->dtmf()->SetDtmfFeedbackStatus(false) == -1) {
- LOG_RTCERR1(SetDtmfFeedbackStatus, false);
- }
-
- initialized_ = true;
- return true;
-}
-
-void WebRtcVoiceEngine::Terminate() {
- LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
- initialized_ = false;
-
- StopAecDump();
-
- voe_wrapper_->base()->Terminate();
-}
-
-VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call,
- const AudioOptions& options) {
- return new WebRtcVoiceMediaChannel(this, options, call);
-}
-
-bool WebRtcVoiceEngine::SetOptions(const AudioOptions& options) {
- if (!ApplyOptions(options)) {
- return false;
- }
- options_ = options;
- return true;
-}
-
-// AudioOptions defaults are set in InitInternal (for options with corresponding
-// MediaEngineInterface flags) and in SetOptions(int) for flagless options.
-bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
- LOG(LS_INFO) << "ApplyOptions: " << options_in.ToString();
- AudioOptions options = options_in; // The options are modified below.
- // kEcConference is AEC with high suppression.
- webrtc::EcModes ec_mode = webrtc::kEcConference;
- webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone;
- webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
- webrtc::NsModes ns_mode = webrtc::kNsHighSuppression;
- bool aecm_comfort_noise = false;
- if (options.aecm_generate_comfort_noise.Get(&aecm_comfort_noise)) {
- LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
- << aecm_comfort_noise << " (default is false).";
- }
-
-#if defined(IOS)
- // On iOS, VPIO provides built-in EC and AGC.
- options.echo_cancellation.Set(false);
- options.auto_gain_control.Set(false);
- LOG(LS_INFO) << "Always disable AEC and AGC on iOS. Use built-in instead.";
-#elif defined(ANDROID)
- ec_mode = webrtc::kEcAecm;
-#endif
-
-#if defined(IOS) || defined(ANDROID)
- // Set the AGC mode for iOS as well despite disabling it above, to avoid
- // unsupported configuration errors from webrtc.
- agc_mode = webrtc::kAgcFixedDigital;
- options.typing_detection.Set(false);
- options.experimental_agc.Set(false);
- options.extended_filter_aec.Set(false);
- options.experimental_ns.Set(false);
-#endif
-
- // Delay Agnostic AEC automatically turns on EC if not set except on iOS
- // where the feature is not supported.
- bool use_delay_agnostic_aec = false;
-#if !defined(IOS)
- if (options.delay_agnostic_aec.Get(&use_delay_agnostic_aec)) {
- if (use_delay_agnostic_aec) {
- options.echo_cancellation.Set(true);
- options.extended_filter_aec.Set(true);
- ec_mode = webrtc::kEcConference;
- }
- }
-#endif
-
- webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();
-
- bool echo_cancellation = false;
- if (options.echo_cancellation.Get(&echo_cancellation)) {
- // Check if platform supports built-in EC. Currently only supported on
- // Android and in combination with Java based audio layer.
- // TODO(henrika): investigate possibility to support built-in EC also
- // in combination with Open SL ES audio.
- const bool built_in_aec = voe_wrapper_->hw()->BuiltInAECIsAvailable();
- if (built_in_aec) {
- // Built-in EC exists on this device and use_delay_agnostic_aec is not
- // overriding it. Enable/Disable it according to the echo_cancellation
- // audio option.
- const bool enable_built_in_aec =
- echo_cancellation && !use_delay_agnostic_aec;
- if (voe_wrapper_->hw()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
- enable_built_in_aec) {
- // Disable internal software EC if built-in EC is enabled,
- // i.e., replace the software EC with the built-in EC.
- options.echo_cancellation.Set(false);
- echo_cancellation = false;
- LOG(LS_INFO) << "Disabling EC since built-in EC will be used instead";
- }
- }
- if (voep->SetEcStatus(echo_cancellation, ec_mode) == -1) {
- LOG_RTCERR2(SetEcStatus, echo_cancellation, ec_mode);
- return false;
- } else {
- LOG(LS_INFO) << "Echo control set to " << echo_cancellation
- << " with mode " << ec_mode;
- }
-#if !defined(ANDROID)
- // TODO(ajm): Remove the error return on Android from webrtc.
- if (voep->SetEcMetricsStatus(echo_cancellation) == -1) {
- LOG_RTCERR1(SetEcMetricsStatus, echo_cancellation);
- return false;
- }
-#endif
- if (ec_mode == webrtc::kEcAecm) {
- if (voep->SetAecmMode(aecm_mode, aecm_comfort_noise) != 0) {
- LOG_RTCERR2(SetAecmMode, aecm_mode, aecm_comfort_noise);
- return false;
- }
- }
- }
-
- bool auto_gain_control = false;
- if (options.auto_gain_control.Get(&auto_gain_control)) {
- const bool built_in_agc = voe_wrapper_->hw()->BuiltInAGCIsAvailable();
- if (built_in_agc) {
- if (voe_wrapper_->hw()->EnableBuiltInAGC(auto_gain_control) == 0 &&
- auto_gain_control) {
- // Disable internal software AGC if built-in AGC is enabled,
- // i.e., replace the software AGC with the built-in AGC.
- options.auto_gain_control.Set(false);
- auto_gain_control = false;
- LOG(LS_INFO) << "Disabling AGC since built-in AGC will be used instead";
- }
- }
- if (voep->SetAgcStatus(auto_gain_control, agc_mode) == -1) {
- LOG_RTCERR2(SetAgcStatus, auto_gain_control, agc_mode);
- return false;
- } else {
- LOG(LS_INFO) << "Auto gain set to " << auto_gain_control << " with mode "
- << agc_mode;
- }
- }
-
- if (options.tx_agc_target_dbov.IsSet() ||
- options.tx_agc_digital_compression_gain.IsSet() ||
- options.tx_agc_limiter.IsSet()) {
- // Override default_agc_config_. Generally, an unset option means "leave
- // the VoE bits alone" in this function, so we want whatever is set to be
- // stored as the new "default". If we didn't, then setting e.g.
- // tx_agc_target_dbov would reset digital compression gain and limiter
- // settings.
- // Also, if we don't update default_agc_config_, then adjust_agc_delta
- // would be an offset from the original values, and not whatever was set
- // explicitly.
- default_agc_config_.targetLeveldBOv =
- options.tx_agc_target_dbov.GetWithDefaultIfUnset(
- default_agc_config_.targetLeveldBOv);
- default_agc_config_.digitalCompressionGaindB =
- options.tx_agc_digital_compression_gain.GetWithDefaultIfUnset(
- default_agc_config_.digitalCompressionGaindB);
- default_agc_config_.limiterEnable =
- options.tx_agc_limiter.GetWithDefaultIfUnset(
- default_agc_config_.limiterEnable);
- if (voe_wrapper_->processing()->SetAgcConfig(default_agc_config_) == -1) {
- LOG_RTCERR3(SetAgcConfig,
- default_agc_config_.targetLeveldBOv,
- default_agc_config_.digitalCompressionGaindB,
- default_agc_config_.limiterEnable);
- return false;
- }
- }
-
- bool noise_suppression = false;
- if (options.noise_suppression.Get(&noise_suppression)) {
- const bool built_in_ns = voe_wrapper_->hw()->BuiltInNSIsAvailable();
- if (built_in_ns) {
- if (voe_wrapper_->hw()->EnableBuiltInNS(noise_suppression) == 0 &&
- noise_suppression) {
- // Disable internal software NS if built-in NS is enabled,
- // i.e., replace the software NS with the built-in NS.
- options.noise_suppression.Set(false);
- noise_suppression = false;
- LOG(LS_INFO) << "Disabling NS since built-in NS will be used instead";
- }
- }
- if (voep->SetNsStatus(noise_suppression, ns_mode) == -1) {
- LOG_RTCERR2(SetNsStatus, noise_suppression, ns_mode);
- return false;
- } else {
- LOG(LS_INFO) << "Noise suppression set to " << noise_suppression
- << " with mode " << ns_mode;
- }
- }
-
- bool highpass_filter;
- if (options.highpass_filter.Get(&highpass_filter)) {
- LOG(LS_INFO) << "High pass filter enabled? " << highpass_filter;
- if (voep->EnableHighPassFilter(highpass_filter) == -1) {
- LOG_RTCERR1(SetHighpassFilterStatus, highpass_filter);
- return false;
- }
- }
-
- bool stereo_swapping;
- if (options.stereo_swapping.Get(&stereo_swapping)) {
- LOG(LS_INFO) << "Stereo swapping enabled? " << stereo_swapping;
- voep->EnableStereoChannelSwapping(stereo_swapping);
- if (voep->IsStereoChannelSwappingEnabled() != stereo_swapping) {
- LOG_RTCERR1(EnableStereoChannelSwapping, stereo_swapping);
- return false;
- }
- }
-
- int audio_jitter_buffer_max_packets;
- if (options.audio_jitter_buffer_max_packets.Get(
- &audio_jitter_buffer_max_packets)) {
- LOG(LS_INFO) << "NetEq capacity is " << audio_jitter_buffer_max_packets;
- voe_config_.Set<webrtc::NetEqCapacityConfig>(
- new webrtc::NetEqCapacityConfig(audio_jitter_buffer_max_packets));
- }
-
- bool audio_jitter_buffer_fast_accelerate;
- if (options.audio_jitter_buffer_fast_accelerate.Get(
- &audio_jitter_buffer_fast_accelerate)) {
- LOG(LS_INFO) << "NetEq fast mode? " << audio_jitter_buffer_fast_accelerate;
- voe_config_.Set<webrtc::NetEqFastAccelerate>(
- new webrtc::NetEqFastAccelerate(audio_jitter_buffer_fast_accelerate));
- }
-
- bool typing_detection;
- if (options.typing_detection.Get(&typing_detection)) {
- LOG(LS_INFO) << "Typing detection is enabled? " << typing_detection;
- if (voep->SetTypingDetectionStatus(typing_detection) == -1) {
- // In case of error, log the info and continue
- LOG_RTCERR1(SetTypingDetectionStatus, typing_detection);
- }
- }
-
- int adjust_agc_delta;
- if (options.adjust_agc_delta.Get(&adjust_agc_delta)) {
- LOG(LS_INFO) << "Adjust agc delta is " << adjust_agc_delta;
- if (!AdjustAgcLevel(adjust_agc_delta)) {
- return false;
- }
- }
-
- bool aec_dump;
- if (options.aec_dump.Get(&aec_dump)) {
- LOG(LS_INFO) << "Aec dump is enabled? " << aec_dump;
- if (aec_dump)
- StartAecDump(kAecDumpByAudioOptionFilename);
- else
- StopAecDump();
- }
-
- webrtc::Config config;
-
- delay_agnostic_aec_.SetFrom(options.delay_agnostic_aec);
- bool delay_agnostic_aec;
- if (delay_agnostic_aec_.Get(&delay_agnostic_aec)) {
- LOG(LS_INFO) << "Delay agnostic aec is enabled? " << delay_agnostic_aec;
- config.Set<webrtc::DelayAgnostic>(
- new webrtc::DelayAgnostic(delay_agnostic_aec));
- }
-
- extended_filter_aec_.SetFrom(options.extended_filter_aec);
- bool extended_filter;
- if (extended_filter_aec_.Get(&extended_filter)) {
- LOG(LS_INFO) << "Extended filter aec is enabled? " << extended_filter;
- config.Set<webrtc::ExtendedFilter>(
- new webrtc::ExtendedFilter(extended_filter));
- }
-
- experimental_ns_.SetFrom(options.experimental_ns);
- bool experimental_ns;
- if (experimental_ns_.Get(&experimental_ns)) {
- LOG(LS_INFO) << "Experimental ns is enabled? " << experimental_ns;
- config.Set<webrtc::ExperimentalNs>(
- new webrtc::ExperimentalNs(experimental_ns));
- }
-
- // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
- // returns NULL on audio_processing().
- webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
- if (audioproc) {
- audioproc->SetExtraOptions(config);
- }
-
- uint32_t recording_sample_rate;
- if (options.recording_sample_rate.Get(&recording_sample_rate)) {
- LOG(LS_INFO) << "Recording sample rate is " << recording_sample_rate;
- if (voe_wrapper_->hw()->SetRecordingSampleRate(recording_sample_rate)) {
- LOG_RTCERR1(SetRecordingSampleRate, recording_sample_rate);
- }
- }
-
- uint32_t playout_sample_rate;
- if (options.playout_sample_rate.Get(&playout_sample_rate)) {
- LOG(LS_INFO) << "Playout sample rate is " << playout_sample_rate;
- if (voe_wrapper_->hw()->SetPlayoutSampleRate(playout_sample_rate)) {
- LOG_RTCERR1(SetPlayoutSampleRate, playout_sample_rate);
- }
- }
-
- return true;
-}
-
-// TODO(juberti): Refactor this so that the core logic can be used to set the
-// soundclip device. At that time, reinstate the soundclip pause/resume code.
-bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
- const Device* out_device) {
-#if !defined(IOS)
- int in_id = in_device ? rtc::FromString<int>(in_device->id) :
- kDefaultAudioDeviceId;
- int out_id = out_device ? rtc::FromString<int>(out_device->id) :
- kDefaultAudioDeviceId;
- // The device manager uses -1 as the default device, which was the case for
- // VoE 3.5. VoE 4.0, however, uses 0 as the default in Linux and Mac.
-#ifndef WIN32
- if (-1 == in_id) {
- in_id = kDefaultAudioDeviceId;
- }
- if (-1 == out_id) {
- out_id = kDefaultAudioDeviceId;
- }
-#endif
-
- std::string in_name = (in_id != kDefaultAudioDeviceId) ?
- in_device->name : "Default device";
- std::string out_name = (out_id != kDefaultAudioDeviceId) ?
- out_device->name : "Default device";
- LOG(LS_INFO) << "Setting microphone to (id=" << in_id << ", name=" << in_name
- << ") and speaker to (id=" << out_id << ", name=" << out_name
- << ")";
-
- // Must also pause all audio playback and capture.
- bool ret = true;
- for (WebRtcVoiceMediaChannel* channel : channels_) {
- if (!channel->PausePlayout()) {
- LOG(LS_WARNING) << "Failed to pause playout";
- ret = false;
- }
- if (!channel->PauseSend()) {
- LOG(LS_WARNING) << "Failed to pause send";
- ret = false;
- }
- }
-
- // Find the recording device id in VoiceEngine and set recording device.
- if (!FindWebRtcAudioDeviceId(true, in_name, in_id, &in_id)) {
- ret = false;
- }
- if (ret) {
- if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
- LOG_RTCERR2(SetRecordingDevice, in_name, in_id);
- ret = false;
- }
- webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
- if (ap)
- ap->Initialize();
- }
-
- // Find the playout device id in VoiceEngine and set playout device.
- if (!FindWebRtcAudioDeviceId(false, out_name, out_id, &out_id)) {
- LOG(LS_WARNING) << "Failed to find VoiceEngine device id for " << out_name;
- ret = false;
- }
- if (ret) {
- if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
- LOG_RTCERR2(SetPlayoutDevice, out_name, out_id);
- ret = false;
- }
- }
-
- // Resume all audio playback and capture.
- for (WebRtcVoiceMediaChannel* channel : channels_) {
- if (!channel->ResumePlayout()) {
- LOG(LS_WARNING) << "Failed to resume playout";
- ret = false;
- }
- if (!channel->ResumeSend()) {
- LOG(LS_WARNING) << "Failed to resume send";
- ret = false;
- }
- }
-
- if (ret) {
- LOG(LS_INFO) << "Set microphone to (id=" << in_id <<" name=" << in_name
- << ") and speaker to (id="<< out_id << " name=" << out_name
- << ")";
- }
-
- return ret;
-#else
- return true;
-#endif // !IOS
-}
-
-bool WebRtcVoiceEngine::FindWebRtcAudioDeviceId(
- bool is_input, const std::string& dev_name, int dev_id, int* rtc_id) {
- // In Linux, VoiceEngine uses the same device dev_id as the device manager.
-#if defined(LINUX) || defined(ANDROID)
- *rtc_id = dev_id;
- return true;
-#else
- // In Windows and Mac, we need to find the VoiceEngine device id by name
- // unless the input dev_id is the default device id.
- if (kDefaultAudioDeviceId == dev_id) {
- *rtc_id = dev_id;
- return true;
- }
-
- // Get the number of VoiceEngine audio devices.
- int count = 0;
- if (is_input) {
- if (-1 == voe_wrapper_->hw()->GetNumOfRecordingDevices(count)) {
- LOG_RTCERR0(GetNumOfRecordingDevices);
- return false;
- }
- } else {
- if (-1 == voe_wrapper_->hw()->GetNumOfPlayoutDevices(count)) {
- LOG_RTCERR0(GetNumOfPlayoutDevices);
- return false;
- }
- }
-
- for (int i = 0; i < count; ++i) {
- char name[128];
- char guid[128];
- if (is_input) {
- voe_wrapper_->hw()->GetRecordingDeviceName(i, name, guid);
- LOG(LS_VERBOSE) << "VoiceEngine microphone " << i << ": " << name;
- } else {
- voe_wrapper_->hw()->GetPlayoutDeviceName(i, name, guid);
- LOG(LS_VERBOSE) << "VoiceEngine speaker " << i << ": " << name;
- }
-
- std::string webrtc_name(name);
- if (dev_name.compare(0, webrtc_name.size(), webrtc_name) == 0) {
- *rtc_id = i;
- return true;
- }
- }
- LOG(LS_WARNING) << "VoiceEngine cannot find device: " << dev_name;
- return false;
-#endif
-}
-
-bool WebRtcVoiceEngine::GetOutputVolume(int* level) {
- unsigned int ulevel;
- if (voe_wrapper_->volume()->GetSpeakerVolume(ulevel) == -1) {
- LOG_RTCERR1(GetSpeakerVolume, level);
- return false;
- }
- *level = ulevel;
- return true;
-}
-
-bool WebRtcVoiceEngine::SetOutputVolume(int level) {
- RTC_DCHECK(level >= 0 && level <= 255);
- if (voe_wrapper_->volume()->SetSpeakerVolume(level) == -1) {
- LOG_RTCERR1(SetSpeakerVolume, level);
- return false;
- }
- return true;
-}
-
-int WebRtcVoiceEngine::GetInputLevel() {
- unsigned int ulevel;
- return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
- static_cast<int>(ulevel) : -1;
-}
-
-const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
- return codecs_;
-}
-
-bool WebRtcVoiceEngine::FindCodec(const AudioCodec& in) {
- return FindWebRtcCodec(in, NULL);
-}
-
-// Get the VoiceEngine codec that matches |in|, with the supplied settings.
-bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in,
- webrtc::CodecInst* out) {
- int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
- for (int i = 0; i < ncodecs; ++i) {
- webrtc::CodecInst voe_codec;
- if (GetVoeCodec(i, &voe_codec)) {
+ static bool ToCodecInst(const AudioCodec& in,
+ webrtc::CodecInst* out) {
+ for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
+ // Change the sample rate of G722 to 8000 to match SDP.
+ MaybeFixupG722(&voe_codec, 8000);
AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
voe_codec.rate, voe_codec.channels, 0);
bool multi_rate = IsCodecMultiRate(voe_codec);
@@ -1102,84 +372,575 @@
return true;
}
}
+ return false;
}
- return false;
-}
-const std::vector<RtpHeaderExtension>&
-WebRtcVoiceEngine::rtp_header_extensions() const {
- return rtp_header_extensions_;
-}
-void WebRtcVoiceEngine::SetLogging(int min_sev, const char* filter) {
- // if min_sev == -1, we keep the current log level.
- if (min_sev >= 0) {
- SetTraceFilter(SeverityToFilter(min_sev));
+ static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
+ for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
+ if (IsCodec(codec, kCodecPrefs[i].name) &&
+ kCodecPrefs[i].clockrate == codec.plfreq) {
+ return kCodecPrefs[i].is_multi_rate;
+ }
+ }
+ return false;
}
- log_options_ = filter;
- SetTraceOptions(initialized_ ? log_options_ : "");
+
+ // If the AudioCodec param kCodecParamPTime is set, then we will set it to
+ // codec pacsize if it's valid, or we will pick the next smallest value we
+ // support.
+ // TODO(Brave): Query supported packet sizes from ACM when the API is ready.
+ static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
+ for (const CodecPref& codec_pref : kCodecPrefs) {
+ if ((IsCodec(*codec, codec_pref.name) &&
+ codec_pref.clockrate == codec->plfreq) ||
+ IsCodec(*codec, kG722CodecName)) {
+ int packet_size_ms = SelectPacketSize(codec_pref, ptime_ms);
+ if (packet_size_ms) {
+ // Convert unit from milli-seconds to samples.
+ codec->pacsize = (codec->plfreq / 1000) * packet_size_ms;
+ return true;
+ }
+ }
+ }
+ return false;
+ }
+
+ private:
+ static const int kMaxNumPacketSize = 6;
+ struct CodecPref {
+ const char* name;
+ int clockrate;
+ size_t channels;
+ int payload_type;
+ bool is_multi_rate;
+ int packet_sizes_ms[kMaxNumPacketSize];
+ };
+ // Note: keep the supported packet sizes in ascending order.
+ static const CodecPref kCodecPrefs[12];
+
+ static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
+ int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
+ for (int packet_size_ms : codec_pref.packet_sizes_ms) {
+ if (packet_size_ms && packet_size_ms <= ptime_ms) {
+ selected_packet_size_ms = packet_size_ms;
+ }
+ }
+ return selected_packet_size_ms;
+ }
+
+ // Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
+ // which says that G722 should be advertised as 8 kHz although it is a 16 kHz
+ // codec.
+ static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
+ if (IsCodec(*voe_codec, kG722CodecName)) {
+ // If the ASSERT triggers, the codec definition in WebRTC VoiceEngine
+ // has changed, and this special case is no longer needed.
+ RTC_DCHECK(voe_codec->plfreq != new_plfreq);
+ voe_codec->plfreq = new_plfreq;
+ }
+ }
+};
+
+const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[12] = {
+ { kOpusCodecName, 48000, 2, 111, true, { 10, 20, 40, 60 } },
+ { kIsacCodecName, 16000, 1, 103, true, { 30, 60 } },
+ { kIsacCodecName, 32000, 1, 104, true, { 30 } },
+ // G722 should be advertised as 8000 Hz because of the RFC "bug".
+ { kG722CodecName, 8000, 1, 9, false, { 10, 20, 30, 40, 50, 60 } },
+ { kIlbcCodecName, 8000, 1, 102, false, { 20, 30, 40, 60 } },
+ { kPcmuCodecName, 8000, 1, 0, false, { 10, 20, 30, 40, 50, 60 } },
+ { kPcmaCodecName, 8000, 1, 8, false, { 10, 20, 30, 40, 50, 60 } },
+ { kCnCodecName, 32000, 1, 106, false, { } },
+ { kCnCodecName, 16000, 1, 105, false, { } },
+ { kCnCodecName, 8000, 1, 13, false, { } },
+ { kRedCodecName, 8000, 1, 127, false, { } },
+ { kDtmfCodecName, 8000, 1, 126, false, { } },
+};
+} // namespace {
+
+bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
+ webrtc::CodecInst* out) {
+ return WebRtcVoiceCodecs::ToCodecInst(in, out);
}
-int WebRtcVoiceEngine::GetLastEngineError() {
- return voe_wrapper_->error();
+WebRtcVoiceEngine::WebRtcVoiceEngine()
+ : voe_wrapper_(new VoEWrapper()),
+ audio_state_(webrtc::AudioState::Create(MakeAudioStateConfig(voe()))) {
+ Construct();
}
-void WebRtcVoiceEngine::SetTraceFilter(int filter) {
- log_filter_ = filter;
- tracing_->SetTraceFilter(filter);
+WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper)
+ : voe_wrapper_(voe_wrapper) {
+ Construct();
}
-// We suppport three different logging settings for VoiceEngine:
-// 1. Observer callback that goes into talk diagnostic logfile.
-// Use --logfile and --loglevel
-//
-// 2. Encrypted VoiceEngine log for debugging VoiceEngine.
-// Use --voice_loglevel --voice_logfilter "tracefile file_name"
-//
-// 3. EC log and dump for debugging QualityEngine.
-// Use --voice_loglevel --voice_logfilter "recordEC file_name"
-//
-// For more details see: "https://sites.google.com/a/google.com/wavelet/Home/
-// Magic-Flute--RTC-Engine-/Magic-Flute-Command-Line-Parameters"
-void WebRtcVoiceEngine::SetTraceOptions(const std::string& options) {
- // Set encrypted trace file.
- std::vector<std::string> opts;
- rtc::tokenize(options, ' ', '"', '"', &opts);
- std::vector<std::string>::iterator tracefile =
- std::find(opts.begin(), opts.end(), "tracefile");
- if (tracefile != opts.end() && ++tracefile != opts.end()) {
- // Write encrypted debug output (at same loglevel) to file
- // EncryptedTraceFile no longer supported.
- if (tracing_->SetTraceFile(tracefile->c_str()) == -1) {
- LOG_RTCERR1(SetTraceFile, *tracefile);
+void WebRtcVoiceEngine::Construct() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
+
+ signal_thread_checker_.DetachFromThread();
+ std::memset(&default_agc_config_, 0, sizeof(default_agc_config_));
+ voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
+
+ webrtc::Trace::set_level_filter(kDefaultTraceFilter);
+ webrtc::Trace::SetTraceCallback(this);
+
+ // Load our audio codec list.
+ codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
+}
+
+WebRtcVoiceEngine::~WebRtcVoiceEngine() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
+ if (adm_) {
+ voe_wrapper_.reset();
+ adm_->Release();
+ adm_ = NULL;
+ }
+ webrtc::Trace::SetTraceCallback(nullptr);
+}
+
+bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread == rtc::Thread::Current());
+ LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
+ bool res = InitInternal();
+ if (res) {
+ LOG(LS_INFO) << "WebRtcVoiceEngine::Init Done!";
+ } else {
+ LOG(LS_ERROR) << "WebRtcVoiceEngine::Init failed";
+ Terminate();
+ }
+ return res;
+}
+
+bool WebRtcVoiceEngine::InitInternal() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ // Temporarily turn logging level up for the Init call
+ webrtc::Trace::set_level_filter(kElevatedTraceFilter);
+ LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
+ if (voe_wrapper_->base()->Init(adm_) == -1) {
+ LOG_RTCERR0_EX(Init, voe_wrapper_->error());
+ return false;
+ }
+ webrtc::Trace::set_level_filter(kDefaultTraceFilter);
+
+ // Save the default AGC configuration settings. This must happen before
+ // calling ApplyOptions or the default will be overwritten.
+ if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) {
+ LOG_RTCERR0(GetAgcConfig);
+ return false;
+ }
+
+ // Print our codec list again for the call diagnostic log
+ LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
+ for (const AudioCodec& codec : codecs_) {
+ LOG(LS_INFO) << ToString(codec);
+ }
+
+ SetDefaultDevices();
+
+ initialized_ = true;
+ return true;
+}
+
+void WebRtcVoiceEngine::Terminate() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
+ initialized_ = false;
+
+ StopAecDump();
+
+ voe_wrapper_->base()->Terminate();
+}
+
+rtc::scoped_refptr<webrtc::AudioState>
+ WebRtcVoiceEngine::GetAudioState() const {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ return audio_state_;
+}
+
+VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call,
+ const AudioOptions& options) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ return new WebRtcVoiceMediaChannel(this, options, call);
+}
+
+bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << "ApplyOptions: " << options_in.ToString();
+
+ // Default engine options.
+ AudioOptions options;
+ options.echo_cancellation = rtc::Optional<bool>(true);
+ options.auto_gain_control = rtc::Optional<bool>(true);
+ options.noise_suppression = rtc::Optional<bool>(true);
+ options.highpass_filter = rtc::Optional<bool>(true);
+ options.stereo_swapping = rtc::Optional<bool>(false);
+ options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
+ options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
+ options.typing_detection = rtc::Optional<bool>(true);
+ options.adjust_agc_delta = rtc::Optional<int>(0);
+ options.experimental_agc = rtc::Optional<bool>(false);
+ options.extended_filter_aec = rtc::Optional<bool>(false);
+ options.delay_agnostic_aec = rtc::Optional<bool>(false);
+ options.experimental_ns = rtc::Optional<bool>(false);
+ options.aec_dump = rtc::Optional<bool>(false);
+
+ // Apply any given options on top.
+ options.SetAll(options_in);
+
+ // kEcConference is AEC with high suppression.
+ webrtc::EcModes ec_mode = webrtc::kEcConference;
+ webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone;
+ webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
+ webrtc::NsModes ns_mode = webrtc::kNsHighSuppression;
+ if (options.aecm_generate_comfort_noise) {
+ LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
+ << *options.aecm_generate_comfort_noise
+ << " (default is false).";
+ }
+
+#if defined(WEBRTC_IOS)
+ // On iOS, VPIO provides built-in EC and AGC.
+ options.echo_cancellation = rtc::Optional<bool>(false);
+ options.auto_gain_control = rtc::Optional<bool>(false);
+ LOG(LS_INFO) << "Always disable AEC and AGC on iOS. Use built-in instead.";
+#elif defined(ANDROID)
+ ec_mode = webrtc::kEcAecm;
+#endif
+
+#if defined(WEBRTC_IOS) || defined(ANDROID)
+ // Set the AGC mode for iOS as well despite disabling it above, to avoid
+ // unsupported configuration errors from webrtc.
+ agc_mode = webrtc::kAgcFixedDigital;
+ options.typing_detection = rtc::Optional<bool>(false);
+ options.experimental_agc = rtc::Optional<bool>(false);
+ options.extended_filter_aec = rtc::Optional<bool>(false);
+ options.experimental_ns = rtc::Optional<bool>(false);
+#endif
+
+ // Delay Agnostic AEC automatically turns on EC if not set except on iOS
+ // where the feature is not supported.
+ bool use_delay_agnostic_aec = false;
+#if !defined(WEBRTC_IOS)
+ if (options.delay_agnostic_aec) {
+ use_delay_agnostic_aec = *options.delay_agnostic_aec;
+ if (use_delay_agnostic_aec) {
+ options.echo_cancellation = rtc::Optional<bool>(true);
+ options.extended_filter_aec = rtc::Optional<bool>(true);
+ ec_mode = webrtc::kEcConference;
+ }
+ }
+#endif
+
+ webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();
+
+ if (options.echo_cancellation) {
+ // Check if platform supports built-in EC. Currently only supported on
+ // Android and in combination with Java based audio layer.
+ // TODO(henrika): investigate possibility to support built-in EC also
+ // in combination with Open SL ES audio.
+ const bool built_in_aec = voe_wrapper_->hw()->BuiltInAECIsAvailable();
+ if (built_in_aec) {
+ // Built-in EC exists on this device and use_delay_agnostic_aec is not
+ // overriding it. Enable/Disable it according to the echo_cancellation
+ // audio option.
+ const bool enable_built_in_aec =
+ *options.echo_cancellation && !use_delay_agnostic_aec;
+ if (voe_wrapper_->hw()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
+ enable_built_in_aec) {
+ // Disable internal software EC if built-in EC is enabled,
+ // i.e., replace the software EC with the built-in EC.
+ options.echo_cancellation = rtc::Optional<bool>(false);
+ LOG(LS_INFO) << "Disabling EC since built-in EC will be used instead";
+ }
+ }
+ if (voep->SetEcStatus(*options.echo_cancellation, ec_mode) == -1) {
+ LOG_RTCERR2(SetEcStatus, *options.echo_cancellation, ec_mode);
+ return false;
+ } else {
+ LOG(LS_INFO) << "Echo control set to " << *options.echo_cancellation
+ << " with mode " << ec_mode;
+ }
+#if !defined(ANDROID)
+ // TODO(ajm): Remove the error return on Android from webrtc.
+ if (voep->SetEcMetricsStatus(*options.echo_cancellation) == -1) {
+ LOG_RTCERR1(SetEcMetricsStatus, *options.echo_cancellation);
+ return false;
+ }
+#endif
+ if (ec_mode == webrtc::kEcAecm) {
+ bool cn = options.aecm_generate_comfort_noise.value_or(false);
+ if (voep->SetAecmMode(aecm_mode, cn) != 0) {
+ LOG_RTCERR2(SetAecmMode, aecm_mode, cn);
+ return false;
+ }
}
}
- // Allow trace options to override the trace filter. We default
- // it to log_filter_ (as a translation of libjingle log levels)
- // elsewhere, but this allows clients to explicitly set webrtc
- // log levels.
- std::vector<std::string>::iterator tracefilter =
- std::find(opts.begin(), opts.end(), "tracefilter");
- if (tracefilter != opts.end() && ++tracefilter != opts.end()) {
- if (!tracing_->SetTraceFilter(rtc::FromString<int>(*tracefilter))) {
- LOG_RTCERR1(SetTraceFilter, *tracefilter);
+ if (options.auto_gain_control) {
+ const bool built_in_agc = voe_wrapper_->hw()->BuiltInAGCIsAvailable();
+ if (built_in_agc) {
+ if (voe_wrapper_->hw()->EnableBuiltInAGC(*options.auto_gain_control) ==
+ 0 &&
+ *options.auto_gain_control) {
+ // Disable internal software AGC if built-in AGC is enabled,
+ // i.e., replace the software AGC with the built-in AGC.
+ options.auto_gain_control = rtc::Optional<bool>(false);
+ LOG(LS_INFO) << "Disabling AGC since built-in AGC will be used instead";
+ }
+ }
+ if (voep->SetAgcStatus(*options.auto_gain_control, agc_mode) == -1) {
+ LOG_RTCERR2(SetAgcStatus, *options.auto_gain_control, agc_mode);
+ return false;
+ } else {
+ LOG(LS_INFO) << "Auto gain set to " << *options.auto_gain_control
+ << " with mode " << agc_mode;
}
}
- // Set AEC dump file
- std::vector<std::string>::iterator recordEC =
- std::find(opts.begin(), opts.end(), "recordEC");
- if (recordEC != opts.end()) {
- ++recordEC;
- if (recordEC != opts.end())
- StartAecDump(recordEC->c_str());
+ if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain ||
+ options.tx_agc_limiter) {
+ // Override default_agc_config_. Generally, an unset option means "leave
+ // the VoE bits alone" in this function, so we want whatever is set to be
+ // stored as the new "default". If we didn't, then setting e.g.
+ // tx_agc_target_dbov would reset digital compression gain and limiter
+ // settings.
+ // Also, if we don't update default_agc_config_, then adjust_agc_delta
+ // would be an offset from the original values, and not whatever was set
+ // explicitly.
+ default_agc_config_.targetLeveldBOv = options.tx_agc_target_dbov.value_or(
+ default_agc_config_.targetLeveldBOv);
+ default_agc_config_.digitalCompressionGaindB =
+ options.tx_agc_digital_compression_gain.value_or(
+ default_agc_config_.digitalCompressionGaindB);
+ default_agc_config_.limiterEnable =
+ options.tx_agc_limiter.value_or(default_agc_config_.limiterEnable);
+ if (voe_wrapper_->processing()->SetAgcConfig(default_agc_config_) == -1) {
+ LOG_RTCERR3(SetAgcConfig,
+ default_agc_config_.targetLeveldBOv,
+ default_agc_config_.digitalCompressionGaindB,
+ default_agc_config_.limiterEnable);
+ return false;
+ }
+ }
+
+ if (options.noise_suppression) {
+ const bool built_in_ns = voe_wrapper_->hw()->BuiltInNSIsAvailable();
+ if (built_in_ns) {
+ if (voe_wrapper_->hw()->EnableBuiltInNS(*options.noise_suppression) ==
+ 0 &&
+ *options.noise_suppression) {
+ // Disable internal software NS if built-in NS is enabled,
+ // i.e., replace the software NS with the built-in NS.
+ options.noise_suppression = rtc::Optional<bool>(false);
+ LOG(LS_INFO) << "Disabling NS since built-in NS will be used instead";
+ }
+ }
+ if (voep->SetNsStatus(*options.noise_suppression, ns_mode) == -1) {
+ LOG_RTCERR2(SetNsStatus, *options.noise_suppression, ns_mode);
+ return false;
+ } else {
+ LOG(LS_INFO) << "Noise suppression set to " << *options.noise_suppression
+ << " with mode " << ns_mode;
+ }
+ }
+
+ if (options.highpass_filter) {
+ LOG(LS_INFO) << "High pass filter enabled? " << *options.highpass_filter;
+ if (voep->EnableHighPassFilter(*options.highpass_filter) == -1) {
+ LOG_RTCERR1(SetHighpassFilterStatus, *options.highpass_filter);
+ return false;
+ }
+ }
+
+ if (options.stereo_swapping) {
+ LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping;
+ voep->EnableStereoChannelSwapping(*options.stereo_swapping);
+ if (voep->IsStereoChannelSwappingEnabled() != *options.stereo_swapping) {
+ LOG_RTCERR1(EnableStereoChannelSwapping, *options.stereo_swapping);
+ return false;
+ }
+ }
+
+ if (options.audio_jitter_buffer_max_packets) {
+ LOG(LS_INFO) << "NetEq capacity is "
+ << *options.audio_jitter_buffer_max_packets;
+ voe_config_.Set<webrtc::NetEqCapacityConfig>(
+ new webrtc::NetEqCapacityConfig(
+ *options.audio_jitter_buffer_max_packets));
+ }
+
+ if (options.audio_jitter_buffer_fast_accelerate) {
+ LOG(LS_INFO) << "NetEq fast mode? "
+ << *options.audio_jitter_buffer_fast_accelerate;
+ voe_config_.Set<webrtc::NetEqFastAccelerate>(
+ new webrtc::NetEqFastAccelerate(
+ *options.audio_jitter_buffer_fast_accelerate));
+ }
+
+ if (options.typing_detection) {
+ LOG(LS_INFO) << "Typing detection is enabled? "
+ << *options.typing_detection;
+ if (voep->SetTypingDetectionStatus(*options.typing_detection) == -1) {
+ // In case of error, log the info and continue
+ LOG_RTCERR1(SetTypingDetectionStatus, *options.typing_detection);
+ }
+ }
+
+ if (options.adjust_agc_delta) {
+ LOG(LS_INFO) << "Adjust agc delta is " << *options.adjust_agc_delta;
+ if (!AdjustAgcLevel(*options.adjust_agc_delta)) {
+ return false;
+ }
+ }
+
+ if (options.aec_dump) {
+ LOG(LS_INFO) << "Aec dump is enabled? " << *options.aec_dump;
+ if (*options.aec_dump)
+ StartAecDump(kAecDumpByAudioOptionFilename);
else
StopAecDump();
}
+
+ webrtc::Config config;
+
+ if (options.delay_agnostic_aec)
+ delay_agnostic_aec_ = options.delay_agnostic_aec;
+ if (delay_agnostic_aec_) {
+ LOG(LS_INFO) << "Delay agnostic aec is enabled? " << *delay_agnostic_aec_;
+ config.Set<webrtc::DelayAgnostic>(
+ new webrtc::DelayAgnostic(*delay_agnostic_aec_));
+ }
+
+ if (options.extended_filter_aec) {
+ extended_filter_aec_ = options.extended_filter_aec;
+ }
+ if (extended_filter_aec_) {
+ LOG(LS_INFO) << "Extended filter aec is enabled? " << *extended_filter_aec_;
+ config.Set<webrtc::ExtendedFilter>(
+ new webrtc::ExtendedFilter(*extended_filter_aec_));
+ }
+
+ if (options.experimental_ns) {
+ experimental_ns_ = options.experimental_ns;
+ }
+ if (experimental_ns_) {
+ LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_;
+ config.Set<webrtc::ExperimentalNs>(
+ new webrtc::ExperimentalNs(*experimental_ns_));
+ }
+
+ // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
+ // returns NULL on audio_processing().
+ webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
+ if (audioproc) {
+ audioproc->SetExtraOptions(config);
+ }
+
+ if (options.recording_sample_rate) {
+ LOG(LS_INFO) << "Recording sample rate is "
+ << *options.recording_sample_rate;
+ if (voe_wrapper_->hw()->SetRecordingSampleRate(
+ *options.recording_sample_rate)) {
+ LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
+ }
+ }
+
+ if (options.playout_sample_rate) {
+ LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
+ if (voe_wrapper_->hw()->SetPlayoutSampleRate(
+ *options.playout_sample_rate)) {
+ LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
+ }
+ }
+
+ return true;
+}
+
+void WebRtcVoiceEngine::SetDefaultDevices() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+#if !defined(WEBRTC_IOS)
+ int in_id = kDefaultAudioDeviceId;
+ int out_id = kDefaultAudioDeviceId;
+ LOG(LS_INFO) << "Setting microphone to (id=" << in_id
+ << ") and speaker to (id=" << out_id << ")";
+
+ bool ret = true;
+ if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
+ LOG_RTCERR1(SetRecordingDevice, in_id);
+ ret = false;
+ }
+ webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
+ if (ap) {
+ ap->Initialize();
+ }
+
+ if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
+ LOG_RTCERR1(SetPlayoutDevice, out_id);
+ ret = false;
+ }
+
+ if (ret) {
+ LOG(LS_INFO) << "Set microphone to (id=" << in_id
+ << ") and speaker to (id=" << out_id << ")";
+ }
+#endif // !WEBRTC_IOS
+}
+
+bool WebRtcVoiceEngine::GetOutputVolume(int* level) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ unsigned int ulevel;
+ if (voe_wrapper_->volume()->GetSpeakerVolume(ulevel) == -1) {
+ LOG_RTCERR1(GetSpeakerVolume, level);
+ return false;
+ }
+ *level = ulevel;
+ return true;
+}
+
+bool WebRtcVoiceEngine::SetOutputVolume(int level) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(level >= 0 && level <= 255);
+ if (voe_wrapper_->volume()->SetSpeakerVolume(level) == -1) {
+ LOG_RTCERR1(SetSpeakerVolume, level);
+ return false;
+ }
+ return true;
+}
+
+int WebRtcVoiceEngine::GetInputLevel() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ unsigned int ulevel;
+ return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
+ static_cast<int>(ulevel) : -1;
+}
+
+const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
+ RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+ return codecs_;
+}
+
+RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
+ RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+ RtpCapabilities capabilities;
+ capabilities.header_extensions.push_back(RtpHeaderExtension(
+ kRtpAudioLevelHeaderExtension, kRtpAudioLevelHeaderExtensionDefaultId));
+ capabilities.header_extensions.push_back(
+ RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
+ kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
+ return capabilities;
+}
+
+int WebRtcVoiceEngine::GetLastEngineError() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ return voe_wrapper_->error();
}
void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
int length) {
+ // Note: This callback can happen on any thread!
rtc::LoggingSeverity sev = rtc::LS_VERBOSE;
if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
sev = rtc::LS_ERROR;
@@ -1201,34 +962,24 @@
}
}
-void WebRtcVoiceEngine::CallbackOnError(int channel_id, int err_code) {
- RTC_DCHECK(channel_id == -1);
- LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel "
- << channel_id << ".";
- rtc::CritScope lock(&channels_cs_);
- for (WebRtcVoiceMediaChannel* channel : channels_) {
- channel->OnError(err_code);
- }
-}
-
void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
- RTC_DCHECK(channel != NULL);
- rtc::CritScope lock(&channels_cs_);
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(channel);
channels_.push_back(channel);
}
void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
- rtc::CritScope lock(&channels_cs_);
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
auto it = std::find(channels_.begin(), channels_.end(), channel);
- if (it != channels_.end()) {
- channels_.erase(it);
- }
+ RTC_DCHECK(it != channels_.end());
+ channels_.erase(it);
}
// Adjusts the default AGC target level by the specified delta.
// NB: If we start messing with other config fields, we'll want
// to save the current webrtc::AgcConfig as well.
bool WebRtcVoiceEngine::AdjustAgcLevel(int delta) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
webrtc::AgcConfig config = default_agc_config_;
config.targetLeveldBOv -= delta;
@@ -1244,6 +995,7 @@
}
bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (initialized_) {
LOG(LS_WARNING) << "SetAudioDeviceModule can not be called after Init.";
return false;
@@ -1260,6 +1012,7 @@
}
bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
if (!aec_dump_file_stream) {
LOG(LS_ERROR) << "Could not open AEC dump file stream.";
@@ -1279,6 +1032,7 @@
}
void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (!is_dumping_aec_) {
// Start dumping AEC when we are not dumping.
if (voe_wrapper_->processing()->StartDebugRecording(
@@ -1291,6 +1045,7 @@
}
void WebRtcVoiceEngine::StopAecDump() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (is_dumping_aec_) {
// Stop dumping AEC when we are dumping.
if (voe_wrapper_->processing()->StopDebugRecording() !=
@@ -1302,14 +1057,17 @@
}
bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return voe_wrapper_->codec()->GetEventLog()->StartLogging(file);
}
void WebRtcVoiceEngine::StopRtcEventLog() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
voe_wrapper_->codec()->GetEventLog()->StopLogging();
}
int WebRtcVoiceEngine::CreateVoEChannel() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
return voe_wrapper_->base()->CreateChannel(voe_config_);
}
@@ -1317,33 +1075,61 @@
: public AudioRenderer::Sink {
public:
WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport,
- uint32_t ssrc, webrtc::Call* call)
- : channel_(ch),
- voe_audio_transport_(voe_audio_transport),
- call_(call) {
+ uint32_t ssrc, const std::string& c_name,
+ const std::vector<webrtc::RtpExtension>& extensions,
+ webrtc::Call* call)
+ : voe_audio_transport_(voe_audio_transport),
+ call_(call),
+ config_(nullptr) {
RTC_DCHECK_GE(ch, 0);
// TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
// RTC_DCHECK(voe_audio_transport);
RTC_DCHECK(call);
audio_capture_thread_checker_.DetachFromThread();
- webrtc::AudioSendStream::Config config(nullptr);
- config.voe_channel_id = channel_;
- config.rtp.ssrc = ssrc;
- stream_ = call_->CreateAudioSendStream(config);
- RTC_DCHECK(stream_);
+ config_.rtp.ssrc = ssrc;
+ config_.rtp.c_name = c_name;
+ config_.voe_channel_id = ch;
+ RecreateAudioSendStream(extensions);
}
+
~WebRtcAudioSendStream() override {
- RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
Stop();
call_->DestroyAudioSendStream(stream_);
}
+ void RecreateAudioSendStream(
+ const std::vector<webrtc::RtpExtension>& extensions) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ if (stream_) {
+ call_->DestroyAudioSendStream(stream_);
+ stream_ = nullptr;
+ }
+ config_.rtp.extensions = extensions;
+ RTC_DCHECK(!stream_);
+ stream_ = call_->CreateAudioSendStream(config_);
+ RTC_CHECK(stream_);
+ }
+
+ bool SendTelephoneEvent(int payload_type, uint8_t event,
+ uint32_t duration_ms) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(stream_);
+ return stream_->SendTelephoneEvent(payload_type, event, duration_ms);
+ }
+
+ webrtc::AudioSendStream::Stats GetStats() const {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(stream_);
+ return stream_->GetStats();
+ }
+
// Starts the rendering by setting a sink to the renderer to get data
// callback.
// This method is called on the libjingle worker thread.
// TODO(xians): Make sure Start() is called only once.
void Start(AudioRenderer* renderer) {
- RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(renderer);
if (renderer_) {
RTC_DCHECK(renderer_ == renderer);
@@ -1353,16 +1139,11 @@
renderer_ = renderer;
}
- webrtc::AudioSendStream::Stats GetStats() const {
- RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
- return stream_->GetStats();
- }
-
// Stops rendering by setting the sink of the renderer to nullptr. No data
// callback will be received after this method.
// This method is called on the libjingle worker thread.
void Stop() {
- RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (renderer_) {
renderer_->SetSink(nullptr);
renderer_ = nullptr;
@@ -1374,11 +1155,12 @@
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
- int number_of_channels,
+ size_t number_of_channels,
size_t number_of_frames) override {
+ RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());
RTC_DCHECK(voe_audio_transport_);
- voe_audio_transport_->OnData(channel_,
+ voe_audio_transport_->OnData(config_.voe_channel_id,
audio_data,
bits_per_sample,
sample_rate,
@@ -1389,7 +1171,7 @@
// Callback from the |renderer_| when it is going away. In case Start() has
// never been called, this callback won't be triggered.
void OnClose() override {
- RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Set |renderer_| to nullptr to make sure no more callback will get into
// the renderer.
renderer_ = nullptr;
@@ -1397,16 +1179,18 @@
// Accessor to the VoE channel ID.
int channel() const {
- RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
- return channel_;
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ return config_.voe_channel_id;
}
private:
- rtc::ThreadChecker signal_thread_checker_;
+ rtc::ThreadChecker worker_thread_checker_;
rtc::ThreadChecker audio_capture_thread_checker_;
- const int channel_ = -1;
webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
webrtc::Call* call_ = nullptr;
+ webrtc::AudioSendStream::Config config_;
+ // The stream is owned by WebRtcAudioSendStream and may be reallocated if
+ // configuration changes.
webrtc::AudioSendStream* stream_ = nullptr;
// Raw pointer to AudioRenderer owned by LocalAudioTrackHandler.
@@ -1419,80 +1203,163 @@
class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
public:
- explicit WebRtcAudioReceiveStream(int voe_channel_id)
- : channel_(voe_channel_id) {}
+ WebRtcAudioReceiveStream(int ch, uint32_t remote_ssrc, uint32_t local_ssrc,
+ bool use_combined_bwe, const std::string& sync_group,
+ const std::vector<webrtc::RtpExtension>& extensions,
+ webrtc::Call* call)
+ : call_(call),
+ config_() {
+ RTC_DCHECK_GE(ch, 0);
+ RTC_DCHECK(call);
+ config_.rtp.remote_ssrc = remote_ssrc;
+ config_.rtp.local_ssrc = local_ssrc;
+ config_.voe_channel_id = ch;
+ config_.sync_group = sync_group;
+ RecreateAudioReceiveStream(use_combined_bwe, extensions);
+ }
- int channel() { return channel_; }
+ ~WebRtcAudioReceiveStream() {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ call_->DestroyAudioReceiveStream(stream_);
+ }
+
+ void RecreateAudioReceiveStream(
+ const std::vector<webrtc::RtpExtension>& extensions) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RecreateAudioReceiveStream(config_.combined_audio_video_bwe, extensions);
+ }
+ void RecreateAudioReceiveStream(bool use_combined_bwe) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RecreateAudioReceiveStream(use_combined_bwe, config_.rtp.extensions);
+ }
+
+ webrtc::AudioReceiveStream::Stats GetStats() const {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(stream_);
+ return stream_->GetStats();
+ }
+
+ int channel() const {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ return config_.voe_channel_id;
+ }
+
+ void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ stream_->SetSink(std::move(sink));
+ }
private:
- int channel_;
+ void RecreateAudioReceiveStream(bool use_combined_bwe,
+ const std::vector<webrtc::RtpExtension>& extensions) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ if (stream_) {
+ call_->DestroyAudioReceiveStream(stream_);
+ stream_ = nullptr;
+ }
+ config_.rtp.extensions = extensions;
+ config_.combined_audio_video_bwe = use_combined_bwe;
+ RTC_DCHECK(!stream_);
+ stream_ = call_->CreateAudioReceiveStream(config_);
+ RTC_CHECK(stream_);
+ }
+
+ rtc::ThreadChecker worker_thread_checker_;
+ webrtc::Call* call_ = nullptr;
+ webrtc::AudioReceiveStream::Config config_;
+ // The stream is owned by WebRtcAudioReceiveStream and may be reallocated if
+ // configuration changes.
+ webrtc::AudioReceiveStream* stream_ = nullptr;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
};
-// WebRtcVoiceMediaChannel
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
const AudioOptions& options,
webrtc::Call* call)
- : engine_(engine),
- send_bitrate_setting_(false),
- send_bitrate_bps_(0),
- options_(),
- dtmf_allowed_(false),
- desired_playout_(false),
- nack_enabled_(false),
- playout_(false),
- typing_noise_detected_(false),
- desired_send_(SEND_NOTHING),
- send_(SEND_NOTHING),
- call_(call) {
+ : engine_(engine), call_(call) {
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
- RTC_DCHECK(nullptr != call);
+ RTC_DCHECK(call);
engine->RegisterChannel(this);
SetOptions(options);
}
WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel";
-
- // Remove any remaining send streams.
+ // TODO(solenberg): Should be able to delete the streams directly, without
+ // going through RemoveNnStream(), once stream objects handle
+ // all (de)configuration.
while (!send_streams_.empty()) {
RemoveSendStream(send_streams_.begin()->first);
}
-
- // Remove any remaining receive streams.
- while (!receive_channels_.empty()) {
- RemoveRecvStream(receive_channels_.begin()->first);
+ while (!recv_streams_.empty()) {
+ RemoveRecvStream(recv_streams_.begin()->first);
}
- RTC_DCHECK(receive_streams_.empty());
-
- // Unregister ourselves from the engine.
engine()->UnregisterChannel(this);
}
bool WebRtcVoiceMediaChannel::SetSendParameters(
const AudioSendParameters& params) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
+ << params.ToString();
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
- return (SetSendCodecs(params.codecs) &&
- SetSendRtpHeaderExtensions(params.extensions) &&
- SetMaxSendBandwidth(params.max_bandwidth_bps) &&
- SetOptions(params.options));
+
+ if (!SetSendCodecs(params.codecs)) {
+ return false;
+ }
+
+ if (!ValidateRtpExtensions(params.extensions)) {
+ return false;
+ }
+ std::vector<webrtc::RtpExtension> filtered_extensions =
+ FilterRtpExtensions(params.extensions,
+ webrtc::RtpExtension::IsSupportedForAudio, true);
+ if (send_rtp_extensions_ != filtered_extensions) {
+ send_rtp_extensions_.swap(filtered_extensions);
+ for (auto& it : send_streams_) {
+ it.second->RecreateAudioSendStream(send_rtp_extensions_);
+ }
+ }
+
+ if (!SetMaxSendBandwidth(params.max_bandwidth_bps)) {
+ return false;
+ }
+ return SetOptions(params.options);
}
bool WebRtcVoiceMediaChannel::SetRecvParameters(
const AudioRecvParameters& params) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: "
+ << params.ToString();
// TODO(pthatcher): Refactor this to be more clean now that we have
// all the information at once.
- return (SetRecvCodecs(params.codecs) &&
- SetRecvRtpHeaderExtensions(params.extensions));
+
+ if (!SetRecvCodecs(params.codecs)) {
+ return false;
+ }
+
+ if (!ValidateRtpExtensions(params.extensions)) {
+ return false;
+ }
+ std::vector<webrtc::RtpExtension> filtered_extensions =
+ FilterRtpExtensions(params.extensions,
+ webrtc::RtpExtension::IsSupportedForAudio, false);
+ if (recv_rtp_extensions_ != filtered_extensions) {
+ recv_rtp_extensions_.swap(filtered_extensions);
+ for (auto& it : recv_streams_) {
+ it.second->RecreateAudioReceiveStream(recv_rtp_extensions_);
+ }
+ }
+
+ return true;
}
bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "Setting voice channel options: "
<< options.ToString();
@@ -1503,26 +1370,27 @@
// on top. This means there is no way to "clear" options such that
// they go back to the engine default.
options_.SetAll(options);
-
- if (send_ != SEND_NOTHING) {
- if (!engine()->ApplyOptions(options_)) {
- LOG(LS_WARNING) <<
- "Failed to apply engine options during channel SetOptions.";
- return false;
- }
+ if (!engine()->ApplyOptions(options_)) {
+ LOG(LS_WARNING) <<
+ "Failed to apply engine options during channel SetOptions.";
+ return false;
}
if (dscp_option_changed) {
rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
- if (options_.dscp.GetWithDefaultIfUnset(false))
+ if (options_.dscp.value_or(false)) {
dscp = kAudioDscpValue;
+ }
if (MediaChannel::SetDscp(dscp) != 0) {
LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
}
}
// TODO(solenberg): Don't recreate unless options changed.
- RecreateAudioReceiveStreams();
+ for (auto& it : recv_streams_) {
+ it.second->RecreateAudioReceiveStream(
+ options_.combined_audio_video_bwe.value_or(false));
+ }
LOG(LS_INFO) << "Set voice channel options. Current options: "
<< options_.ToString();
@@ -1531,7 +1399,7 @@
bool WebRtcVoiceMediaChannel::SetRecvCodecs(
const std::vector<AudioCodec>& codecs) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Set the payload types to be used for incoming media.
LOG(LS_INFO) << "Setting receive voice codecs.";
@@ -1568,7 +1436,26 @@
PausePlayout();
}
- bool result = SetRecvCodecsInternal(new_codecs);
+ bool result = true;
+ for (const AudioCodec& codec : new_codecs) {
+ webrtc::CodecInst voe_codec;
+ if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
+ LOG(LS_INFO) << ToString(codec);
+ voe_codec.pltype = codec.id;
+ for (const auto& ch : recv_streams_) {
+ if (engine()->voe()->codec()->SetRecPayloadType(
+ ch.second->channel(), voe_codec) == -1) {
+ LOG_RTCERR2(SetRecPayloadType, ch.second->channel(),
+ ToString(voe_codec));
+ result = false;
+ }
+ }
+ } else {
+ LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
+ result = false;
+ break;
+ }
+ }
if (result) {
recv_codecs_ = codecs;
}
@@ -1588,7 +1475,7 @@
engine()->voe()->codec()->SetFECStatus(channel, false);
// Scan through the list to figure out the codec to use for sending, along
- // with the proper configuration for VAD and DTMF.
+ // with the proper configuration for VAD.
bool found_send_codec = false;
webrtc::CodecInst send_codec;
memset(&send_codec, 0, sizeof(send_codec));
@@ -1603,7 +1490,7 @@
// Ignore codecs we don't know about. The negotiation step should prevent
// this, but double-check to be sure.
webrtc::CodecInst voe_codec;
- if (!engine()->FindWebRtcCodec(codec, &voe_codec)) {
+ if (!WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
continue;
}
@@ -1644,7 +1531,7 @@
// Set packet size if the AudioCodec param kCodecParamPTime is set.
int ptime_ms = 0;
if (codec.GetParam(kCodecParamPTime, &ptime_ms)) {
- if (!SetPTimeAsPacketSize(&send_codec, ptime_ms)) {
+ if (!WebRtcVoiceCodecs::SetPTimeAsPacketSize(&send_codec, ptime_ms)) {
LOG(LS_WARNING) << "Failed to set packet size for codec "
<< send_codec.plname;
return false;
@@ -1687,7 +1574,7 @@
// Set Opus internal DTX.
LOG(LS_INFO) << "Attempt to "
- << GetEnableString(enable_opus_dtx)
+ << (enable_opus_dtx ? "enable" : "disable")
<< " Opus DTX on channel "
<< channel;
if (engine()->voe()->codec()->SetOpusDtx(channel, enable_opus_dtx)) {
@@ -1717,25 +1604,17 @@
SetSendBitrateInternal(send_bitrate_bps_);
}
- // Loop through the codecs list again to config the telephone-event/CN codec.
+ // Loop through the codecs list again to config the CN codec.
for (const AudioCodec& codec : codecs) {
// Ignore codecs we don't know about. The negotiation step should prevent
// this, but double-check to be sure.
webrtc::CodecInst voe_codec;
- if (!engine()->FindWebRtcCodec(codec, &voe_codec)) {
+ if (!WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
continue;
}
- // Find the DTMF telephone event "codec" and tell VoiceEngine channels
- // about it.
- if (IsCodec(codec, kDtmfCodecName)) {
- if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
- channel, codec.id) == -1) {
- LOG_RTCERR2(SetSendTelephoneEventPayloadType, channel, codec.id);
- return false;
- }
- } else if (IsCodec(codec, kCnCodecName)) {
+ if (IsCodec(codec, kCnCodecName)) {
// Turn voice activity detection/comfort noise on if supported.
// Set the wideband CN payload type appropriately.
// (narrowband always uses the static payload type 13).
@@ -1789,13 +1668,17 @@
bool WebRtcVoiceMediaChannel::SetSendCodecs(
const std::vector<AudioCodec>& codecs) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ // TODO(solenberg): Validate input - that payload types don't overlap, are
+ // within range, filter out codecs we don't support,
+ // redundant codecs etc.
- dtmf_allowed_ = false;
+ // Find the DTMF telephone event "codec" payload type.
+ dtmf_payload_type_ = rtc::Optional<int>();
for (const AudioCodec& codec : codecs) {
- // Find the DTMF telephone event "codec".
if (IsCodec(codec, kDtmfCodecName)) {
- dtmf_allowed_ = true;
+ dtmf_payload_type_ = rtc::Optional<int>(codec.id);
+ break;
}
}
@@ -1808,7 +1691,7 @@
}
// Set nack status on receive channels and update |nack_enabled_|.
- for (const auto& ch : receive_channels_) {
+ for (const auto& ch : recv_streams_) {
SetNack(ch.second->channel(), nack_enabled_);
}
@@ -1844,106 +1727,6 @@
return true;
}
-bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (receive_extensions_ == extensions) {
- return true;
- }
-
- for (const auto& ch : receive_channels_) {
- if (!SetChannelRecvRtpHeaderExtensions(ch.second->channel(), extensions)) {
- return false;
- }
- }
-
- receive_extensions_ = extensions;
-
- // Recreate AudioReceiveStream:s.
- {
- std::vector<webrtc::RtpExtension> exts;
-
- const RtpHeaderExtension* audio_level_extension =
- FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
- if (audio_level_extension) {
- exts.push_back({
- kRtpAudioLevelHeaderExtension, audio_level_extension->id});
- }
-
- const RtpHeaderExtension* send_time_extension =
- FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
- if (send_time_extension) {
- exts.push_back({
- kRtpAbsoluteSenderTimeHeaderExtension, send_time_extension->id});
- }
-
- recv_rtp_extensions_.swap(exts);
- RecreateAudioReceiveStreams();
- }
-
- return true;
-}
-
-bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
- int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
- const RtpHeaderExtension* audio_level_extension =
- FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
- if (!SetHeaderExtension(
- &webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus, channel_id,
- audio_level_extension)) {
- return false;
- }
-
- const RtpHeaderExtension* send_time_extension =
- FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
- if (!SetHeaderExtension(
- &webrtc::VoERTP_RTCP::SetReceiveAbsoluteSenderTimeStatus, channel_id,
- send_time_extension)) {
- return false;
- }
-
- return true;
-}
-
-bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (send_extensions_ == extensions) {
- return true;
- }
-
- for (const auto& ch : send_streams_) {
- if (!SetChannelSendRtpHeaderExtensions(ch.second->channel(), extensions)) {
- return false;
- }
- }
-
- send_extensions_ = extensions;
- return true;
-}
-
-bool WebRtcVoiceMediaChannel::SetChannelSendRtpHeaderExtensions(
- int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
- const RtpHeaderExtension* audio_level_extension =
- FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
-
- if (!SetHeaderExtension(
- &webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, channel_id,
- audio_level_extension)) {
- return false;
- }
-
- const RtpHeaderExtension* send_time_extension =
- FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
- if (!SetHeaderExtension(
- &webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, channel_id,
- send_time_extension)) {
- return false;
- }
-
- return true;
-}
-
bool WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
desired_playout_ = playout;
return ChangePlayout(desired_playout_);
@@ -1958,12 +1741,12 @@
}
bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (playout_ == playout) {
return true;
}
- for (const auto& ch : receive_channels_) {
+ for (const auto& ch : recv_streams_) {
if (!SetPlayout(ch.second->channel(), playout)) {
LOG(LS_ERROR) << "SetPlayout " << playout << " on channel "
<< ch.second->channel() << " failed";
@@ -1995,7 +1778,7 @@
return true;
}
- // Apply channel specific options.
+ // Apply channel specific options when channel is enabled for sending.
if (send == SEND_MICROPHONE) {
engine()->ApplyOptions(options_);
}
@@ -2007,13 +1790,6 @@
}
}
- // Clear up the options after stopping sending. Since we may previously have
- // applied the channel specific options, now apply the original options stored
- // in WebRtcVoiceEngine.
- if (send == SEND_NOTHING) {
- engine()->ApplyOptions(engine()->GetOptions());
- }
-
send_ = send;
return true;
}
@@ -2039,7 +1815,7 @@
bool enable,
const AudioOptions* options,
AudioRenderer* renderer) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// TODO(solenberg): The state change should be fully rolled back if any one of
// these calls fail.
if (!SetLocalRenderer(ssrc, renderer)) {
@@ -2068,7 +1844,7 @@
return id;
}
-bool WebRtcVoiceMediaChannel::DeleteChannel(int channel) {
+bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) {
if (engine()->voe()->network()->DeRegisterExternalTransport(channel) == -1) {
LOG_RTCERR1(DeRegisterExternalTransport, channel);
}
@@ -2080,7 +1856,7 @@
}
bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
uint32_t ssrc = sp.first_ssrc();
@@ -2097,33 +1873,12 @@
return false;
}
- // Enable RTCP (for quality stats and feedback messages).
- if (engine()->voe()->rtp()->SetRTCPStatus(channel, true) == -1) {
- LOG_RTCERR2(SetRTCPStatus, channel, 1);
- }
-
- SetChannelSendRtpHeaderExtensions(channel, send_extensions_);
-
- // Set the local (send) SSRC.
- if (engine()->voe()->rtp()->SetLocalSSRC(channel, ssrc) == -1) {
- LOG_RTCERR2(SetLocalSSRC, channel, ssrc);
- DeleteChannel(channel);
- return false;
- }
-
- if (engine()->voe()->rtp()->SetRTCP_CNAME(channel, sp.cname.c_str()) == -1) {
- LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
- DeleteChannel(channel);
- return false;
- }
-
// Save the channel to send_streams_, so that RemoveSendStream() can still
// delete the channel in case failure happens below.
webrtc::AudioTransport* audio_transport =
engine()->voe()->base()->audio_transport();
- send_streams_.insert(
- std::make_pair(ssrc,
- new WebRtcAudioSendStream(channel, audio_transport, ssrc, call_)));
+ send_streams_.insert(std::make_pair(ssrc, new WebRtcAudioSendStream(
+ channel, audio_transport, ssrc, sp.cname, send_rtp_extensions_, call_)));
// Set the current codecs to be used for the new channel. We need to do this
// after adding the channel to send_channels_, because of how max bitrate is
@@ -2138,10 +1893,10 @@
// with the same SSRC in order to send receiver reports.
if (send_streams_.size() == 1) {
receiver_reports_ssrc_ = ssrc;
- for (const auto& ch : receive_channels_) {
- int recv_channel = ch.second->channel();
+ for (const auto& stream : recv_streams_) {
+ int recv_channel = stream.second->channel();
if (engine()->voe()->rtp()->SetLocalSSRC(recv_channel, ssrc) != 0) {
- LOG_RTCERR2(SetLocalSSRC, ch.second->channel(), ssrc);
+ LOG_RTCERR2(SetLocalSSRC, recv_channel, ssrc);
return false;
}
engine()->voe()->base()->AssociateSendChannel(recv_channel, channel);
@@ -2154,7 +1909,9 @@
}
bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
+
auto it = send_streams_.find(ssrc);
if (it == send_streams_.end()) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
@@ -2165,15 +1922,12 @@
int channel = it->second->channel();
ChangeSend(channel, SEND_NOTHING);
- // Delete the WebRtcVoiceChannelRenderer object connected to the channel,
- // this will disconnect the audio renderer with the send channel.
- delete it->second;
- send_streams_.erase(it);
-
- // Clean up and delete the send channel.
+ // Clean up and delete the send stream+channel.
LOG(LS_INFO) << "Removing audio send stream " << ssrc
<< " with VoiceEngine channel #" << channel << ".";
- if (!DeleteChannel(channel)) {
+ delete it->second;
+ send_streams_.erase(it);
+ if (!DeleteVoEChannel(channel)) {
return false;
}
if (send_streams_.empty()) {
@@ -2183,14 +1937,14 @@
}
bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
if (!ValidateStreamParams(sp)) {
return false;
}
- uint32_t ssrc = sp.first_ssrc();
+ const uint32_t ssrc = sp.first_ssrc();
if (ssrc == 0) {
LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
return false;
@@ -2202,37 +1956,43 @@
RemoveRecvStream(ssrc);
}
- if (receive_channels_.find(ssrc) != receive_channels_.end()) {
+ if (GetReceiveChannelId(ssrc) != -1) {
LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
return false;
}
- RTC_DCHECK(receive_stream_params_.find(ssrc) == receive_stream_params_.end());
// Create a new channel for receiving audio data.
- int channel = CreateVoEChannel();
+ const int channel = CreateVoEChannel();
if (channel == -1) {
return false;
}
- if (!ConfigureRecvChannel(channel)) {
- DeleteChannel(channel);
- return false;
+
+ // Turn off all supported codecs.
+ // TODO(solenberg): Remove once "no codecs" is the default state of a stream.
+ for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
+ voe_codec.pltype = -1;
+ if (engine()->voe()->codec()->SetRecPayloadType(channel, voe_codec) == -1) {
+ LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
+ DeleteVoEChannel(channel);
+ return false;
+ }
}
- WebRtcAudioReceiveStream* stream = new WebRtcAudioReceiveStream(channel);
- receive_channels_.insert(std::make_pair(ssrc, stream));
- receive_stream_params_[ssrc] = sp;
- AddAudioReceiveStream(ssrc);
+ // Only enable those configured for this channel.
+ for (const auto& codec : recv_codecs_) {
+ webrtc::CodecInst voe_codec;
+ if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
+ voe_codec.pltype = codec.id;
+ if (engine()->voe()->codec()->SetRecPayloadType(
+ channel, voe_codec) == -1) {
+ LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
+ DeleteVoEChannel(channel);
+ return false;
+ }
+ }
+ }
- LOG(LS_INFO) << "New audio stream " << ssrc
- << " registered to VoiceEngine channel #"
- << channel << ".";
- return true;
-}
-
-bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
-
- int send_channel = GetSendChannelId(receiver_reports_ssrc_);
+ const int send_channel = GetSendChannelId(receiver_reports_ssrc_);
if (send_channel != -1) {
// Associate receive channel with first send channel (so the receive channel
// can obtain RTT from the send channel)
@@ -2240,76 +2000,43 @@
LOG(LS_INFO) << "VoiceEngine channel #" << channel
<< " is associated with channel #" << send_channel << ".";
}
- if (engine()->voe()->rtp()->SetLocalSSRC(channel,
- receiver_reports_ssrc_) == -1) {
- LOG_RTCERR1(SetLocalSSRC, channel);
- return false;
- }
- // Turn off all supported codecs.
- int ncodecs = engine()->voe()->codec()->NumOfCodecs();
- for (int i = 0; i < ncodecs; ++i) {
- webrtc::CodecInst voe_codec;
- if (engine()->voe()->codec()->GetCodec(i, voe_codec) != -1) {
- voe_codec.pltype = -1;
- if (engine()->voe()->codec()->SetRecPayloadType(
- channel, voe_codec) == -1) {
- LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
- return false;
- }
- }
- }
-
- // Only enable those configured for this channel.
- for (const auto& codec : recv_codecs_) {
- webrtc::CodecInst voe_codec;
- if (engine()->FindWebRtcCodec(codec, &voe_codec)) {
- voe_codec.pltype = codec.id;
- if (engine()->voe()->codec()->SetRecPayloadType(
- channel, voe_codec) == -1) {
- LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
- return false;
- }
- }
- }
+ recv_streams_.insert(std::make_pair(ssrc, new WebRtcAudioReceiveStream(
+ channel, ssrc, receiver_reports_ssrc_,
+ options_.combined_audio_video_bwe.value_or(false), sp.sync_label,
+ recv_rtp_extensions_, call_)));
SetNack(channel, nack_enabled_);
-
- // Set RTP header extension for the new channel.
- if (!SetChannelRecvRtpHeaderExtensions(channel, receive_extensions_)) {
- return false;
- }
-
SetPlayout(channel, playout_);
+
return true;
}
bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
- auto it = receive_channels_.find(ssrc);
- if (it == receive_channels_.end()) {
+ const auto it = recv_streams_.find(ssrc);
+ if (it == recv_streams_.end()) {
LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
<< " which doesn't exist.";
return false;
}
- RemoveAudioReceiveStream(ssrc);
- receive_stream_params_.erase(ssrc);
-
- const int channel = it->second->channel();
- delete it->second;
- receive_channels_.erase(it);
-
// Deregister default channel, if that's the one being destroyed.
if (IsDefaultRecvStream(ssrc)) {
default_recv_ssrc_ = -1;
}
- LOG(LS_INFO) << "Removing audio stream " << ssrc
+ const int channel = it->second->channel();
+
+ // Clean up and delete the receive stream+channel.
+ LOG(LS_INFO) << "Removing audio receive stream " << ssrc
<< " with VoiceEngine channel #" << channel << ".";
- return DeleteChannel(channel);
+ it->second->SetRawAudioSink(nullptr);
+ delete it->second;
+ recv_streams_.erase(it);
+ return DeleteVoEChannel(channel);
}
bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32_t ssrc,
@@ -2337,9 +2064,9 @@
bool WebRtcVoiceMediaChannel::GetActiveStreams(
AudioInfo::StreamList* actives) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
actives->clear();
- for (const auto& ch : receive_channels_) {
+ for (const auto& ch : recv_streams_) {
int level = GetOutputLevel(ch.second->channel());
if (level > 0) {
actives->push_back(std::make_pair(ch.first, level));
@@ -2349,9 +2076,9 @@
}
int WebRtcVoiceMediaChannel::GetOutputLevel() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int highest = 0;
- for (const auto& ch : receive_channels_) {
+ for (const auto& ch : recv_streams_) {
highest = std::max(GetOutputLevel(ch.second->channel()), highest);
}
return highest;
@@ -2383,7 +2110,7 @@
}
bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
if (ssrc == 0) {
default_recv_volume_ = volume;
if (default_recv_ssrc_ == -1) {
@@ -2408,64 +2135,48 @@
}
bool WebRtcVoiceMediaChannel::CanInsertDtmf() {
- return dtmf_allowed_;
+ return dtmf_payload_type_ ? true : false;
}
-bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
- int event,
- int duration,
- int flags) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- if (!dtmf_allowed_) {
+bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, int event,
+ int duration) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ LOG(LS_INFO) << "WebRtcVoiceMediaChannel::InsertDtmf";
+ if (!dtmf_payload_type_) {
return false;
}
- // Send the event.
- if (flags & cricket::DF_SEND) {
- int channel = -1;
- if (ssrc == 0) {
- if (send_streams_.size() > 0) {
- channel = send_streams_.begin()->second->channel();
- }
- } else {
- channel = GetSendChannelId(ssrc);
- }
- if (channel == -1) {
- LOG(LS_WARNING) << "InsertDtmf - The specified ssrc "
- << ssrc << " is not in use.";
- return false;
- }
- // Send DTMF using out-of-band DTMF. ("true", as 3rd arg)
- if (engine()->voe()->dtmf()->SendTelephoneEvent(
- channel, event, true, duration) == -1) {
- LOG_RTCERR4(SendTelephoneEvent, channel, event, true, duration);
- return false;
- }
+ // Figure out which WebRtcAudioSendStream to send the event on.
+ auto it = ssrc != 0 ? send_streams_.find(ssrc) : send_streams_.begin();
+ if (it == send_streams_.end()) {
+ LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
+ return false;
}
-
- // Play the event.
- if (flags & cricket::DF_PLAY) {
- // Play DTMF tone locally.
- if (engine()->voe()->dtmf()->PlayDtmfTone(event, duration) == -1) {
- LOG_RTCERR2(PlayDtmfTone, event, duration);
- return false;
- }
+ if (event < kMinTelephoneEventCode ||
+ event > kMaxTelephoneEventCode) {
+ LOG(LS_WARNING) << "DTMF event code " << event << " out of range.";
+ return false;
}
-
- return true;
+ if (duration < kMinTelephoneEventDuration ||
+ duration > kMaxTelephoneEventDuration) {
+ LOG(LS_WARNING) << "DTMF event duration " << duration << " out of range.";
+ return false;
+ }
+ return it->second->SendTelephoneEvent(*dtmf_payload_type_, event, duration);
}
void WebRtcVoiceMediaChannel::OnPacketReceived(
rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
uint32_t ssrc = 0;
if (!GetRtpSsrc(packet->data(), packet->size(), &ssrc)) {
return;
}
- if (receive_channels_.empty()) {
- // Create new channel, which will be the default receive channel.
+ // If we don't have a default channel, and the SSRC is unknown, create a
+ // default channel.
+ if (default_recv_ssrc_ == -1 && GetReceiveChannelId(ssrc) == -1) {
StreamParams sp;
sp.ssrcs.push_back(ssrc);
LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
@@ -2485,7 +2196,13 @@
reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
webrtc_packet_time);
if (webrtc::PacketReceiver::DELIVERY_OK != delivery_result) {
- return;
+ // If the SSRC is unknown here, route it to the default channel, if we have
+ // one. See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
+ if (default_recv_ssrc_ == -1) {
+ return;
+ } else {
+ ssrc = default_recv_ssrc_;
+ }
}
// Find the channel to send this packet to. It must exist since webrtc::Call
@@ -2500,7 +2217,7 @@
void WebRtcVoiceMediaChannel::OnRtcpReceived(
rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
// Forward packet to Call as well.
const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
@@ -2542,7 +2259,7 @@
}
bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
int channel = GetSendChannelId(ssrc);
if (channel == -1) {
LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
@@ -2601,7 +2318,7 @@
return true;
webrtc::CodecInst codec = *send_codec_;
- bool is_multi_rate = IsCodecMultiRate(codec);
+ bool is_multi_rate = WebRtcVoiceCodecs::IsCodecMultiRate(codec);
if (is_multi_rate) {
// If codec is multi-rate then just set the bitrate.
@@ -2629,7 +2346,7 @@
}
bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
RTC_DCHECK(info);
// Get SSRC and stats for each sender.
@@ -2652,15 +2369,14 @@
sinfo.echo_delay_std_ms = stats.echo_delay_std_ms;
sinfo.echo_return_loss = stats.echo_return_loss;
sinfo.echo_return_loss_enhancement = stats.echo_return_loss_enhancement;
- sinfo.typing_noise_detected = typing_noise_detected_;
- // TODO(solenberg): Move to AudioSendStream.
- // sinfo.typing_noise_detected = stats.typing_noise_detected;
+ sinfo.typing_noise_detected =
+ (send_ == SEND_NOTHING ? false : stats.typing_noise_detected);
info->senders.push_back(sinfo);
}
// Get SSRC and stats for each receiver.
RTC_DCHECK(info->receivers.size() == 0);
- for (const auto& stream : receive_streams_) {
+ for (const auto& stream : recv_streams_) {
webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
VoiceReceiverInfo rinfo;
rinfo.add_ssrc(stats.remote_ssrc);
@@ -2694,15 +2410,17 @@
return true;
}
-void WebRtcVoiceMediaChannel::OnError(int error) {
- if (send_ == SEND_NOTHING) {
+void WebRtcVoiceMediaChannel::SetRawAudioSink(
+ uint32_t ssrc,
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink";
+ const auto it = recv_streams_.find(ssrc);
+ if (it == recv_streams_.end()) {
+ LOG(LS_WARNING) << "SetRawAudioSink: no recv stream" << ssrc;
return;
}
- if (error == VE_TYPING_NOISE_WARNING) {
- typing_noise_detected_ = true;
- } else if (error == VE_TYPING_NOISE_OFF_WARNING) {
- typing_noise_detected_ = false;
- }
+ it->second->SetRawAudioSink(std::move(sink));
}
int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
@@ -2712,16 +2430,16 @@
}
int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- const auto it = receive_channels_.find(ssrc);
- if (it != receive_channels_.end()) {
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+ const auto it = recv_streams_.find(ssrc);
+ if (it != recv_streams_.end()) {
return it->second->channel();
}
return -1;
}
int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
const auto it = send_streams_.find(ssrc);
if (it != send_streams_.end()) {
return it->second->channel();
@@ -2762,7 +2480,7 @@
if (codec.id == red_pt) {
// If we find the right codec, that will be the codec we pass to
// SetSendCodec, with the desired payload type.
- if (engine()->FindWebRtcCodec(codec, send_codec)) {
+ if (WebRtcVoiceEngine::ToCodecInst(codec, send_codec)) {
return true;
} else {
break;
@@ -2786,117 +2504,6 @@
}
return true;
}
-
-// Convert VoiceEngine error code into VoiceMediaChannel::Error enum.
-VoiceMediaChannel::Error
- WebRtcVoiceMediaChannel::WebRtcErrorToChannelError(int err_code) {
- switch (err_code) {
- case 0:
- return ERROR_NONE;
- case VE_CANNOT_START_RECORDING:
- case VE_MIC_VOL_ERROR:
- case VE_GET_MIC_VOL_ERROR:
- case VE_CANNOT_ACCESS_MIC_VOL:
- return ERROR_REC_DEVICE_OPEN_FAILED;
- case VE_SATURATION_WARNING:
- return ERROR_REC_DEVICE_SATURATION;
- case VE_REC_DEVICE_REMOVED:
- return ERROR_REC_DEVICE_REMOVED;
- case VE_RUNTIME_REC_WARNING:
- case VE_RUNTIME_REC_ERROR:
- return ERROR_REC_RUNTIME_ERROR;
- case VE_CANNOT_START_PLAYOUT:
- case VE_SPEAKER_VOL_ERROR:
- case VE_GET_SPEAKER_VOL_ERROR:
- case VE_CANNOT_ACCESS_SPEAKER_VOL:
- return ERROR_PLAY_DEVICE_OPEN_FAILED;
- case VE_RUNTIME_PLAY_WARNING:
- case VE_RUNTIME_PLAY_ERROR:
- return ERROR_PLAY_RUNTIME_ERROR;
- case VE_TYPING_NOISE_WARNING:
- return ERROR_REC_TYPING_NOISE_DETECTED;
- default:
- return VoiceMediaChannel::ERROR_OTHER;
- }
-}
-
-bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
- int channel_id, const RtpHeaderExtension* extension) {
- bool enable = false;
- int id = 0;
- std::string uri;
- if (extension) {
- enable = true;
- id = extension->id;
- uri = extension->uri;
- }
- if ((engine()->voe()->rtp()->*setter)(channel_id, enable, id) != 0) {
- LOG_RTCERR4(*setter, uri, channel_id, enable, id);
- return false;
- }
- return true;
-}
-
-void WebRtcVoiceMediaChannel::RecreateAudioReceiveStreams() {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- for (const auto& it : receive_channels_) {
- RemoveAudioReceiveStream(it.first);
- }
- for (const auto& it : receive_channels_) {
- AddAudioReceiveStream(it.first);
- }
-}
-
-void WebRtcVoiceMediaChannel::AddAudioReceiveStream(uint32_t ssrc) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- WebRtcAudioReceiveStream* stream = receive_channels_[ssrc];
- RTC_DCHECK(stream != nullptr);
- RTC_DCHECK(receive_streams_.find(ssrc) == receive_streams_.end());
- webrtc::AudioReceiveStream::Config config;
- config.rtp.remote_ssrc = ssrc;
- // Only add RTP extensions if we support combined A/V BWE.
- config.rtp.extensions = recv_rtp_extensions_;
- config.combined_audio_video_bwe =
- options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false);
- config.voe_channel_id = stream->channel();
- config.sync_group = receive_stream_params_[ssrc].sync_label;
- webrtc::AudioReceiveStream* s = call_->CreateAudioReceiveStream(config);
- receive_streams_.insert(std::make_pair(ssrc, s));
-}
-
-void WebRtcVoiceMediaChannel::RemoveAudioReceiveStream(uint32_t ssrc) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- auto stream_it = receive_streams_.find(ssrc);
- if (stream_it != receive_streams_.end()) {
- call_->DestroyAudioReceiveStream(stream_it->second);
- receive_streams_.erase(stream_it);
- }
-}
-
-bool WebRtcVoiceMediaChannel::SetRecvCodecsInternal(
- const std::vector<AudioCodec>& new_codecs) {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- for (const AudioCodec& codec : new_codecs) {
- webrtc::CodecInst voe_codec;
- if (engine()->FindWebRtcCodec(codec, &voe_codec)) {
- LOG(LS_INFO) << ToString(codec);
- voe_codec.pltype = codec.id;
- for (const auto& ch : receive_channels_) {
- if (engine()->voe()->codec()->SetRecPayloadType(
- ch.second->channel(), voe_codec) == -1) {
- LOG_RTCERR2(SetRecPayloadType, ch.second->channel(),
- ToString(voe_codec));
- return false;
- }
- }
- } else {
- LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
- return false;
- }
- }
- return true;
-}
-
} // namespace cricket
#endif // HAVE_WEBRTC_VOICE
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 1cf05e7..0f2f59e 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -29,7 +29,6 @@
#define TALK_MEDIA_WEBRTCVOICEENGINE_H_
#include <map>
-#include <set>
#include <string>
#include <vector>
@@ -37,9 +36,8 @@
#include "talk/media/webrtc/webrtccommon.h"
#include "talk/media/webrtc/webrtcvoe.h"
#include "talk/session/media/channel.h"
+#include "webrtc/audio_state.h"
#include "webrtc/base/buffer.h"
-#include "webrtc/base/byteorder.h"
-#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/stream.h"
#include "webrtc/base/thread_checker.h"
@@ -51,43 +49,34 @@
class AudioDeviceModule;
class AudioRenderer;
-class VoETraceWrapper;
class VoEWrapper;
class WebRtcVoiceMediaChannel;
// WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
// It uses the WebRtc VoiceEngine library for audio handling.
-class WebRtcVoiceEngine
- : public webrtc::VoiceEngineObserver,
- public webrtc::TraceCallback {
+class WebRtcVoiceEngine final : public webrtc::TraceCallback {
friend class WebRtcVoiceMediaChannel;
-
public:
+ // Exposed for the WVoE/MC unit test.
+ static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out);
+
WebRtcVoiceEngine();
// Dependency injection for testing.
- WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
+ explicit WebRtcVoiceEngine(VoEWrapper* voe_wrapper);
~WebRtcVoiceEngine();
bool Init(rtc::Thread* worker_thread);
void Terminate();
- webrtc::VoiceEngine* GetVoE() { return voe()->engine(); }
+ rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
VoiceMediaChannel* CreateChannel(webrtc::Call* call,
const AudioOptions& options);
- AudioOptions GetOptions() const { return options_; }
- bool SetOptions(const AudioOptions& options);
- bool SetDevices(const Device* in_device, const Device* out_device);
bool GetOutputVolume(int* level);
bool SetOutputVolume(int level);
int GetInputLevel();
const std::vector<AudioCodec>& codecs();
- bool FindCodec(const AudioCodec& codec);
- bool FindWebRtcCodec(const AudioCodec& codec, webrtc::CodecInst* gcodec);
-
- const std::vector<RtpHeaderExtension>& rtp_header_extensions() const;
-
- void SetLogging(int min_sev, const char* filter);
+ RtpCapabilities GetCapabilities() const;
// For tracking WebRtc channels. Needed because we have to pause them
// all when switching devices.
@@ -120,68 +109,49 @@
private:
void Construct();
- void ConstructCodecs();
- bool GetVoeCodec(int index, webrtc::CodecInst* codec);
bool InitInternal();
- void SetTraceFilter(int filter);
- void SetTraceOptions(const std::string& options);
// Every option that is "set" will be applied. Every option not "set" will be
// ignored. This allows us to selectively turn on and off different options
// easily at any time.
bool ApplyOptions(const AudioOptions& options);
+ void SetDefaultDevices();
// webrtc::TraceCallback:
void Print(webrtc::TraceLevel level, const char* trace, int length) override;
- // webrtc::VoiceEngineObserver:
- void CallbackOnError(int channel_id, int errCode) override;
-
- // Given the device type, name, and id, find device id. Return true and
- // set the output parameter rtc_id if successful.
- bool FindWebRtcAudioDeviceId(
- bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
-
void StartAecDump(const std::string& filename);
int CreateVoEChannel();
- static const int kDefaultLogSeverity = rtc::LS_WARNING;
+ rtc::ThreadChecker signal_thread_checker_;
+ rtc::ThreadChecker worker_thread_checker_;
// The primary instance of WebRtc VoiceEngine.
rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
- rtc::scoped_ptr<VoETraceWrapper> tracing_;
+ rtc::scoped_refptr<webrtc::AudioState> audio_state_;
// The external audio device manager
- webrtc::AudioDeviceModule* adm_;
- int log_filter_;
- std::string log_options_;
- bool is_dumping_aec_;
+ webrtc::AudioDeviceModule* adm_ = nullptr;
std::vector<AudioCodec> codecs_;
- std::vector<RtpHeaderExtension> rtp_header_extensions_;
std::vector<WebRtcVoiceMediaChannel*> channels_;
- // channels_ can be read from WebRtc callback thread. We need a lock on that
- // callback as well as the RegisterChannel/UnregisterChannel.
- rtc::CriticalSection channels_cs_;
- webrtc::AgcConfig default_agc_config_;
-
webrtc::Config voe_config_;
+ bool initialized_ = false;
+ bool is_dumping_aec_ = false;
- bool initialized_;
- AudioOptions options_;
-
+ webrtc::AgcConfig default_agc_config_;
// Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
// values, and apply them in case they are missing in the audio options. We
// need to do this because SetExtraOptions() will revert to defaults for
// options which are not provided.
- Settable<bool> extended_filter_aec_;
- Settable<bool> delay_agnostic_aec_;
- Settable<bool> experimental_ns_;
+ rtc::Optional<bool> extended_filter_aec_;
+ rtc::Optional<bool> delay_agnostic_aec_;
+ rtc::Optional<bool> experimental_ns_;
RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcVoiceEngine);
};
// WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
// WebRtc Voice Engine.
-class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
- public webrtc::Transport {
+class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
+ public webrtc::Transport {
public:
WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
const AudioOptions& options,
@@ -217,7 +187,7 @@
bool SetOutputVolume(uint32_t ssrc, double volume) override;
bool CanInsertDtmf() override;
- bool InsertDtmf(uint32_t ssrc, int event, int duration, int flags) override;
+ bool InsertDtmf(uint32_t ssrc, int event, int duration) override;
void OnPacketReceived(rtc::Buffer* packet,
const rtc::PacketTime& packet_time) override;
@@ -226,6 +196,10 @@
void OnReadyToSend(bool ready) override {}
bool GetStats(VoiceMediaInfo* info) override;
+ void SetRawAudioSink(
+ uint32_t ssrc,
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
+
// implements Transport interface
bool SendRtp(const uint8_t* data,
size_t len,
@@ -243,20 +217,14 @@
return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
}
- void OnError(int error);
-
int GetReceiveChannelId(uint32_t ssrc) const;
int GetSendChannelId(uint32_t ssrc) const;
private:
bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
- bool SetSendRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions);
bool SetOptions(const AudioOptions& options);
bool SetMaxSendBandwidth(int bps);
bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
- bool SetRecvRtpHeaderExtensions(
- const std::vector<RtpHeaderExtension>& extensions);
bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer);
bool MuteStream(uint32_t ssrc, bool mute);
@@ -267,82 +235,55 @@
const std::vector<AudioCodec>& all_codecs,
webrtc::CodecInst* send_codec);
bool SetPlayout(int channel, bool playout);
- static Error WebRtcErrorToChannelError(int err_code);
-
- typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool,
- unsigned char);
-
void SetNack(int channel, bool nack_enabled);
bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
bool ChangePlayout(bool playout);
bool ChangeSend(SendFlags send);
bool ChangeSend(int channel, SendFlags send);
- bool ConfigureRecvChannel(int channel);
int CreateVoEChannel();
- bool DeleteChannel(int channel);
+ bool DeleteVoEChannel(int channel);
bool IsDefaultRecvStream(uint32_t ssrc) {
return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
}
bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
bool SetSendBitrateInternal(int bps);
- bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
- const RtpHeaderExtension* extension);
- void RecreateAudioReceiveStreams();
- void AddAudioReceiveStream(uint32_t ssrc);
- void RemoveAudioReceiveStream(uint32_t ssrc);
- bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs);
+ rtc::ThreadChecker worker_thread_checker_;
- bool SetChannelRecvRtpHeaderExtensions(
- int channel_id,
- const std::vector<RtpHeaderExtension>& extensions);
- bool SetChannelSendRtpHeaderExtensions(
- int channel_id,
- const std::vector<RtpHeaderExtension>& extensions);
-
- rtc::ThreadChecker thread_checker_;
-
- WebRtcVoiceEngine* const engine_;
+ WebRtcVoiceEngine* const engine_ = nullptr;
std::vector<AudioCodec> recv_codecs_;
std::vector<AudioCodec> send_codecs_;
rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
- bool send_bitrate_setting_;
- int send_bitrate_bps_;
+ bool send_bitrate_setting_ = false;
+ int send_bitrate_bps_ = 0;
AudioOptions options_;
- bool dtmf_allowed_;
- bool desired_playout_;
- bool nack_enabled_;
- bool playout_;
- bool typing_noise_detected_;
- SendFlags desired_send_;
- SendFlags send_;
- webrtc::Call* const call_;
+ rtc::Optional<int> dtmf_payload_type_;
+ bool desired_playout_ = false;
+ bool nack_enabled_ = false;
+ bool playout_ = false;
+ SendFlags desired_send_ = SEND_NOTHING;
+ SendFlags send_ = SEND_NOTHING;
+ webrtc::Call* const call_ = nullptr;
// SSRC of unsignalled receive stream, or -1 if there isn't one.
int64_t default_recv_ssrc_ = -1;
// Volume for unsignalled stream, which may be set before the stream exists.
double default_recv_volume_ = 1.0;
- // SSRC to use for RTCP receiver reports; default to 1 in case of no signaled
+ // Default SSRC to use for RTCP receiver reports in case of no signaled
// send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
- uint32_t receiver_reports_ssrc_ = 1;
+ // and https://code.google.com/p/chromium/issues/detail?id=547661
+ uint32_t receiver_reports_ssrc_ = 0xFA17FA17u;
class WebRtcAudioSendStream;
std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
- std::vector<RtpHeaderExtension> send_extensions_;
+ std::vector<webrtc::RtpExtension> send_rtp_extensions_;
class WebRtcAudioReceiveStream;
- std::map<uint32_t, WebRtcAudioReceiveStream*> receive_channels_;
- std::map<uint32_t, webrtc::AudioReceiveStream*> receive_streams_;
- std::map<uint32_t, StreamParams> receive_stream_params_;
- // receive_channels_ can be read from WebRtc callback thread. Access from
- // the WebRtc thread must be synchronized with edits on the worker thread.
- // Reads on the worker thread are ok.
- std::vector<RtpHeaderExtension> receive_extensions_;
+ std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
};
-
} // namespace cricket
#endif // TALK_MEDIA_WEBRTCVOICEENGINE_H_
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index ce5115c..a62bcb2 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -25,6 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/gunit.h"
#include "webrtc/call.h"
@@ -53,10 +54,6 @@
const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1, 0);
const cricket::AudioCodec kTelephoneEventCodec(106, "telephone-event", 8000, 0,
1, 0);
-const cricket::AudioCodec* const kAudioCodecs[] = {
- &kPcmuCodec, &kIsacCodec, &kOpusCodec, &kG722CodecVoE, &kRedCodec,
- &kCn8000Codec, &kCn16000Codec, &kTelephoneEventCodec,
-};
const uint32_t kSsrc1 = 0x99;
const uint32_t kSsrc2 = 0x98;
const uint32_t kSsrcs4[] = { 1, 2, 3, 4 };
@@ -67,37 +64,22 @@
: cricket::VoEWrapper(engine, // processing
engine, // base
engine, // codec
- engine, // dtmf
engine, // hw
engine, // network
engine, // rtp
engine) { // volume
}
};
-
-class FakeVoETraceWrapper : public cricket::VoETraceWrapper {
- public:
- int SetTraceFilter(const unsigned int filter) override {
- filter_ = filter;
- return 0;
- }
- int SetTraceFile(const char* fileNameUTF8) override { return 0; }
- int SetTraceCallback(webrtc::TraceCallback* callback) override { return 0; }
- unsigned int filter_;
-};
} // namespace
class WebRtcVoiceEngineTestFake : public testing::Test {
public:
WebRtcVoiceEngineTestFake()
: call_(webrtc::Call::Config()),
- voe_(kAudioCodecs, ARRAY_SIZE(kAudioCodecs)),
- trace_wrapper_(new FakeVoETraceWrapper()),
- engine_(new FakeVoEWrapper(&voe_), trace_wrapper_),
+ engine_(new FakeVoEWrapper(&voe_)),
channel_(nullptr) {
send_parameters_.codecs.push_back(kPcmuCodec);
recv_parameters_.codecs.push_back(kPcmuCodec);
- options_adjust_agc_.adjust_agc_delta.Set(-10);
}
bool SetupEngine() {
if (!engine_.Init(rtc::Thread::Current())) {
@@ -123,12 +105,10 @@
void SetupForMultiSendStream() {
EXPECT_TRUE(SetupEngineWithSendStream());
// Remove stream added in Setup.
- int default_channel_num = voe_.GetLastChannel();
- EXPECT_EQ(kSsrc1, voe_.GetLocalSSRC(default_channel_num));
+ EXPECT_TRUE(call_.GetAudioSendStream(kSsrc1));
EXPECT_TRUE(channel_->RemoveSendStream(kSsrc1));
-
// Verify the channel does not exist.
- EXPECT_EQ(-1, voe_.GetChannelFromLocalSsrc(kSsrc1));
+ EXPECT_FALSE(call_.GetAudioSendStream(kSsrc1));
}
void DeliverPacket(const void* data, int len) {
rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len);
@@ -139,6 +119,24 @@
engine_.Terminate();
}
+ const cricket::FakeAudioSendStream& GetSendStream(uint32_t ssrc) {
+ const auto* send_stream = call_.GetAudioSendStream(ssrc);
+ EXPECT_TRUE(send_stream);
+ return *send_stream;
+ }
+
+ const webrtc::AudioSendStream::Config& GetSendStreamConfig(uint32_t ssrc) {
+ const auto* send_stream = call_.GetAudioSendStream(ssrc);
+ EXPECT_TRUE(send_stream);
+ return send_stream->GetConfig();
+ }
+
+ const webrtc::AudioReceiveStream::Config& GetRecvStreamConfig(uint32_t ssrc) {
+ const auto* recv_stream = call_.GetAudioReceiveStream(ssrc);
+ EXPECT_TRUE(recv_stream);
+ return recv_stream->GetConfig();
+ }
+
void TestInsertDtmf(uint32_t ssrc, bool caller) {
EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
channel_ = engine_.CreateChannel(&call_, cricket::AudioOptions());
@@ -154,39 +152,30 @@
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
EXPECT_FALSE(channel_->CanInsertDtmf());
- EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111, cricket::DF_SEND));
+ EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111));
send_parameters_.codecs.push_back(kTelephoneEventCodec);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_TRUE(channel_->CanInsertDtmf());
if (!caller) {
// If this is callee, there's no active send channel yet.
- EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123, cricket::DF_SEND));
+ EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123));
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kSsrc1)));
}
// Check we fail if the ssrc is invalid.
- EXPECT_FALSE(channel_->InsertDtmf(-1, 1, 111, cricket::DF_SEND));
+ EXPECT_FALSE(channel_->InsertDtmf(-1, 1, 111));
- // Test send
- int channel_id = voe_.GetLastChannel();
- EXPECT_FALSE(voe_.WasSendTelephoneEventCalled(channel_id, 2, 123));
- EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123, cricket::DF_SEND));
- EXPECT_TRUE(voe_.WasSendTelephoneEventCalled(channel_id, 2, 123));
-
- // Test play
- EXPECT_FALSE(voe_.WasPlayDtmfToneCalled(3, 134));
- EXPECT_TRUE(channel_->InsertDtmf(ssrc, 3, 134, cricket::DF_PLAY));
- EXPECT_TRUE(voe_.WasPlayDtmfToneCalled(3, 134));
-
- // Test send and play
- EXPECT_FALSE(voe_.WasSendTelephoneEventCalled(channel_id, 4, 145));
- EXPECT_FALSE(voe_.WasPlayDtmfToneCalled(4, 145));
- EXPECT_TRUE(channel_->InsertDtmf(ssrc, 4, 145,
- cricket::DF_PLAY | cricket::DF_SEND));
- EXPECT_TRUE(voe_.WasSendTelephoneEventCalled(channel_id, 4, 145));
- EXPECT_TRUE(voe_.WasPlayDtmfToneCalled(4, 145));
+ // Test send.
+ cricket::FakeAudioSendStream::TelephoneEvent telephone_event =
+ GetSendStream(kSsrc1).GetLatestTelephoneEvent();
+ EXPECT_EQ(-1, telephone_event.payload_type);
+ EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123));
+ telephone_event = GetSendStream(kSsrc1).GetLatestTelephoneEvent();
+ EXPECT_EQ(kTelephoneEventCodec.id, telephone_event.payload_type);
+ EXPECT_EQ(2, telephone_event.event_code);
+ EXPECT_EQ(123, telephone_event.duration_ms);
}
// Test that send bandwidth is set correctly.
@@ -211,81 +200,85 @@
void TestSetSendRtpHeaderExtensions(const std::string& ext) {
EXPECT_TRUE(SetupEngineWithSendStream());
- int channel_num = voe_.GetLastChannel();
// Ensure extensions are off by default.
- EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
+ EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());
// Ensure unknown extensions won't cause an error.
send_parameters_.extensions.push_back(cricket::RtpHeaderExtension(
"urn:ietf:params:unknownextention", 1));
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
- EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
+ EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());
// Ensure extensions stay off with an empty list of headers.
send_parameters_.extensions.clear();
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
- EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
+ EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());
// Ensure extension is set properly.
const int id = 1;
send_parameters_.extensions.push_back(cricket::RtpHeaderExtension(ext, id));
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
- EXPECT_EQ(id, voe_.GetSendRtpExtensionId(channel_num, ext));
+ EXPECT_EQ(1u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());
+ EXPECT_EQ(ext, GetSendStreamConfig(kSsrc1).rtp.extensions[0].name);
+ EXPECT_EQ(id, GetSendStreamConfig(kSsrc1).rtp.extensions[0].id);
- // Ensure extension is set properly on new channels.
+ // Ensure extension is set properly on new stream.
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(kSsrc2)));
- int new_channel_num = voe_.GetLastChannel();
- EXPECT_NE(channel_num, new_channel_num);
- EXPECT_EQ(id, voe_.GetSendRtpExtensionId(new_channel_num, ext));
+ EXPECT_NE(call_.GetAudioSendStream(kSsrc1),
+ call_.GetAudioSendStream(kSsrc2));
+ EXPECT_EQ(1u, GetSendStreamConfig(kSsrc2).rtp.extensions.size());
+ EXPECT_EQ(ext, GetSendStreamConfig(kSsrc2).rtp.extensions[0].name);
+ EXPECT_EQ(id, GetSendStreamConfig(kSsrc2).rtp.extensions[0].id);
// Ensure all extensions go back off with an empty list.
send_parameters_.codecs.push_back(kPcmuCodec);
send_parameters_.extensions.clear();
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
- EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(channel_num, ext));
- EXPECT_EQ(-1, voe_.GetSendRtpExtensionId(new_channel_num, ext));
+ EXPECT_EQ(0u, GetSendStreamConfig(kSsrc1).rtp.extensions.size());
+ EXPECT_EQ(0u, GetSendStreamConfig(kSsrc2).rtp.extensions.size());
}
void TestSetRecvRtpHeaderExtensions(const std::string& ext) {
EXPECT_TRUE(SetupEngineWithRecvStream());
- int channel_num = voe_.GetLastChannel();
// Ensure extensions are off by default.
- EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
+ EXPECT_EQ(0u, GetRecvStreamConfig(kSsrc1).rtp.extensions.size());
- cricket::AudioRecvParameters parameters;
// Ensure unknown extensions won't cause an error.
- parameters.extensions.push_back(cricket::RtpHeaderExtension(
+ recv_parameters_.extensions.push_back(cricket::RtpHeaderExtension(
"urn:ietf:params:unknownextention", 1));
- EXPECT_TRUE(channel_->SetRecvParameters(parameters));
- EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
+ EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
+ EXPECT_EQ(0u, GetRecvStreamConfig(kSsrc1).rtp.extensions.size());
// Ensure extensions stay off with an empty list of headers.
- parameters.extensions.clear();
- EXPECT_TRUE(channel_->SetRecvParameters(parameters));
- EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
+ recv_parameters_.extensions.clear();
+ EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
+ EXPECT_EQ(0u, GetRecvStreamConfig(kSsrc1).rtp.extensions.size());
// Ensure extension is set properly.
const int id = 2;
- parameters.extensions.push_back(cricket::RtpHeaderExtension(ext, id));
- EXPECT_TRUE(channel_->SetRecvParameters(parameters));
- EXPECT_EQ(id, voe_.GetReceiveRtpExtensionId(channel_num, ext));
+ recv_parameters_.extensions.push_back(cricket::RtpHeaderExtension(ext, id));
+ EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
+ EXPECT_EQ(1u, GetRecvStreamConfig(kSsrc1).rtp.extensions.size());
+ EXPECT_EQ(ext, GetRecvStreamConfig(kSsrc1).rtp.extensions[0].name);
+ EXPECT_EQ(id, GetRecvStreamConfig(kSsrc1).rtp.extensions[0].id);
- // Ensure extension is set properly on new channel.
- // The first stream to occupy the default channel.
+ // Ensure extension is set properly on new stream.
EXPECT_TRUE(channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kSsrc2)));
- int new_channel_num = voe_.GetLastChannel();
- EXPECT_NE(channel_num, new_channel_num);
- EXPECT_EQ(id, voe_.GetReceiveRtpExtensionId(new_channel_num, ext));
+ EXPECT_NE(call_.GetAudioReceiveStream(kSsrc1),
+ call_.GetAudioReceiveStream(kSsrc2));
+ EXPECT_EQ(1u, GetRecvStreamConfig(kSsrc2).rtp.extensions.size());
+ EXPECT_EQ(ext, GetRecvStreamConfig(kSsrc2).rtp.extensions[0].name);
+ EXPECT_EQ(id, GetRecvStreamConfig(kSsrc2).rtp.extensions[0].id);
// Ensure all extensions go back off with an empty list.
- parameters.extensions.clear();
- EXPECT_TRUE(channel_->SetRecvParameters(parameters));
- EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(channel_num, ext));
- EXPECT_EQ(-1, voe_.GetReceiveRtpExtensionId(new_channel_num, ext));
+ recv_parameters_.extensions.clear();
+ EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
+ EXPECT_EQ(0u, GetRecvStreamConfig(kSsrc1).rtp.extensions.size());
+ EXPECT_EQ(0u, GetRecvStreamConfig(kSsrc2).rtp.extensions.size());
}
webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const {
@@ -313,7 +306,8 @@
s->SetStats(GetAudioSendStreamStats());
}
}
- void VerifyVoiceSenderInfo(const cricket::VoiceSenderInfo& info) {
+ void VerifyVoiceSenderInfo(const cricket::VoiceSenderInfo& info,
+ bool is_sending) {
const auto stats = GetAudioSendStreamStats();
EXPECT_EQ(info.ssrc(), stats.local_ssrc);
EXPECT_EQ(info.bytes_sent, stats.bytes_sent);
@@ -331,8 +325,8 @@
EXPECT_EQ(info.echo_return_loss, stats.echo_return_loss);
EXPECT_EQ(info.echo_return_loss_enhancement,
stats.echo_return_loss_enhancement);
- // TODO(solenberg): Move typing noise detection into AudioSendStream.
- // EXPECT_EQ(info.typing_noise_detected, stats.typing_noise_detected);
+ EXPECT_EQ(info.typing_noise_detected,
+ stats.typing_noise_detected && is_sending);
}
webrtc::AudioReceiveStream::Stats GetAudioReceiveStreamStats() const {
@@ -401,13 +395,10 @@
protected:
cricket::FakeCall call_;
cricket::FakeWebRtcVoiceEngine voe_;
- FakeVoETraceWrapper* trace_wrapper_;
cricket::WebRtcVoiceEngine engine_;
cricket::VoiceMediaChannel* channel_;
-
cricket::AudioSendParameters send_parameters_;
cricket::AudioRecvParameters recv_parameters_;
- cricket::AudioOptions options_adjust_agc_;
};
// Tests that our stub library "works".
@@ -448,32 +439,33 @@
cricket::AudioCodec codec;
webrtc::CodecInst codec_inst;
// Find PCMU with explicit clockrate and bitrate.
- EXPECT_TRUE(engine_.FindWebRtcCodec(kPcmuCodec, &codec_inst));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kPcmuCodec, &codec_inst));
// Find ISAC with explicit clockrate and 0 bitrate.
- EXPECT_TRUE(engine_.FindWebRtcCodec(kIsacCodec, &codec_inst));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kIsacCodec, &codec_inst));
// Find telephone-event with explicit clockrate and 0 bitrate.
- EXPECT_TRUE(engine_.FindWebRtcCodec(kTelephoneEventCodec, &codec_inst));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kTelephoneEventCodec,
+ &codec_inst));
// Find ISAC with a different payload id.
codec = kIsacCodec;
codec.id = 127;
- EXPECT_TRUE(engine_.FindWebRtcCodec(codec, &codec_inst));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst));
EXPECT_EQ(codec.id, codec_inst.pltype);
// Find PCMU with a 0 clockrate.
codec = kPcmuCodec;
codec.clockrate = 0;
- EXPECT_TRUE(engine_.FindWebRtcCodec(codec, &codec_inst));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst));
EXPECT_EQ(codec.id, codec_inst.pltype);
EXPECT_EQ(8000, codec_inst.plfreq);
// Find PCMU with a 0 bitrate.
codec = kPcmuCodec;
codec.bitrate = 0;
- EXPECT_TRUE(engine_.FindWebRtcCodec(codec, &codec_inst));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst));
EXPECT_EQ(codec.id, codec_inst.pltype);
EXPECT_EQ(64000, codec_inst.rate);
// Find ISAC with an explicit bitrate.
codec = kIsacCodec;
codec.bitrate = 32000;
- EXPECT_TRUE(engine_.FindWebRtcCodec(codec, &codec_inst));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(codec, &codec_inst));
EXPECT_EQ(codec.id, codec_inst.pltype);
EXPECT_EQ(32000, codec_inst.rate);
}
@@ -492,14 +484,13 @@
cricket::StreamParams::CreateLegacy(kSsrc1)));
int channel_num = voe_.GetLastChannel();
webrtc::CodecInst gcodec;
- rtc::strcpyn(gcodec.plname, ARRAY_SIZE(gcodec.plname), "ISAC");
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
gcodec.plfreq = 16000;
gcodec.channels = 1;
EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
EXPECT_EQ(106, gcodec.pltype);
EXPECT_STREQ("ISAC", gcodec.plname);
- rtc::strcpyn(gcodec.plname, ARRAY_SIZE(gcodec.plname),
- "telephone-event");
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
gcodec.plfreq = 8000;
EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num, gcodec));
EXPECT_EQ(126, gcodec.pltype);
@@ -537,7 +528,7 @@
cricket::StreamParams::CreateLegacy(kSsrc1)));
int channel_num = voe_.GetLastChannel();
webrtc::CodecInst opus;
- engine_.FindWebRtcCodec(kOpusCodec, &opus);
+ cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
// Even without stereo parameters, recv codecs still specify channels = 2.
EXPECT_EQ(2, opus.channels);
EXPECT_EQ(111, opus.pltype);
@@ -560,7 +551,7 @@
cricket::StreamParams::CreateLegacy(kSsrc1)));
int channel_num2 = voe_.GetLastChannel();
webrtc::CodecInst opus;
- engine_.FindWebRtcCodec(kOpusCodec, &opus);
+ cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
// Even when stereo is off, recv codecs still specify channels = 2.
EXPECT_EQ(2, opus.channels);
EXPECT_EQ(111, opus.pltype);
@@ -583,7 +574,7 @@
cricket::StreamParams::CreateLegacy(kSsrc1)));
int channel_num2 = voe_.GetLastChannel();
webrtc::CodecInst opus;
- engine_.FindWebRtcCodec(kOpusCodec, &opus);
+ cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &opus);
EXPECT_EQ(2, opus.channels);
EXPECT_EQ(111, opus.pltype);
EXPECT_STREQ("opus", opus.plname);
@@ -606,14 +597,13 @@
cricket::StreamParams::CreateLegacy(kSsrc1)));
int channel_num2 = voe_.GetLastChannel();
webrtc::CodecInst gcodec;
- rtc::strcpyn(gcodec.plname, ARRAY_SIZE(gcodec.plname), "ISAC");
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
gcodec.plfreq = 16000;
gcodec.channels = 1;
EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
EXPECT_EQ(106, gcodec.pltype);
EXPECT_STREQ("ISAC", gcodec.plname);
- rtc::strcpyn(gcodec.plname, ARRAY_SIZE(gcodec.plname),
- "telephone-event");
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "telephone-event");
gcodec.plfreq = 8000;
gcodec.channels = 1;
EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
@@ -630,7 +620,7 @@
int channel_num2 = voe_.GetLastChannel();
webrtc::CodecInst gcodec;
- rtc::strcpyn(gcodec.plname, ARRAY_SIZE(gcodec.plname), "ISAC");
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "ISAC");
gcodec.plfreq = 16000;
gcodec.channels = 1;
EXPECT_EQ(0, voe_.GetRecPayloadType(channel_num2, gcodec));
@@ -669,7 +659,7 @@
int channel_num = voe_.GetLastChannel();
EXPECT_TRUE(voe_.GetPlayout(channel_num));
webrtc::CodecInst gcodec;
- EXPECT_TRUE(engine_.FindWebRtcCodec(kOpusCodec, &gcodec));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(kOpusCodec, &gcodec));
EXPECT_EQ(kOpusCodec.id, gcodec.pltype);
}
@@ -782,7 +772,7 @@
EXPECT_FALSE(voe_.GetRED(channel_num));
EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
EXPECT_EQ(105, voe_.GetSendCNPayloadType(channel_num, true));
- EXPECT_EQ(106, voe_.GetSendTelephoneEventPayloadType(channel_num));
+ EXPECT_FALSE(channel_->CanInsertDtmf());
}
// Test that VoE Channel doesn't call SetSendCodec again if same codec is tried
@@ -1623,7 +1613,7 @@
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_EQ(96, gcodec.pltype);
EXPECT_STREQ("ISAC", gcodec.plname);
- EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
+ EXPECT_TRUE(channel_->CanInsertDtmf());
}
// Test that we can set send codecs even with CN codec as the first
@@ -1669,7 +1659,7 @@
EXPECT_FALSE(voe_.GetRED(channel_num));
EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
- EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
+ EXPECT_TRUE(channel_->CanInsertDtmf());
}
// Test that we set VAD and DTMF types correctly as callee.
@@ -1702,7 +1692,7 @@
EXPECT_FALSE(voe_.GetRED(channel_num));
EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
- EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
+ EXPECT_TRUE(channel_->CanInsertDtmf());
}
// Test that we only apply VAD if we have a CN codec that matches the
@@ -1766,7 +1756,7 @@
EXPECT_FALSE(voe_.GetRED(channel_num));
EXPECT_EQ(13, voe_.GetSendCNPayloadType(channel_num, false));
EXPECT_EQ(97, voe_.GetSendCNPayloadType(channel_num, true));
- EXPECT_EQ(98, voe_.GetSendTelephoneEventPayloadType(channel_num));
+ EXPECT_TRUE(channel_->CanInsertDtmf());
}
// Test that we set up RED correctly as caller.
@@ -1976,21 +1966,16 @@
for (uint32_t ssrc : kSsrcs4) {
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(ssrc)));
- EXPECT_NE(nullptr, call_.GetAudioSendStream(ssrc));
-
// Verify that we are in a sending state for all the created streams.
- int channel_num = voe_.GetChannelFromLocalSsrc(ssrc);
- EXPECT_TRUE(voe_.GetSend(channel_num));
+ EXPECT_TRUE(voe_.GetSend(GetSendStreamConfig(ssrc).voe_channel_id));
}
- EXPECT_EQ(ARRAY_SIZE(kSsrcs4), call_.GetAudioSendStreams().size());
+ EXPECT_EQ(arraysize(kSsrcs4), call_.GetAudioSendStreams().size());
// Delete the send streams.
for (uint32_t ssrc : kSsrcs4) {
EXPECT_TRUE(channel_->RemoveSendStream(ssrc));
- EXPECT_EQ(nullptr, call_.GetAudioSendStream(ssrc));
- // Stream should already be deleted.
+ EXPECT_FALSE(call_.GetAudioSendStream(ssrc));
EXPECT_FALSE(channel_->RemoveSendStream(ssrc));
- EXPECT_EQ(-1, voe_.GetChannelFromLocalSsrc(ssrc));
}
EXPECT_EQ(0u, call_.GetAudioSendStreams().size());
}
@@ -2015,7 +2000,7 @@
// Verify ISAC and VAD are corrected configured on all send channels.
webrtc::CodecInst gcodec;
for (uint32_t ssrc : kSsrcs4) {
- int channel_num = voe_.GetChannelFromLocalSsrc(ssrc);
+ int channel_num = GetSendStreamConfig(ssrc).voe_channel_id;
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_STREQ("ISAC", gcodec.plname);
EXPECT_TRUE(voe_.GetVAD(channel_num));
@@ -2026,7 +2011,7 @@
parameters.codecs[0] = kPcmuCodec;
EXPECT_TRUE(channel_->SetSendParameters(parameters));
for (uint32_t ssrc : kSsrcs4) {
- int channel_num = voe_.GetChannelFromLocalSsrc(ssrc);
+ int channel_num = GetSendStreamConfig(ssrc).voe_channel_id;
EXPECT_EQ(0, voe_.GetSendCodec(channel_num, gcodec));
EXPECT_STREQ("PCMU", gcodec.plname);
EXPECT_FALSE(voe_.GetVAD(channel_num));
@@ -2049,7 +2034,7 @@
EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
for (uint32_t ssrc : kSsrcs4) {
// Verify that we are in a sending state for all the send streams.
- int channel_num = voe_.GetChannelFromLocalSsrc(ssrc);
+ int channel_num = GetSendStreamConfig(ssrc).voe_channel_id;
EXPECT_TRUE(voe_.GetSend(channel_num));
}
@@ -2057,7 +2042,7 @@
EXPECT_TRUE(channel_->SetSend(cricket::SEND_NOTHING));
for (uint32_t ssrc : kSsrcs4) {
// Verify that we are in a stop state for all the send streams.
- int channel_num = voe_.GetChannelFromLocalSsrc(ssrc);
+ int channel_num = GetSendStreamConfig(ssrc).voe_channel_id;
EXPECT_FALSE(voe_.GetSend(channel_num));
}
}
@@ -2087,9 +2072,9 @@
EXPECT_EQ(true, channel_->GetStats(&info));
// We have added 4 send streams. We should see empty stats for all.
- EXPECT_EQ(static_cast<size_t>(ARRAY_SIZE(kSsrcs4)), info.senders.size());
+ EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
for (const auto& sender : info.senders) {
- VerifyVoiceSenderInfo(sender);
+ VerifyVoiceSenderInfo(sender, false);
}
// We have added one receive stream. We should see empty stats.
@@ -2102,7 +2087,7 @@
cricket::VoiceMediaInfo info;
EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc2));
EXPECT_EQ(true, channel_->GetStats(&info));
- EXPECT_EQ(static_cast<size_t>(ARRAY_SIZE(kSsrcs4)), info.senders.size());
+ EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
EXPECT_EQ(0u, info.receivers.size());
}
@@ -2113,7 +2098,7 @@
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
SetAudioReceiveStreamStats();
EXPECT_EQ(true, channel_->GetStats(&info));
- EXPECT_EQ(static_cast<size_t>(ARRAY_SIZE(kSsrcs4)), info.senders.size());
+ EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
EXPECT_EQ(1u, info.receivers.size());
VerifyVoiceReceiverInfo(info.receivers[0]);
}
@@ -2173,96 +2158,17 @@
EXPECT_FALSE(voe_.GetPlayout(channel_num1));
}
-// Test that we can set the devices to use.
-TEST_F(WebRtcVoiceEngineTestFake, SetDevices) {
- EXPECT_TRUE(SetupEngineWithSendStream());
- int send_channel = voe_.GetLastChannel();
- EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
- int recv_channel = voe_.GetLastChannel();
- EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-
- cricket::Device default_dev(cricket::kFakeDefaultDeviceName,
- cricket::kFakeDefaultDeviceId);
- cricket::Device dev(cricket::kFakeDeviceName,
- cricket::kFakeDeviceId);
-
- // Test SetDevices() while not sending or playing.
- EXPECT_TRUE(engine_.SetDevices(&default_dev, &default_dev));
-
- // Test SetDevices() while sending and playing.
- EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
- EXPECT_TRUE(channel_->SetPlayout(true));
- EXPECT_TRUE(voe_.GetSend(send_channel));
- EXPECT_TRUE(voe_.GetPlayout(recv_channel));
-
- EXPECT_TRUE(engine_.SetDevices(&dev, &dev));
-
- EXPECT_TRUE(voe_.GetSend(send_channel));
- EXPECT_TRUE(voe_.GetPlayout(recv_channel));
-
- // Test that failure to open newly selected devices does not prevent opening
- // ones after that.
- voe_.set_playout_fail_channel(recv_channel);
- voe_.set_send_fail_channel(send_channel);
-
- EXPECT_FALSE(engine_.SetDevices(&default_dev, &default_dev));
-
- EXPECT_FALSE(voe_.GetSend(send_channel));
- EXPECT_FALSE(voe_.GetPlayout(recv_channel));
-
- voe_.set_playout_fail_channel(-1);
- voe_.set_send_fail_channel(-1);
-
- EXPECT_TRUE(engine_.SetDevices(&dev, &dev));
-
- EXPECT_TRUE(voe_.GetSend(send_channel));
- EXPECT_TRUE(voe_.GetPlayout(recv_channel));
-}
-
-// Test that we can set the devices to use even if we failed to
-// open the initial ones.
-TEST_F(WebRtcVoiceEngineTestFake, SetDevicesWithInitiallyBadDevices) {
- EXPECT_TRUE(SetupEngineWithSendStream());
- int send_channel = voe_.GetLastChannel();
- EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
- int recv_channel = voe_.GetLastChannel();
- EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
-
- cricket::Device default_dev(cricket::kFakeDefaultDeviceName,
- cricket::kFakeDefaultDeviceId);
- cricket::Device dev(cricket::kFakeDeviceName,
- cricket::kFakeDeviceId);
-
- // Test that failure to open devices selected before starting
- // send/play does not prevent opening newly selected ones after that.
- voe_.set_playout_fail_channel(recv_channel);
- voe_.set_send_fail_channel(send_channel);
-
- EXPECT_TRUE(engine_.SetDevices(&default_dev, &default_dev));
-
- EXPECT_FALSE(channel_->SetSend(cricket::SEND_MICROPHONE));
- EXPECT_FALSE(channel_->SetPlayout(true));
- EXPECT_FALSE(voe_.GetSend(send_channel));
- EXPECT_FALSE(voe_.GetPlayout(recv_channel));
-
- voe_.set_playout_fail_channel(-1);
- voe_.set_send_fail_channel(-1);
-
- EXPECT_TRUE(engine_.SetDevices(&dev, &dev));
-
- EXPECT_TRUE(voe_.GetSend(send_channel));
- EXPECT_TRUE(voe_.GetPlayout(recv_channel));
-}
-
// Test that we can create a channel configured for Codian bridges,
// and start sending on it.
TEST_F(WebRtcVoiceEngineTestFake, CodianSend) {
EXPECT_TRUE(SetupEngineWithSendStream());
+ cricket::AudioOptions options_adjust_agc;
+ options_adjust_agc.adjust_agc_delta = rtc::Optional<int>(-10);
int channel_num = voe_.GetLastChannel();
webrtc::AgcConfig agc_config;
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
EXPECT_EQ(0, agc_config.targetLeveldBOv);
- send_parameters_.options = options_adjust_agc_;
+ send_parameters_.options = options_adjust_agc;
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
EXPECT_TRUE(voe_.GetSend(channel_num));
@@ -2271,7 +2177,6 @@
EXPECT_TRUE(channel_->SetSend(cricket::SEND_NOTHING));
EXPECT_FALSE(voe_.GetSend(channel_num));
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
- EXPECT_EQ(0, agc_config.targetLeveldBOv); // level was restored
}
TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
@@ -2279,14 +2184,12 @@
webrtc::AgcConfig agc_config;
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
EXPECT_EQ(0, agc_config.targetLeveldBOv);
-
- cricket::AudioOptions options;
- options.tx_agc_target_dbov.Set(3);
- options.tx_agc_digital_compression_gain.Set(9);
- options.tx_agc_limiter.Set(true);
- options.auto_gain_control.Set(true);
- EXPECT_TRUE(engine_.SetOptions(options));
-
+ send_parameters_.options.tx_agc_target_dbov = rtc::Optional<uint16_t>(3);
+ send_parameters_.options.tx_agc_digital_compression_gain =
+ rtc::Optional<uint16_t>(9);
+ send_parameters_.options.tx_agc_limiter = rtc::Optional<bool>(true);
+ send_parameters_.options.auto_gain_control = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
EXPECT_EQ(3, agc_config.targetLeveldBOv);
EXPECT_EQ(9, agc_config.digitalCompressionGaindB);
@@ -2294,19 +2197,18 @@
// Check interaction with adjust_agc_delta. Both should be respected, for
// backwards compatibility.
- options.adjust_agc_delta.Set(-10);
- EXPECT_TRUE(engine_.SetOptions(options));
-
+ send_parameters_.options.adjust_agc_delta = rtc::Optional<int>(-10);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_EQ(0, voe_.GetAgcConfig(agc_config));
EXPECT_EQ(13, agc_config.targetLeveldBOv);
}
TEST_F(WebRtcVoiceEngineTestFake, SampleRatesViaOptions) {
EXPECT_TRUE(SetupEngineWithSendStream());
- cricket::AudioOptions options;
- options.recording_sample_rate.Set(48000u);
- options.playout_sample_rate.Set(44100u);
- EXPECT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.recording_sample_rate =
+ rtc::Optional<uint32_t>(48000);
+ send_parameters_.options.playout_sample_rate = rtc::Optional<uint32_t>(44100);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
unsigned int recording_sample_rate, playout_sample_rate;
EXPECT_EQ(0, voe_.RecordingSampleRate(&recording_sample_rate));
@@ -2315,30 +2217,11 @@
EXPECT_EQ(44100u, playout_sample_rate);
}
-TEST_F(WebRtcVoiceEngineTestFake, TraceFilterViaTraceOptions) {
- EXPECT_TRUE(SetupEngineWithSendStream());
- engine_.SetLogging(rtc::LS_INFO, "");
- EXPECT_EQ(
- // Info:
- webrtc::kTraceStateInfo | webrtc::kTraceInfo |
- // Warning:
- webrtc::kTraceTerseInfo | webrtc::kTraceWarning |
- // Error:
- webrtc::kTraceError | webrtc::kTraceCritical,
- static_cast<int>(trace_wrapper_->filter_));
- // Now set it explicitly
- std::string filter =
- "tracefilter " + rtc::ToString(webrtc::kTraceDefault);
- engine_.SetLogging(rtc::LS_VERBOSE, filter.c_str());
- EXPECT_EQ(static_cast<unsigned int>(webrtc::kTraceDefault),
- trace_wrapper_->filter_);
-}
-
// Test that we can set the outgoing SSRC properly.
// SSRC is set in SetupEngine by calling AddSendStream.
TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrc) {
EXPECT_TRUE(SetupEngineWithSendStream());
- EXPECT_EQ(kSsrc1, voe_.GetLocalSSRC(voe_.GetLastChannel()));
+ EXPECT_TRUE(call_.GetAudioSendStream(kSsrc1));
}
TEST_F(WebRtcVoiceEngineTestFake, GetStats) {
@@ -2359,12 +2242,20 @@
// We have added one send stream. We should see the stats we've set.
EXPECT_EQ(1u, info.senders.size());
- VerifyVoiceSenderInfo(info.senders[0]);
+ VerifyVoiceSenderInfo(info.senders[0], false);
// We have added one receive stream. We should see empty stats.
EXPECT_EQ(info.receivers.size(), 1u);
EXPECT_EQ(info.receivers[0].ssrc(), 0);
}
+ // Start sending - this affects some reported stats.
+ {
+ cricket::VoiceMediaInfo info;
+ EXPECT_TRUE(channel_->SetSend(cricket::SEND_MICROPHONE));
+ EXPECT_EQ(true, channel_->GetStats(&info));
+ VerifyVoiceSenderInfo(info.senders[0], true);
+ }
+
// Remove the kSsrc2 stream. No receiver stats.
{
cricket::VoiceMediaInfo info;
@@ -2391,9 +2282,10 @@
// SSRC is set in SetupEngine by calling AddSendStream.
TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcWithMultipleStreams) {
EXPECT_TRUE(SetupEngineWithSendStream());
- EXPECT_EQ(kSsrc1, voe_.GetLocalSSRC(voe_.GetLastChannel()));
- EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
- EXPECT_EQ(kSsrc1, voe_.GetLocalSSRC(voe_.GetLastChannel()));
+ EXPECT_TRUE(call_.GetAudioSendStream(kSsrc1));
+ EXPECT_TRUE(channel_->AddRecvStream(
+ cricket::StreamParams::CreateLegacy(kSsrc2)));
+ EXPECT_EQ(kSsrc1, GetRecvStreamConfig(kSsrc2).rtp.local_ssrc);
}
// Test that the local SSRC is the same on sending and receiving channels if the
@@ -2406,25 +2298,23 @@
int receive_channel_num = voe_.GetLastChannel();
EXPECT_TRUE(channel_->AddSendStream(
cricket::StreamParams::CreateLegacy(1234)));
- int send_channel_num = voe_.GetLastChannel();
- EXPECT_EQ(1234U, voe_.GetLocalSSRC(send_channel_num));
+ EXPECT_TRUE(call_.GetAudioSendStream(1234));
EXPECT_EQ(1234U, voe_.GetLocalSSRC(receive_channel_num));
}
// Test that we can properly receive packets.
TEST_F(WebRtcVoiceEngineTestFake, Recv) {
EXPECT_TRUE(SetupEngine());
+ EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
int channel_num = voe_.GetLastChannel();
- EXPECT_TRUE(voe_.CheckPacket(channel_num, kPcmuFrame,
- sizeof(kPcmuFrame)));
+ EXPECT_TRUE(voe_.CheckPacket(channel_num, kPcmuFrame, sizeof(kPcmuFrame)));
}
// Test that we can properly receive packets on multiple streams.
TEST_F(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) {
- EXPECT_TRUE(SetupEngineWithSendStream());
- EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
+ EXPECT_TRUE(SetupEngine());
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
int channel_num1 = voe_.GetLastChannel();
EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
@@ -2433,37 +2323,97 @@
int channel_num3 = voe_.GetLastChannel();
// Create packets with the right SSRCs.
char packets[4][sizeof(kPcmuFrame)];
- for (size_t i = 0; i < ARRAY_SIZE(packets); ++i) {
+ for (size_t i = 0; i < arraysize(packets); ++i) {
memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame));
rtc::SetBE32(packets[i] + 8, static_cast<uint32_t>(i));
}
EXPECT_TRUE(voe_.CheckNoPacket(channel_num1));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num2));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num3));
+
DeliverPacket(packets[0], sizeof(packets[0]));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num1));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num2));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num3));
+
DeliverPacket(packets[1], sizeof(packets[1]));
- EXPECT_TRUE(voe_.CheckPacket(channel_num1, packets[1],
- sizeof(packets[1])));
+ EXPECT_TRUE(voe_.CheckPacket(channel_num1, packets[1], sizeof(packets[1])));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num2));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num3));
+
DeliverPacket(packets[2], sizeof(packets[2]));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num1));
- EXPECT_TRUE(voe_.CheckPacket(channel_num2, packets[2],
- sizeof(packets[2])));
+ EXPECT_TRUE(voe_.CheckPacket(channel_num2, packets[2], sizeof(packets[2])));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num3));
+
DeliverPacket(packets[3], sizeof(packets[3]));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num1));
EXPECT_TRUE(voe_.CheckNoPacket(channel_num2));
- EXPECT_TRUE(voe_.CheckPacket(channel_num3, packets[3],
- sizeof(packets[3])));
+ EXPECT_TRUE(voe_.CheckPacket(channel_num3, packets[3], sizeof(packets[3])));
+
EXPECT_TRUE(channel_->RemoveRecvStream(3));
EXPECT_TRUE(channel_->RemoveRecvStream(2));
EXPECT_TRUE(channel_->RemoveRecvStream(1));
}
+// Test that receiving on an unsignalled stream works (default channel will be
+// created).
+TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignalled) {
+ EXPECT_TRUE(SetupEngine());
+ DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
+ int channel_num = voe_.GetLastChannel();
+ EXPECT_TRUE(voe_.CheckPacket(channel_num, kPcmuFrame, sizeof(kPcmuFrame)));
+}
+
+// Test that receiving on an unsignalled stream works (default channel will be
+// created), and that packets will be forwarded to the default channel
+// regardless of their SSRCs.
+TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignalledWithSsrcSwitch) {
+ EXPECT_TRUE(SetupEngine());
+ char packet[sizeof(kPcmuFrame)];
+ memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
+
+ // Note that the first unknown SSRC cannot be 0, because we only support
+ // creating receive streams for SSRC!=0.
+ DeliverPacket(packet, sizeof(packet));
+ int channel_num = voe_.GetLastChannel();
+ EXPECT_TRUE(voe_.CheckPacket(channel_num, packet, sizeof(packet)));
+ // Once we have the default channel, SSRC==0 will be ok.
+ for (uint32_t ssrc = 0; ssrc < 10; ++ssrc) {
+ rtc::SetBE32(&packet[8], ssrc);
+ DeliverPacket(packet, sizeof(packet));
+ EXPECT_TRUE(voe_.CheckPacket(channel_num, packet, sizeof(packet)));
+ }
+}
+
+// Test that a default channel is created even after a signalled stream has been
+// added, and that this stream will get any packets for unknown SSRCs.
+TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignalledAfterSignalled) {
+ EXPECT_TRUE(SetupEngine());
+ char packet[sizeof(kPcmuFrame)];
+ memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
+
+ // Add a known stream, send packet and verify we got it.
+ EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
+ int signalled_channel_num = voe_.GetLastChannel();
+ DeliverPacket(packet, sizeof(packet));
+ EXPECT_TRUE(voe_.CheckPacket(signalled_channel_num, packet, sizeof(packet)));
+
+ // Note that the first unknown SSRC cannot be 0, because we only support
+ // creating receive streams for SSRC!=0.
+ rtc::SetBE32(&packet[8], 7011);
+ DeliverPacket(packet, sizeof(packet));
+ int channel_num = voe_.GetLastChannel();
+ EXPECT_NE(channel_num, signalled_channel_num);
+ EXPECT_TRUE(voe_.CheckPacket(channel_num, packet, sizeof(packet)));
+ // Once we have the default channel, SSRC==0 will be ok.
+ for (uint32_t ssrc = 0; ssrc < 20; ssrc += 2) {
+ rtc::SetBE32(&packet[8], ssrc);
+ DeliverPacket(packet, sizeof(packet));
+ EXPECT_TRUE(voe_.CheckPacket(channel_num, packet, sizeof(packet)));
+ }
+}
+
// Test that we properly handle failures to add a receive stream.
TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamFail) {
EXPECT_TRUE(SetupEngine());
@@ -2498,7 +2448,7 @@
cricket::StreamParams::CreateLegacy(kSsrc1)));
int channel_num2 = voe_.GetLastChannel();
webrtc::CodecInst gcodec;
- rtc::strcpyn(gcodec.plname, ARRAY_SIZE(gcodec.plname), "opus");
+ rtc::strcpyn(gcodec.plname, arraysize(gcodec.plname), "opus");
gcodec.plfreq = 48000;
gcodec.channels = 2;
EXPECT_EQ(-1, voe_.GetRecPayloadType(channel_num2, gcodec));
@@ -2602,10 +2552,12 @@
EXPECT_TRUE(typing_detection_enabled);
EXPECT_EQ(ec_mode, webrtc::kEcConference);
EXPECT_EQ(ns_mode, webrtc::kNsHighSuppression);
+ EXPECT_EQ(50, voe_.GetNetEqCapacity());
+ EXPECT_FALSE(voe_.GetNetEqFastAccelerate());
- // Nothing set, so all ignored.
- cricket::AudioOptions options;
- ASSERT_TRUE(engine_.SetOptions(options));
+ // Nothing set in AudioOptions, so everything should be as default.
+ send_parameters_.options = cricket::AudioOptions();
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAecmMode(aecm_mode, cng_enabled);
voe_.GetAgcStatus(agc_enabled, agc_mode);
@@ -2625,20 +2577,19 @@
EXPECT_TRUE(typing_detection_enabled);
EXPECT_EQ(ec_mode, webrtc::kEcConference);
EXPECT_EQ(ns_mode, webrtc::kNsHighSuppression);
- EXPECT_EQ(50, voe_.GetNetEqCapacity()); // From GetDefaultEngineOptions().
- EXPECT_FALSE(
- voe_.GetNetEqFastAccelerate()); // From GetDefaultEngineOptions().
+ EXPECT_EQ(50, voe_.GetNetEqCapacity());
+ EXPECT_FALSE(voe_.GetNetEqFastAccelerate());
// Turn echo cancellation off
- options.echo_cancellation.Set(false);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.echo_cancellation = rtc::Optional<bool>(false);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
EXPECT_FALSE(ec_enabled);
// Turn echo cancellation back on, with settings, and make sure
// nothing else changed.
- options.echo_cancellation.Set(true);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.echo_cancellation = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAecmMode(aecm_mode, cng_enabled);
voe_.GetAgcStatus(agc_enabled, agc_mode);
@@ -2660,8 +2611,8 @@
// Turn on delay agnostic aec and make sure nothing change w.r.t. echo
// control.
- options.delay_agnostic_aec.Set(true);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.delay_agnostic_aec = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAecmMode(aecm_mode, cng_enabled);
EXPECT_TRUE(ec_enabled);
@@ -2669,41 +2620,41 @@
EXPECT_EQ(ec_mode, webrtc::kEcConference);
// Turn off echo cancellation and delay agnostic aec.
- options.delay_agnostic_aec.Set(false);
- options.extended_filter_aec.Set(false);
- options.echo_cancellation.Set(false);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.delay_agnostic_aec = rtc::Optional<bool>(false);
+ send_parameters_.options.extended_filter_aec = rtc::Optional<bool>(false);
+ send_parameters_.options.echo_cancellation = rtc::Optional<bool>(false);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
EXPECT_FALSE(ec_enabled);
// Turning delay agnostic aec back on should also turn on echo cancellation.
- options.delay_agnostic_aec.Set(true);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.delay_agnostic_aec = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
EXPECT_TRUE(ec_enabled);
EXPECT_TRUE(voe_.ec_metrics_enabled());
EXPECT_EQ(ec_mode, webrtc::kEcConference);
// Turn off AGC
- options.auto_gain_control.Set(false);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.auto_gain_control = rtc::Optional<bool>(false);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetAgcStatus(agc_enabled, agc_mode);
EXPECT_FALSE(agc_enabled);
// Turn AGC back on
- options.auto_gain_control.Set(true);
- options.adjust_agc_delta.Clear();
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.auto_gain_control = rtc::Optional<bool>(true);
+ send_parameters_.options.adjust_agc_delta = rtc::Optional<int>();
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetAgcStatus(agc_enabled, agc_mode);
EXPECT_TRUE(agc_enabled);
voe_.GetAgcConfig(agc_config);
EXPECT_EQ(0, agc_config.targetLeveldBOv);
// Turn off other options (and stereo swapping on).
- options.noise_suppression.Set(false);
- options.highpass_filter.Set(false);
- options.typing_detection.Set(false);
- options.stereo_swapping.Set(true);
- ASSERT_TRUE(engine_.SetOptions(options));
+ send_parameters_.options.noise_suppression = rtc::Optional<bool>(false);
+ send_parameters_.options.highpass_filter = rtc::Optional<bool>(false);
+ send_parameters_.options.typing_detection = rtc::Optional<bool>(false);
+ send_parameters_.options.stereo_swapping = rtc::Optional<bool>(true);
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetNsStatus(ns_enabled, ns_mode);
highpass_filter_enabled = voe_.IsHighPassFilterEnabled();
stereo_swapping_enabled = voe_.IsStereoChannelSwappingEnabled();
@@ -2714,7 +2665,7 @@
EXPECT_TRUE(stereo_swapping_enabled);
// Set options again to ensure it has no impact.
- ASSERT_TRUE(engine_.SetOptions(options));
+ EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetNsStatus(ns_enabled, ns_mode);
EXPECT_TRUE(ec_enabled);
@@ -2785,9 +2736,9 @@
// AEC and AGC and NS
cricket::AudioSendParameters parameters_options_all = send_parameters_;
- parameters_options_all.options.echo_cancellation.Set(true);
- parameters_options_all.options.auto_gain_control.Set(true);
- parameters_options_all.options.noise_suppression.Set(true);
+ parameters_options_all.options.echo_cancellation = rtc::Optional<bool>(true);
+ parameters_options_all.options.auto_gain_control = rtc::Optional<bool>(true);
+ parameters_options_all.options.noise_suppression = rtc::Optional<bool>(true);
ASSERT_TRUE(channel1->SetSendParameters(parameters_options_all));
EXPECT_EQ(parameters_options_all.options, channel1->options());
ASSERT_TRUE(channel2->SetSendParameters(parameters_options_all));
@@ -2795,24 +2746,26 @@
// unset NS
cricket::AudioSendParameters parameters_options_no_ns = send_parameters_;
- parameters_options_no_ns.options.noise_suppression.Set(false);
+ parameters_options_no_ns.options.noise_suppression =
+ rtc::Optional<bool>(false);
ASSERT_TRUE(channel1->SetSendParameters(parameters_options_no_ns));
cricket::AudioOptions expected_options = parameters_options_all.options;
- expected_options.echo_cancellation.Set(true);
- expected_options.auto_gain_control.Set(true);
- expected_options.noise_suppression.Set(false);
+ expected_options.echo_cancellation = rtc::Optional<bool>(true);
+ expected_options.auto_gain_control = rtc::Optional<bool>(true);
+ expected_options.noise_suppression = rtc::Optional<bool>(false);
EXPECT_EQ(expected_options, channel1->options());
// unset AGC
cricket::AudioSendParameters parameters_options_no_agc = send_parameters_;
- parameters_options_no_agc.options.auto_gain_control.Set(false);
+ parameters_options_no_agc.options.auto_gain_control =
+ rtc::Optional<bool>(false);
ASSERT_TRUE(channel2->SetSendParameters(parameters_options_no_agc));
- expected_options.echo_cancellation.Set(true);
- expected_options.auto_gain_control.Set(false);
- expected_options.noise_suppression.Set(true);
+ expected_options.echo_cancellation = rtc::Optional<bool>(true);
+ expected_options.auto_gain_control = rtc::Optional<bool>(false);
+ expected_options.noise_suppression = rtc::Optional<bool>(true);
EXPECT_EQ(expected_options, channel2->options());
- ASSERT_TRUE(engine_.SetOptions(parameters_options_all.options));
+ ASSERT_TRUE(channel_->SetSendParameters(parameters_options_all));
bool ec_enabled;
webrtc::EcModes ec_mode;
bool agc_enabled;
@@ -2834,14 +2787,6 @@
EXPECT_TRUE(agc_enabled);
EXPECT_FALSE(ns_enabled);
- channel1->SetSend(cricket::SEND_NOTHING);
- voe_.GetEcStatus(ec_enabled, ec_mode);
- voe_.GetAgcStatus(agc_enabled, agc_mode);
- voe_.GetNsStatus(ns_enabled, ns_mode);
- EXPECT_TRUE(ec_enabled);
- EXPECT_TRUE(agc_enabled);
- EXPECT_TRUE(ns_enabled);
-
channel2->SetSend(cricket::SEND_MICROPHONE);
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAgcStatus(agc_enabled, agc_mode);
@@ -2850,25 +2795,19 @@
EXPECT_FALSE(agc_enabled);
EXPECT_TRUE(ns_enabled);
- channel2->SetSend(cricket::SEND_NOTHING);
- voe_.GetEcStatus(ec_enabled, ec_mode);
- voe_.GetAgcStatus(agc_enabled, agc_mode);
- voe_.GetNsStatus(ns_enabled, ns_mode);
- EXPECT_TRUE(ec_enabled);
- EXPECT_TRUE(agc_enabled);
- EXPECT_TRUE(ns_enabled);
-
// Make sure settings take effect while we are sending.
- ASSERT_TRUE(engine_.SetOptions(parameters_options_all.options));
+ ASSERT_TRUE(channel_->SetSendParameters(parameters_options_all));
cricket::AudioSendParameters parameters_options_no_agc_nor_ns =
send_parameters_;
- parameters_options_no_agc_nor_ns.options.auto_gain_control.Set(false);
- parameters_options_no_agc_nor_ns.options.noise_suppression.Set(false);
+ parameters_options_no_agc_nor_ns.options.auto_gain_control =
+ rtc::Optional<bool>(false);
+ parameters_options_no_agc_nor_ns.options.noise_suppression =
+ rtc::Optional<bool>(false);
channel2->SetSend(cricket::SEND_MICROPHONE);
channel2->SetSendParameters(parameters_options_no_agc_nor_ns);
- expected_options.echo_cancellation.Set(true);
- expected_options.auto_gain_control.Set(false);
- expected_options.noise_suppression.Set(false);
+ expected_options.echo_cancellation = rtc::Optional<bool>(true);
+ expected_options.auto_gain_control = rtc::Optional<bool>(false);
+ expected_options.noise_suppression = rtc::Optional<bool>(false);
EXPECT_EQ(expected_options, channel2->options());
voe_.GetEcStatus(ec_enabled, ec_mode);
voe_.GetAgcStatus(agc_enabled, agc_mode);
@@ -2887,13 +2826,13 @@
new cricket::FakeNetworkInterface);
channel->SetInterface(network_interface.get());
cricket::AudioSendParameters parameters = send_parameters_;
- parameters.options.dscp.Set(true);
+ parameters.options.dscp = rtc::Optional<bool>(true);
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_EQ(rtc::DSCP_EF, network_interface->dscp());
// Verify previous value is not modified if dscp option is not set.
EXPECT_TRUE(channel->SetSendParameters(send_parameters_));
EXPECT_EQ(rtc::DSCP_EF, network_interface->dscp());
- parameters.options.dscp.Set(false);
+ parameters.options.dscp = rtc::Optional<bool>(false);
EXPECT_TRUE(channel->SetSendParameters(parameters));
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
}
@@ -3002,7 +2941,7 @@
}
// Enable combined BWE option - now it should be set up.
- send_parameters_.options.combined_audio_video_bwe.Set(true);
+ send_parameters_.options.combined_audio_video_bwe = rtc::Optional<bool>(true);
EXPECT_TRUE(media_channel->SetSendParameters(send_parameters_));
for (uint32_t ssrc : ssrcs) {
const auto* s = call_.GetAudioReceiveStream(ssrc);
@@ -3011,7 +2950,8 @@
}
// Disable combined BWE option - should be disabled again.
- send_parameters_.options.combined_audio_video_bwe.Set(false);
+ send_parameters_.options.combined_audio_video_bwe =
+ rtc::Optional<bool>(false);
EXPECT_TRUE(media_channel->SetSendParameters(send_parameters_));
for (uint32_t ssrc : ssrcs) {
const auto* s = call_.GetAudioReceiveStream(ssrc);
@@ -3028,18 +2968,19 @@
EXPECT_TRUE(SetupEngineWithSendStream());
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
- send_parameters_.options.combined_audio_video_bwe.Set(true);
+ send_parameters_.options.combined_audio_video_bwe = rtc::Optional<bool>(true);
EXPECT_TRUE(media_channel->SetSendParameters(send_parameters_));
- static const uint32_t kSsrcs[] = {1, 2, 3, 4};
- for (unsigned int i = 0; i < ARRAY_SIZE(kSsrcs); ++i) {
+ for (uint32_t ssrc : kSsrcs4) {
EXPECT_TRUE(media_channel->AddRecvStream(
- cricket::StreamParams::CreateLegacy(kSsrcs[i])));
- EXPECT_NE(nullptr, call_.GetAudioReceiveStream(kSsrcs[i]));
+ cricket::StreamParams::CreateLegacy(ssrc)));
+ EXPECT_NE(nullptr, call_.GetAudioReceiveStream(ssrc));
}
- EXPECT_EQ(ARRAY_SIZE(kSsrcs), call_.GetAudioReceiveStreams().size());
+ EXPECT_EQ(arraysize(kSsrcs4), call_.GetAudioReceiveStreams().size());
}
+// TODO(solenberg): Remove, once recv streams are configured through Call.
+// (This is then covered by TestSetRecvRtpHeaderExtensions.)
TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
// Test that setting the header extensions results in the expected state
// changes on an associated Call.
@@ -3050,7 +2991,7 @@
EXPECT_TRUE(SetupEngineWithSendStream());
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
- send_parameters_.options.combined_audio_video_bwe.Set(true);
+ send_parameters_.options.combined_audio_video_bwe = rtc::Optional<bool>(true);
EXPECT_TRUE(media_channel->SetSendParameters(send_parameters_));
for (uint32_t ssrc : ssrcs) {
EXPECT_TRUE(media_channel->AddRecvStream(
@@ -3066,17 +3007,17 @@
}
// Set up receive extensions.
- const auto& e_exts = engine_.rtp_header_extensions();
+ cricket::RtpCapabilities capabilities = engine_.GetCapabilities();
cricket::AudioRecvParameters recv_parameters;
- recv_parameters.extensions = e_exts;
+ recv_parameters.extensions = capabilities.header_extensions;
channel_->SetRecvParameters(recv_parameters);
EXPECT_EQ(2, call_.GetAudioReceiveStreams().size());
for (uint32_t ssrc : ssrcs) {
const auto* s = call_.GetAudioReceiveStream(ssrc);
EXPECT_NE(nullptr, s);
const auto& s_exts = s->GetConfig().rtp.extensions;
- EXPECT_EQ(e_exts.size(), s_exts.size());
- for (const auto& e_ext : e_exts) {
+ EXPECT_EQ(capabilities.header_extensions.size(), s_exts.size());
+ for (const auto& e_ext : capabilities.header_extensions) {
for (const auto& s_ext : s_exts) {
if (e_ext.id == s_ext.id) {
EXPECT_EQ(e_ext.uri, s_ext.name);
@@ -3109,7 +3050,7 @@
EXPECT_TRUE(SetupEngineWithSendStream());
cricket::WebRtcVoiceMediaChannel* media_channel =
static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
- send_parameters_.options.combined_audio_video_bwe.Set(true);
+ send_parameters_.options.combined_audio_video_bwe = rtc::Optional<bool>(true);
EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
EXPECT_TRUE(media_channel->AddRecvStream(
cricket::StreamParams::CreateLegacy(kAudioSsrc)));
@@ -3164,18 +3105,6 @@
EXPECT_EQ(voe_.GetAssociateSendChannel(recv_ch), -1);
}
-// Tests for the actual WebRtc VoE library.
-
-TEST(WebRtcVoiceEngineTest, TestDefaultOptionsBeforeInit) {
- cricket::WebRtcVoiceEngine engine;
- cricket::AudioOptions options = engine.GetOptions();
- // The default options should have at least a few things set. We purposefully
- // don't check the option values here, though.
- EXPECT_TRUE(options.echo_cancellation.IsSet());
- EXPECT_TRUE(options.auto_gain_control.IsSet());
- EXPECT_TRUE(options.noise_suppression.IsSet());
-}
-
// Tests that the library initializes and shuts down properly.
TEST(WebRtcVoiceEngineTest, StartupShutdown) {
cricket::WebRtcVoiceEngine engine;
@@ -3195,54 +3124,60 @@
// Tests that the library is configured with the codecs we want.
TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
- cricket::WebRtcVoiceEngine engine;
// Check codecs by name.
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "OPUS", 48000, 0, 2, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "ISAC", 16000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "ISAC", 32000, 0, 1, 0)));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "OPUS", 48000, 0, 2, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "ISAC", 16000, 0, 1, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "ISAC", 32000, 0, 1, 0), nullptr));
// Check that name matching is case-insensitive.
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "ILBC", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "iLBC", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "PCMU", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "PCMA", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "G722", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "red", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "CN", 32000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "CN", 16000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "CN", 8000, 0, 1, 0)));
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(96, "telephone-event", 8000, 0, 1, 0)));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "ILBC", 8000, 0, 1, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "iLBC", 8000, 0, 1, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "PCMU", 8000, 0, 1, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "PCMA", 8000, 0, 1, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "G722", 8000, 0, 1, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "red", 8000, 0, 1, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "CN", 32000, 0, 1, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "CN", 16000, 0, 1, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "CN", 8000, 0, 1, 0), nullptr));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(96, "telephone-event", 8000, 0, 1, 0), nullptr));
// Check codecs with an id by id.
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(0, "", 8000, 0, 1, 0))); // PCMU
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(8, "", 8000, 0, 1, 0))); // PCMA
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(9, "", 8000, 0, 1, 0))); // G722
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(13, "", 8000, 0, 1, 0))); // CN
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(0, "", 8000, 0, 1, 0), nullptr)); // PCMU
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(8, "", 8000, 0, 1, 0), nullptr)); // PCMA
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(9, "", 8000, 0, 1, 0), nullptr)); // G722
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(13, "", 8000, 0, 1, 0), nullptr)); // CN
// Check sample/bitrate matching.
- EXPECT_TRUE(engine.FindCodec(
- cricket::AudioCodec(0, "PCMU", 8000, 64000, 1, 0)));
+ EXPECT_TRUE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(0, "PCMU", 8000, 64000, 1, 0), nullptr));
// Check that bad codecs fail.
- EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(99, "ABCD", 0, 0, 1, 0)));
- EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(88, "", 0, 0, 1, 0)));
- EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(0, "", 0, 0, 2, 0)));
- EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(0, "", 5000, 0, 1, 0)));
- EXPECT_FALSE(engine.FindCodec(cricket::AudioCodec(0, "", 0, 5000, 1, 0)));
+ EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(99, "ABCD", 0, 0, 1, 0), nullptr));
+ EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(88, "", 0, 0, 1, 0), nullptr));
+ EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(0, "", 0, 0, 2, 0), nullptr));
+ EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(0, "", 5000, 0, 1, 0), nullptr));
+ EXPECT_FALSE(cricket::WebRtcVoiceEngine::ToCodecInst(
+ cricket::AudioCodec(0, "", 0, 5000, 1, 0), nullptr));
+
// Verify the payload id of common audio codecs, including CN, ISAC, and G722.
+ cricket::WebRtcVoiceEngine engine;
for (std::vector<cricket::AudioCodec>::const_iterator it =
engine.codecs().begin(); it != engine.codecs().end(); ++it) {
if (it->name == "CN" && it->clockrate == 16000) {
@@ -3269,7 +3204,6 @@
EXPECT_EQ("1", it->params.find("useinbandfec")->second);
}
}
-
engine.Terminate();
}
@@ -3282,7 +3216,7 @@
cricket::VoiceMediaChannel* channels[32];
int num_channels = 0;
- while (num_channels < ARRAY_SIZE(channels)) {
+ while (num_channels < arraysize(channels)) {
cricket::VoiceMediaChannel* channel =
engine.CreateChannel(call.get(), cricket::AudioOptions());
if (!channel)
@@ -3290,7 +3224,7 @@
channels[num_channels++] = channel;
}
- int expected = ARRAY_SIZE(channels);
+ int expected = arraysize(channels);
EXPECT_EQ(expected, num_channels);
while (num_channels > 0) {
diff --git a/talk/session/media/bundlefilter.cc b/talk/session/media/bundlefilter.cc
index b47d47f..670befe 100755
--- a/talk/session/media/bundlefilter.cc
+++ b/talk/session/media/bundlefilter.cc
@@ -32,78 +32,29 @@
namespace cricket {
-static const uint32_t kSsrc01 = 0x01;
-
BundleFilter::BundleFilter() {
}
BundleFilter::~BundleFilter() {
}
-bool BundleFilter::DemuxPacket(const char* data, size_t len, bool rtcp) {
- // For rtp packets, we check whether the payload type can be found.
- // For rtcp packets, we check whether the ssrc can be found or is the special
- // value 1 except for SDES packets which always pass through. Plus, if
- // |streams_| is empty, we will allow all rtcp packets pass through provided
- // that they are valid rtcp packets in case that they are for early media.
- if (!rtcp) {
- // It may not be a RTP packet (e.g. SCTP).
- if (!IsRtpPacket(data, len))
- return false;
-
- int payload_type = 0;
- if (!GetRtpPayloadType(data, len, &payload_type)) {
- return false;
- }
- return FindPayloadType(payload_type);
+bool BundleFilter::DemuxPacket(const uint8_t* data, size_t len) {
+ // For RTP packets, we check whether the payload type can be found.
+ if (!IsRtpPacket(data, len)) {
+ return false;
}
- // Rtcp packets using ssrc filter.
- int pl_type = 0;
- uint32_t ssrc = 0;
- if (!GetRtcpType(data, len, &pl_type)) return false;
- if (pl_type == kRtcpTypeSDES) {
- // SDES packet parsing not supported.
- LOG(LS_INFO) << "SDES packet received for demux.";
- return true;
- } else {
- if (!GetRtcpSsrc(data, len, &ssrc)) return false;
- if (ssrc == kSsrc01) {
- // SSRC 1 has a special meaning and indicates generic feedback on
- // some systems and should never be dropped. If it is forwarded
- // incorrectly it will be ignored by lower layers anyway.
- return true;
- }
+ int payload_type = 0;
+ if (!GetRtpPayloadType(data, len, &payload_type)) {
+ return false;
}
- // Pass through if |streams_| is empty to allow early rtcp packets in.
- return !HasStreams() || FindStream(ssrc);
+ return FindPayloadType(payload_type);
}
void BundleFilter::AddPayloadType(int payload_type) {
payload_types_.insert(payload_type);
}
-bool BundleFilter::AddStream(const StreamParams& stream) {
- if (GetStreamBySsrc(streams_, stream.first_ssrc())) {
- LOG(LS_WARNING) << "Stream already added to filter";
- return false;
- }
- streams_.push_back(stream);
- return true;
-}
-
-bool BundleFilter::RemoveStream(uint32_t ssrc) {
- return RemoveStreamBySsrc(&streams_, ssrc);
-}
-
-bool BundleFilter::HasStreams() const {
- return !streams_.empty();
-}
-
-bool BundleFilter::FindStream(uint32_t ssrc) const {
- return ssrc == 0 ? false : GetStreamBySsrc(streams_, ssrc) != nullptr;
-}
-
bool BundleFilter::FindPayloadType(int pl_type) const {
return payload_types_.find(pl_type) != payload_types_.end();
}
diff --git a/talk/session/media/bundlefilter.h b/talk/session/media/bundlefilter.h
index 3717376..d9d952f 100755
--- a/talk/session/media/bundlefilter.h
+++ b/talk/session/media/bundlefilter.h
@@ -28,6 +28,8 @@
#ifndef TALK_SESSION_MEDIA_BUNDLEFILTER_H_
#define TALK_SESSION_MEDIA_BUNDLEFILTER_H_
+#include <stdint.h>
+
#include <set>
#include <vector>
@@ -37,42 +39,31 @@
namespace cricket {
// In case of single RTP session and single transport channel, all session
-// ( or media) channels share a common transport channel. Hence they all get
+// (or media) channels share a common transport channel. Hence they all get
// SignalReadPacket when packet received on transport channel. This requires
// cricket::BaseChannel to know all the valid sources, else media channel
// will decode invalid packets.
//
// This class determines whether a packet is destined for cricket::BaseChannel.
-// For rtp packets, this is decided based on the payload type. For rtcp packets,
-// this is decided based on the sender ssrc values.
+// This is only to be used for RTP packets as RTCP packets are not filtered.
+// For RTP packets, this is decided based on the payload type.
class BundleFilter {
public:
BundleFilter();
~BundleFilter();
- // Determines packet belongs to valid cricket::BaseChannel.
- bool DemuxPacket(const char* data, size_t len, bool rtcp);
+ // Determines if a RTP packet belongs to valid cricket::BaseChannel.
+ bool DemuxPacket(const uint8_t* data, size_t len);
// Adds the supported payload type.
void AddPayloadType(int payload_type);
- // Adding a valid source to the filter.
- bool AddStream(const StreamParams& stream);
-
- // Removes source from the filter.
- bool RemoveStream(uint32_t ssrc);
-
- // Utility methods added for unitest.
- // True if |streams_| is not empty.
- bool HasStreams() const;
- bool FindStream(uint32_t ssrc) const;
+ // Public for unittests.
bool FindPayloadType(int pl_type) const;
void ClearAllPayloadTypes();
-
private:
std::set<int> payload_types_;
- std::vector<StreamParams> streams_;
};
} // namespace cricket
diff --git a/talk/session/media/bundlefilter_unittest.cc b/talk/session/media/bundlefilter_unittest.cc
index 806d6ba..f2c35fc 100755
--- a/talk/session/media/bundlefilter_unittest.cc
+++ b/talk/session/media/bundlefilter_unittest.cc
@@ -30,9 +30,6 @@
using cricket::StreamParams;
-static const int kSsrc1 = 0x1111;
-static const int kSsrc2 = 0x2222;
-static const int kSsrc3 = 0x3333;
static const int kPayloadType1 = 0x11;
static const int kPayloadType2 = 0x22;
static const int kPayloadType3 = 0x33;
@@ -55,56 +52,6 @@
0x22,
};
-// PT = 200 = SR, len = 28, SSRC of sender = 0x0001
-// NTP TS = 0, RTP TS = 0, packet count = 0
-static const unsigned char kRtcpPacketSrSsrc01[] = {
- 0x80, 0xC8, 0x00, 0x1B, 0x00, 0x00, 0x00, 0x01,
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
-};
-
-// PT = 200 = SR, len = 28, SSRC of sender = 0x2222
-// NTP TS = 0, RTP TS = 0, packet count = 0
-static const unsigned char kRtcpPacketSrSsrc2[] = {
- 0x80, 0xC8, 0x00, 0x1B, 0x00, 0x00, 0x22, 0x22,
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
-};
-
-// First packet - SR = PT = 200, len = 0, SSRC of sender = 0x1111
-// NTP TS = 0, RTP TS = 0, packet count = 0
-// second packet - SDES = PT = 202, count = 0, SSRC = 0x1111, cname len = 0
-static const unsigned char kRtcpPacketCompoundSrSdesSsrc1[] = {
- 0x80, 0xC8, 0x00, 0x01, 0x00, 0x00, 0x11, 0x11,
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x81, 0xCA, 0x00, 0x00, 0x00, 0x00, 0x11, 0x11, 0x01, 0x00,
-};
-
-// SDES = PT = 202, count = 0, SSRC = 0x2222, cname len = 0
-static const unsigned char kRtcpPacketSdesSsrc2[] = {
- 0x81, 0xCA, 0x00, 0x00, 0x00, 0x00, 0x22, 0x22, 0x01, 0x00,
-};
-
-// Packet has only mandatory fixed RTCP header
-static const unsigned char kRtcpPacketFixedHeaderOnly[] = {
- 0x80, 0xC8, 0x00, 0x00,
-};
-
-// Small packet for SSRC demux.
-static const unsigned char kRtcpPacketTooSmall[] = {
- 0x80, 0xC8, 0x00, 0x00, 0x00, 0x00,
-};
-
-// PT = 206, FMT = 1, Sender SSRC = 0x1111, Media SSRC = 0x1111
-// No FCI information is needed for PLI.
-static const unsigned char kRtcpPacketNonCompoundRtcpPliFeedback[] = {
- 0x81, 0xCE, 0x00, 0x0C, 0x00, 0x00, 0x11, 0x11, 0x00, 0x00, 0x11, 0x11,
-};
-
// An SCTP packet.
static const unsigned char kSctpPacket[] = {
0x00, 0x01, 0x00, 0x01,
@@ -114,100 +61,29 @@
0x00, 0x00, 0x00, 0x00,
};
-TEST(BundleFilterTest, AddRemoveStreamTest) {
- cricket::BundleFilter bundle_filter;
- EXPECT_FALSE(bundle_filter.HasStreams());
- EXPECT_TRUE(bundle_filter.AddStream(StreamParams::CreateLegacy(kSsrc1)));
- StreamParams stream2;
- stream2.ssrcs.push_back(kSsrc2);
- stream2.ssrcs.push_back(kSsrc3);
- EXPECT_TRUE(bundle_filter.AddStream(stream2));
-
- EXPECT_TRUE(bundle_filter.HasStreams());
- EXPECT_TRUE(bundle_filter.FindStream(kSsrc1));
- EXPECT_TRUE(bundle_filter.FindStream(kSsrc2));
- EXPECT_TRUE(bundle_filter.FindStream(kSsrc3));
- EXPECT_TRUE(bundle_filter.RemoveStream(kSsrc1));
- EXPECT_FALSE(bundle_filter.FindStream(kSsrc1));
- EXPECT_TRUE(bundle_filter.RemoveStream(kSsrc3));
- EXPECT_FALSE(bundle_filter.RemoveStream(kSsrc2)); // Already removed.
- EXPECT_FALSE(bundle_filter.HasStreams());
-}
-
TEST(BundleFilterTest, RtpPacketTest) {
cricket::BundleFilter bundle_filter;
bundle_filter.AddPayloadType(kPayloadType1);
- EXPECT_TRUE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtpPacketPt1Ssrc1),
- sizeof(kRtpPacketPt1Ssrc1), false));
+ EXPECT_TRUE(bundle_filter.DemuxPacket(kRtpPacketPt1Ssrc1,
+ sizeof(kRtpPacketPt1Ssrc1)));
bundle_filter.AddPayloadType(kPayloadType2);
- EXPECT_TRUE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtpPacketPt2Ssrc2),
- sizeof(kRtpPacketPt2Ssrc2), false));
+ EXPECT_TRUE(bundle_filter.DemuxPacket(kRtpPacketPt2Ssrc2,
+ sizeof(kRtpPacketPt2Ssrc2)));
// Payload type 0x33 is not added.
- EXPECT_FALSE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtpPacketPt3Ssrc2),
- sizeof(kRtpPacketPt3Ssrc2), false));
+ EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt3Ssrc2,
+ sizeof(kRtpPacketPt3Ssrc2)));
// Size is too small.
- EXPECT_FALSE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtpPacketPt1Ssrc1), 11, false));
+ EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt1Ssrc1, 11));
bundle_filter.ClearAllPayloadTypes();
- EXPECT_FALSE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtpPacketPt1Ssrc1),
- sizeof(kRtpPacketPt1Ssrc1), false));
- EXPECT_FALSE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtpPacketPt2Ssrc2),
- sizeof(kRtpPacketPt2Ssrc2), false));
-}
-
-TEST(BundleFilterTest, RtcpPacketTest) {
- cricket::BundleFilter bundle_filter;
- EXPECT_TRUE(bundle_filter.AddStream(StreamParams::CreateLegacy(kSsrc1)));
- EXPECT_TRUE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtcpPacketCompoundSrSdesSsrc1),
- sizeof(kRtcpPacketCompoundSrSdesSsrc1), true));
- EXPECT_TRUE(bundle_filter.AddStream(StreamParams::CreateLegacy(kSsrc2)));
- EXPECT_TRUE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtcpPacketSrSsrc2),
- sizeof(kRtcpPacketSrSsrc2), true));
- EXPECT_TRUE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtcpPacketSdesSsrc2),
- sizeof(kRtcpPacketSdesSsrc2), true));
- EXPECT_TRUE(bundle_filter.RemoveStream(kSsrc2));
- // RTCP Packets other than SR and RR are demuxed regardless of SSRC.
- EXPECT_TRUE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtcpPacketSdesSsrc2),
- sizeof(kRtcpPacketSdesSsrc2), true));
- // RTCP Packets with 'special' SSRC 0x01 are demuxed also
- EXPECT_TRUE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtcpPacketSrSsrc01),
- sizeof(kRtcpPacketSrSsrc01), true));
- EXPECT_FALSE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtcpPacketSrSsrc2),
- sizeof(kRtcpPacketSrSsrc2), true));
- EXPECT_FALSE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtcpPacketFixedHeaderOnly),
- sizeof(kRtcpPacketFixedHeaderOnly), true));
- EXPECT_FALSE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtcpPacketTooSmall),
- sizeof(kRtcpPacketTooSmall), true));
- EXPECT_TRUE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtcpPacketNonCompoundRtcpPliFeedback),
- sizeof(kRtcpPacketNonCompoundRtcpPliFeedback), true));
- // If the streams_ is empty, rtcp packet passes through
- EXPECT_TRUE(bundle_filter.RemoveStream(kSsrc1));
- EXPECT_FALSE(bundle_filter.HasStreams());
- EXPECT_TRUE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kRtcpPacketSrSsrc2),
- sizeof(kRtcpPacketSrSsrc2), true));
+ EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt1Ssrc1,
+ sizeof(kRtpPacketPt1Ssrc1)));
+ EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt2Ssrc2,
+ sizeof(kRtpPacketPt2Ssrc2)));
}
TEST(BundleFilterTest, InvalidRtpPacket) {
cricket::BundleFilter bundle_filter;
- EXPECT_TRUE(bundle_filter.AddStream(StreamParams::CreateLegacy(kSsrc1)));
- EXPECT_FALSE(bundle_filter.DemuxPacket(
- reinterpret_cast<const char*>(kSctpPacket),
- sizeof(kSctpPacket), false));
+ EXPECT_FALSE(bundle_filter.DemuxPacket(kSctpPacket, sizeof(kSctpPacket)));
}
diff --git a/talk/session/media/channel.cc b/talk/session/media/channel.cc
index 91a6d8c..a59c3f8 100644
--- a/talk/session/media/channel.cc
+++ b/talk/session/media/channel.cc
@@ -25,23 +25,36 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
+#include <utility>
+
#include "talk/session/media/channel.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/rtputils.h"
-#include "webrtc/p2p/base/transportchannel.h"
#include "talk/session/media/channelmanager.h"
+#include "webrtc/audio/audio_sink.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/common.h"
#include "webrtc/base/dscp.h"
#include "webrtc/base/logging.h"
+#include "webrtc/base/trace_event.h"
+#include "webrtc/p2p/base/transportchannel.h"
namespace cricket {
-
using rtc::Bind;
+namespace {
+// See comment below for why we need to use a pointer to a scoped_ptr.
+bool SetRawAudioSink_w(VoiceMediaChannel* channel,
+ uint32_t ssrc,
+ rtc::scoped_ptr<webrtc::AudioSinkInterface>* sink) {
+ channel->SetRawAudioSink(ssrc, std::move(*sink));
+ return true;
+}
+} // namespace
+
enum {
MSG_EARLYMEDIATIMEOUT = 1,
MSG_SCREENCASTWINDOWEVENT,
@@ -101,15 +114,6 @@
DataMediaChannel::Error error;
};
-
-struct VideoChannel::ScreencastDetailsData {
- explicit ScreencastDetailsData(uint32_t s)
- : ssrc(s), fps(0), screencast_max_pixels(0) {}
- uint32_t ssrc;
- int fps;
- int screencast_max_pixels;
-};
-
static const char* PacketType(bool rtcp) {
return (!rtcp) ? "RTP" : "RTCP";
}
@@ -151,6 +155,7 @@
if (desc->rtp_header_extensions_set()) {
params->extensions = desc->rtp_header_extensions();
}
+ params->rtcp.reduced_size = desc->rtcp_reduced_size();
}
template <class Codec, class Options>
@@ -218,11 +223,11 @@
return false;
}
- if (!SetDtlsSrtpCiphers(transport_channel(), false)) {
+ if (!SetDtlsSrtpCryptoSuites(transport_channel(), false)) {
return false;
}
if (rtcp_transport_enabled() &&
- !SetDtlsSrtpCiphers(rtcp_transport_channel(), true)) {
+ !SetDtlsSrtpCryptoSuites(rtcp_transport_channel(), true)) {
return false;
}
@@ -249,21 +254,43 @@
return true;
}
- set_transport_channel(transport_controller_->CreateTransportChannel_w(
- transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP));
- if (!transport_channel()) {
- return false;
+ // When using DTLS-SRTP, we must reset the SrtpFilter every time the transport
+ // changes and wait until the DTLS handshake is complete to set the newly
+ // negotiated parameters.
+ if (ShouldSetupDtlsSrtp()) {
+ // Set |writable_| to false such that UpdateWritableState_w can set up
+ // DTLS-SRTP when the writable_ becomes true again.
+ writable_ = false;
+ srtp_filter_.ResetParams();
}
+
+ // TODO(guoweis): Remove this grossness when we remove non-muxed RTCP.
if (rtcp_transport_enabled()) {
LOG(LS_INFO) << "Create RTCP TransportChannel for " << content_name()
<< " on " << transport_name << " transport ";
- set_rtcp_transport_channel(transport_controller_->CreateTransportChannel_w(
- transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP));
+ set_rtcp_transport_channel(
+ transport_controller_->CreateTransportChannel_w(
+ transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP),
+ false /* update_writablity */);
if (!rtcp_transport_channel()) {
return false;
}
}
+ // We're not updating the writablity during the transition state.
+ set_transport_channel(transport_controller_->CreateTransportChannel_w(
+ transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP));
+ if (!transport_channel()) {
+ return false;
+ }
+
+ // TODO(guoweis): Remove this grossness when we remove non-muxed RTCP.
+ if (rtcp_transport_enabled()) {
+ // We can only update the RTCP ready to send after set_transport_channel has
+ // handled channel writability.
+ SetReadyToSend(
+ true, rtcp_transport_channel() && rtcp_transport_channel()->writable());
+ }
transport_name_ = transport_name;
return true;
}
@@ -299,7 +326,8 @@
SetReadyToSend(false, new_tc && new_tc->writable());
}
-void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc) {
+void BaseChannel::set_rtcp_transport_channel(TransportChannel* new_tc,
+ bool update_writablity) {
ASSERT(worker_thread_ == rtc::Thread::Current());
TransportChannel* old_tc = rtcp_transport_channel_;
@@ -318,16 +346,21 @@
rtcp_transport_channel_ = new_tc;
if (new_tc) {
+ RTC_CHECK(!(ShouldSetupDtlsSrtp() && srtp_filter_.IsActive()))
+ << "Setting RTCP for DTLS/SRTP after SrtpFilter is active "
+ << "should never happen.";
ConnectToTransportChannel(new_tc);
for (const auto& pair : rtcp_socket_options_) {
new_tc->SetOption(pair.first, pair.second);
}
}
- // Update aggregate writable/ready-to-send state between RTP and RTCP upon
- // setting new channel
- UpdateWritableState_w();
- SetReadyToSend(true, new_tc && new_tc->writable());
+ if (update_writablity) {
+ // Update aggregate writable/ready-to-send state between RTP and RTCP upon
+ // setting new channel
+ UpdateWritableState_w();
+ SetReadyToSend(true, new_tc && new_tc->writable());
+ }
}
void BaseChannel::ConnectToTransportChannel(TransportChannel* tc) {
@@ -336,6 +369,7 @@
tc->SignalWritableState.connect(this, &BaseChannel::OnWritableState);
tc->SignalReadPacket.connect(this, &BaseChannel::OnChannelRead);
tc->SignalReadyToSend.connect(this, &BaseChannel::OnReadyToSend);
+ tc->SignalDtlsState.connect(this, &BaseChannel::OnDtlsState);
}
void BaseChannel::DisconnectFromTransportChannel(TransportChannel* tc) {
@@ -344,6 +378,7 @@
tc->SignalWritableState.disconnect(this);
tc->SignalReadPacket.disconnect(this);
tc->SignalReadyToSend.disconnect(this);
+ tc->SignalDtlsState.disconnect(this);
}
bool BaseChannel::Enable(bool enable) {
@@ -374,6 +409,7 @@
bool BaseChannel::SetLocalContent(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
+ TRACE_EVENT0("webrtc", "BaseChannel::SetLocalContent");
return InvokeOnWorker(Bind(&BaseChannel::SetLocalContent_w,
this, content, action, error_desc));
}
@@ -381,6 +417,7 @@
bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
+ TRACE_EVENT0("webrtc", "BaseChannel::SetRemoteContent");
return InvokeOnWorker(Bind(&BaseChannel::SetRemoteContent_w,
this, content, action, error_desc));
}
@@ -416,10 +453,10 @@
bool BaseChannel::IsReadyToSend() const {
// Send outgoing data if we are enabled, have local and remote content,
// and we have had some form of connectivity.
- return enabled() &&
- IsReceiveContentDirection(remote_content_direction_) &&
+ return enabled() && IsReceiveContentDirection(remote_content_direction_) &&
IsSendContentDirection(local_content_direction_) &&
- was_ever_writable();
+ was_ever_writable() &&
+ (srtp_filter_.IsActive() || !ShouldSetupDtlsSrtp());
}
bool BaseChannel::SendPacket(rtc::Buffer* packet,
@@ -459,6 +496,7 @@
const char* data, size_t len,
const rtc::PacketTime& packet_time,
int flags) {
+ TRACE_EVENT0("webrtc", "BaseChannel::OnChannelRead");
// OnChannelRead gets called from P2PSocket; now pass data to MediaEngine
ASSERT(worker_thread_ == rtc::Thread::Current());
@@ -474,6 +512,22 @@
SetReadyToSend(channel == rtcp_transport_channel_, true);
}
+void BaseChannel::OnDtlsState(TransportChannel* channel,
+ DtlsTransportState state) {
+ if (!ShouldSetupDtlsSrtp()) {
+ return;
+ }
+
+ // Reset the srtp filter if it's not the CONNECTED state. For the CONNECTED
+ // state, setting up DTLS-SRTP context is deferred to ChannelWritable_w to
+ // cover other scenarios like the whole channel is writable (not just this
+ // TransportChannel) or when TransportChannel is attached after DTLS is
+ // negotiated.
+ if (state != DTLS_TRANSPORT_CONNECTED) {
+ srtp_filter_.ResetParams();
+ }
+}
+
void BaseChannel::SetReadyToSend(bool rtcp, bool ready) {
if (rtcp) {
rtcp_ready_to_send_ = ready;
@@ -512,7 +566,7 @@
// Avoid a copy by transferring the ownership of the packet data.
int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
PacketMessageData* data = new PacketMessageData;
- data->packet = packet->Pass();
+ data->packet = std::move(*packet);
data->options = options;
worker_thread_->Post(this, message_id, data);
return true;
@@ -628,9 +682,12 @@
<< " packet: wrong size=" << packet->size();
return false;
}
-
- // Bundle filter handles both rtp and rtcp packets.
- return bundle_filter_.DemuxPacket(packet->data<char>(), packet->size(), rtcp);
+ if (rtcp) {
+ // Permit all (seemingly valid) RTCP packets.
+ return true;
+ }
+ // Check whether we handle this payload.
+ return bundle_filter_.DemuxPacket(packet->data<uint8_t>(), packet->size());
}
void BaseChannel::HandlePacket(bool rtcp, rtc::Buffer* packet,
@@ -758,8 +815,9 @@
void BaseChannel::ChannelWritable_w() {
ASSERT(worker_thread_ == rtc::Thread::Current());
- if (writable_)
+ if (writable_) {
return;
+ }
LOG(LS_INFO) << "Channel writable (" << content_name_ << ")"
<< (was_ever_writable_ ? "" : " for the first time");
@@ -775,22 +833,8 @@
}
}
- // If we're doing DTLS-SRTP, now is the time.
- if (!was_ever_writable_ && ShouldSetupDtlsSrtp()) {
- if (!SetupDtlsSrtp(false)) {
- SignalDtlsSetupFailure_w(false);
- return;
- }
-
- if (rtcp_transport_channel_) {
- if (!SetupDtlsSrtp(true)) {
- SignalDtlsSetupFailure_w(true);
- return;
- }
- }
- }
-
was_ever_writable_ = true;
+ MaybeSetupDtlsSrtp_w();
writable_ = true;
ChangeState();
}
@@ -806,20 +850,21 @@
SignalDtlsSetupFailure(this, rtcp);
}
-bool BaseChannel::SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp) {
- std::vector<std::string> ciphers;
- // We always use the default SRTP ciphers for RTCP, but we may use different
- // ciphers for RTP depending on the media type.
+bool BaseChannel::SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp) {
+ std::vector<int> crypto_suites;
+ // We always use the default SRTP crypto suites for RTCP, but we may use
+ // different crypto suites for RTP depending on the media type.
if (!rtcp) {
- GetSrtpCryptoSuiteNames(&ciphers);
+ GetSrtpCryptoSuites(&crypto_suites);
} else {
- GetDefaultSrtpCryptoSuiteNames(&ciphers);
+ GetDefaultSrtpCryptoSuites(&crypto_suites);
}
- return tc->SetSrtpCiphers(ciphers);
+ return tc->SetSrtpCryptoSuites(crypto_suites);
}
bool BaseChannel::ShouldSetupDtlsSrtp() const {
- return true;
+ // Since DTLS is applied to all channels, checking RTP should be enough.
+ return transport_channel_ && transport_channel_->IsDtlsActive();
}
// This function returns true if either DTLS-SRTP is not in use
@@ -830,14 +875,12 @@
TransportChannel* channel =
rtcp_channel ? rtcp_transport_channel_ : transport_channel_;
- // No DTLS
- if (!channel->IsDtlsActive())
- return true;
+ RTC_DCHECK(channel->IsDtlsActive());
- std::string selected_cipher;
+ int selected_crypto_suite;
- if (!channel->GetSrtpCryptoSuite(&selected_cipher)) {
- LOG(LS_ERROR) << "No DTLS-SRTP selected cipher";
+ if (!channel->GetSrtpCryptoSuite(&selected_crypto_suite)) {
+ LOG(LS_ERROR) << "No DTLS-SRTP selected crypto suite";
return false;
}
@@ -893,21 +936,15 @@
}
if (rtcp_channel) {
- ret = srtp_filter_.SetRtcpParams(
- selected_cipher,
- &(*send_key)[0],
- static_cast<int>(send_key->size()),
- selected_cipher,
- &(*recv_key)[0],
- static_cast<int>(recv_key->size()));
+ ret = srtp_filter_.SetRtcpParams(selected_crypto_suite, &(*send_key)[0],
+ static_cast<int>(send_key->size()),
+ selected_crypto_suite, &(*recv_key)[0],
+ static_cast<int>(recv_key->size()));
} else {
- ret = srtp_filter_.SetRtpParams(
- selected_cipher,
- &(*send_key)[0],
- static_cast<int>(send_key->size()),
- selected_cipher,
- &(*recv_key)[0],
- static_cast<int>(recv_key->size()));
+ ret = srtp_filter_.SetRtpParams(selected_crypto_suite, &(*send_key)[0],
+ static_cast<int>(send_key->size()),
+ selected_crypto_suite, &(*recv_key)[0],
+ static_cast<int>(recv_key->size()));
}
if (!ret)
@@ -918,6 +955,28 @@
return ret;
}
+void BaseChannel::MaybeSetupDtlsSrtp_w() {
+ if (srtp_filter_.IsActive()) {
+ return;
+ }
+
+ if (!ShouldSetupDtlsSrtp()) {
+ return;
+ }
+
+ if (!SetupDtlsSrtp(false)) {
+ SignalDtlsSetupFailure_w(false);
+ return;
+ }
+
+ if (rtcp_transport_channel_) {
+ if (!SetupDtlsSrtp(true)) {
+ SignalDtlsSetupFailure_w(true);
+ return;
+ }
+ }
+}
+
void BaseChannel::ChannelNotWritable_w() {
ASSERT(worker_thread_ == rtc::Thread::Current());
if (!writable_)
@@ -1022,7 +1081,7 @@
void BaseChannel::ActivateRtcpMux_w() {
if (!rtcp_mux_filter_.IsActive()) {
rtcp_mux_filter_.SetActive();
- set_rtcp_transport_channel(nullptr);
+ set_rtcp_transport_channel(nullptr, true);
rtcp_transport_enabled_ = false;
}
}
@@ -1045,7 +1104,7 @@
LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name()
<< " by destroying RTCP transport channel for "
<< transport_name();
- set_rtcp_transport_channel(nullptr);
+ set_rtcp_transport_channel(nullptr, true);
rtcp_transport_enabled_ = false;
}
break;
@@ -1075,15 +1134,11 @@
bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
ASSERT(worker_thread() == rtc::Thread::Current());
- if (!media_channel()->AddRecvStream(sp))
- return false;
-
- return bundle_filter_.AddStream(sp);
+ return media_channel()->AddRecvStream(sp);
}
bool BaseChannel::RemoveRecvStream_w(uint32_t ssrc) {
ASSERT(worker_thread() == rtc::Thread::Current());
- bundle_filter_.RemoveStream(ssrc);
return media_channel()->RemoveRecvStream(ssrc);
}
@@ -1243,6 +1298,7 @@
}
void BaseChannel::OnMessage(rtc::Message *pmsg) {
+ TRACE_EVENT0("webrtc", "BaseChannel::OnMessage");
switch (pmsg->message_id) {
case MSG_RTPPACKET:
case MSG_RTCPPACKET: {
@@ -1324,15 +1380,6 @@
}
}
-bool VoiceChannel::PressDTMF(int digit, bool playout) {
- int flags = DF_SEND;
- if (playout) {
- flags |= DF_PLAY;
- }
- int duration_ms = 160;
- return InsertDtmf(0, digit, duration_ms, flags);
-}
-
bool VoiceChannel::CanInsertDtmf() {
return InvokeOnWorker(Bind(&VoiceMediaChannel::CanInsertDtmf,
media_channel()));
@@ -1340,10 +1387,9 @@
bool VoiceChannel::InsertDtmf(uint32_t ssrc,
int event_code,
- int duration,
- int flags) {
+ int duration) {
return InvokeOnWorker(Bind(&VoiceChannel::InsertDtmf_w, this,
- ssrc, event_code, duration, flags));
+ ssrc, event_code, duration));
}
bool VoiceChannel::SetOutputVolume(uint32_t ssrc, double volume) {
@@ -1351,6 +1397,15 @@
media_channel(), ssrc, volume));
}
+void VoiceChannel::SetRawAudioSink(
+ uint32_t ssrc,
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+ // We need to work around Bind's lack of support for scoped_ptr and ownership
+ // passing. So we invoke to our own little routine that gets a pointer to
+ // our local variable. This is OK since we're synchronously invoking.
+ InvokeOnWorker(Bind(&SetRawAudioSink_w, media_channel(), ssrc, &sink));
+}
+
bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
return InvokeOnWorker(Bind(&VoiceMediaChannel::GetStats,
media_channel(), stats));
@@ -1440,6 +1495,7 @@
bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
+ TRACE_EVENT0("webrtc", "VoiceChannel::SetLocalContent_w");
ASSERT(worker_thread() == rtc::Thread::Current());
LOG(LS_INFO) << "Setting local voice description";
@@ -1484,6 +1540,7 @@
bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
+ TRACE_EVENT0("webrtc", "VoiceChannel::SetRemoteContent_w");
ASSERT(worker_thread() == rtc::Thread::Current());
LOG(LS_INFO) << "Setting remote voice description";
@@ -1502,7 +1559,7 @@
AudioSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(audio, &send_params);
if (audio->agc_minus_10db()) {
- send_params.options.adjust_agc_delta.Set(kAgcMinus10db);
+ send_params.options.adjust_agc_delta = rtc::Optional<int>(kAgcMinus10db);
}
if (!media_channel()->SetSendParameters(send_params)) {
SafeSetError("Failed to set remote audio description send parameters.",
@@ -1539,13 +1596,11 @@
bool VoiceChannel::InsertDtmf_w(uint32_t ssrc,
int event,
- int duration,
- int flags) {
+ int duration) {
if (!enabled()) {
return false;
}
-
- return media_channel()->InsertDtmf(ssrc, event, duration, flags);
+ return media_channel()->InsertDtmf(ssrc, event, duration);
}
void VoiceChannel::OnMessage(rtc::Message *pmsg) {
@@ -1581,9 +1636,8 @@
SignalAudioMonitor(this, info);
}
-void VoiceChannel::GetSrtpCryptoSuiteNames(
- std::vector<std::string>* ciphers) const {
- GetSupportedAudioCryptoSuites(ciphers);
+void VoiceChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
+ GetSupportedAudioCryptoSuites(crypto_suites);
}
VideoChannel::VideoChannel(rtc::Thread* thread,
@@ -1653,20 +1707,6 @@
return InvokeOnWorker(Bind(&VideoChannel::IsScreencasting_w, this));
}
-int VideoChannel::GetScreencastFps(uint32_t ssrc) {
- ScreencastDetailsData data(ssrc);
- worker_thread()->Invoke<void>(Bind(
- &VideoChannel::GetScreencastDetails_w, this, &data));
- return data.fps;
-}
-
-int VideoChannel::GetScreencastMaxPixels(uint32_t ssrc) {
- ScreencastDetailsData data(ssrc);
- worker_thread()->Invoke<void>(Bind(
- &VideoChannel::GetScreencastDetails_w, this, &data));
- return data.screencast_max_pixels;
-}
-
bool VideoChannel::SendIntraFrame() {
worker_thread()->Invoke<void>(Bind(
&VideoMediaChannel::SendIntraFrame, media_channel()));
@@ -1726,6 +1766,7 @@
bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
+ TRACE_EVENT0("webrtc", "VideoChannel::SetLocalContent_w");
ASSERT(worker_thread() == rtc::Thread::Current());
LOG(LS_INFO) << "Setting local video description";
@@ -1770,6 +1811,7 @@
bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
+ TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w");
ASSERT(worker_thread() == rtc::Thread::Current());
LOG(LS_INFO) << "Setting remote video description";
@@ -1789,7 +1831,7 @@
VideoSendParameters send_params = last_send_params_;
RtpSendParametersFromMediaDescription(video, &send_params);
if (video->conference_mode()) {
- send_params.options.conference_mode.Set(true);
+ send_params.options.conference_mode = rtc::Optional<bool>(true);
}
if (!media_channel()->SetSendParameters(send_params)) {
SafeSetError("Failed to set remote video description send parameters.",
@@ -1877,18 +1919,6 @@
return !screencast_capturers_.empty();
}
-void VideoChannel::GetScreencastDetails_w(
- ScreencastDetailsData* data) const {
- ScreencastMap::const_iterator iter = screencast_capturers_.find(data->ssrc);
- if (iter == screencast_capturers_.end()) {
- return;
- }
- VideoCapturer* capturer = iter->second;
- const VideoFormat* video_format = capturer->GetCaptureFormat();
- data->fps = VideoFormat::IntervalToFps(video_format->interval);
- data->screencast_max_pixels = capturer->screencast_max_pixels();
-}
-
void VideoChannel::OnScreencastWindowEvent_s(uint32_t ssrc,
rtc::WindowEvent we) {
ASSERT(signaling_thread() == rtc::Thread::Current());
@@ -1971,9 +2001,8 @@
return false;
}
-void VideoChannel::GetSrtpCryptoSuiteNames(
- std::vector<std::string>* ciphers) const {
- GetSupportedVideoCryptoSuites(ciphers);
+void VideoChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
+ GetSupportedVideoCryptoSuites(crypto_suites);
}
DataChannel::DataChannel(rtc::Thread* thread,
@@ -2067,6 +2096,7 @@
bool DataChannel::SetLocalContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
+ TRACE_EVENT0("webrtc", "DataChannel::SetLocalContent_w");
ASSERT(worker_thread() == rtc::Thread::Current());
LOG(LS_INFO) << "Setting local data description";
@@ -2122,6 +2152,7 @@
bool DataChannel::SetRemoteContent_w(const MediaContentDescription* content,
ContentAction action,
std::string* error_desc) {
+ TRACE_EVENT0("webrtc", "DataChannel::SetRemoteContent_w");
ASSERT(worker_thread() == rtc::Thread::Current());
const DataContentDescription* data =
@@ -2279,13 +2310,12 @@
new DataChannelReadyToSendMessageData(writable));
}
-void DataChannel::GetSrtpCryptoSuiteNames(
- std::vector<std::string>* ciphers) const {
- GetSupportedDataCryptoSuites(ciphers);
+void DataChannel::GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const {
+ GetSupportedDataCryptoSuites(crypto_suites);
}
bool DataChannel::ShouldSetupDtlsSrtp() const {
- return (data_channel_type_ == DCT_RTP);
+ return (data_channel_type_ == DCT_RTP) && BaseChannel::ShouldSetupDtlsSrtp();
}
void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
diff --git a/talk/session/media/channel.h b/talk/session/media/channel.h
index 603115c..d8fde67 100644
--- a/talk/session/media/channel.h
+++ b/talk/session/media/channel.h
@@ -38,19 +38,24 @@
#include "talk/media/base/mediaengine.h"
#include "talk/media/base/streamparams.h"
#include "talk/media/base/videocapturer.h"
-#include "webrtc/p2p/base/transportcontroller.h"
-#include "webrtc/p2p/client/socketmonitor.h"
#include "talk/session/media/audiomonitor.h"
#include "talk/session/media/bundlefilter.h"
#include "talk/session/media/mediamonitor.h"
#include "talk/session/media/mediasession.h"
#include "talk/session/media/rtcpmuxfilter.h"
#include "talk/session/media/srtpfilter.h"
+#include "webrtc/audio/audio_sink.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/network.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/base/window.h"
+#include "webrtc/p2p/base/transportcontroller.h"
+#include "webrtc/p2p/client/socketmonitor.h"
+
+namespace webrtc {
+class AudioSinkInterface;
+} // namespace webrtc
namespace cricket {
@@ -174,8 +179,11 @@
// Sets the |transport_channel_| (and |rtcp_transport_channel_|, if |rtcp_| is
// true). Gets the transport channels from |transport_controller_|.
bool SetTransport_w(const std::string& transport_name);
+
void set_transport_channel(TransportChannel* transport);
- void set_rtcp_transport_channel(TransportChannel* transport);
+ void set_rtcp_transport_channel(TransportChannel* transport,
+ bool update_writablity);
+
bool was_ever_writable() const { return was_ever_writable_; }
void set_local_content_direction(MediaContentDirection direction) {
local_content_direction_ = direction;
@@ -213,6 +221,8 @@
int flags);
void OnReadyToSend(TransportChannel* channel);
+ void OnDtlsState(TransportChannel* channel, DtlsTransportState state);
+
bool PacketIsRtcp(const TransportChannel* channel, const char* data,
size_t len);
bool SendPacket(bool rtcp,
@@ -235,8 +245,9 @@
// Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
// |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
bool SetupDtlsSrtp(bool rtcp_channel);
+ void MaybeSetupDtlsSrtp_w();
// Set the DTLS-SRTP cipher policy on this channel as appropriate.
- bool SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp);
+ bool SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp);
virtual void ChangeState() = 0;
@@ -282,9 +293,8 @@
void OnMessage(rtc::Message* pmsg) override;
// Handled in derived classes
- // Get the SRTP ciphers to use for RTP media
- virtual void GetSrtpCryptoSuiteNames(
- std::vector<std::string>* ciphers) const = 0;
+ // Get the SRTP crypto suites to use for RTP media
+ virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const = 0;
virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
const std::vector<ConnectionInfo>& infos) = 0;
@@ -356,8 +366,6 @@
// own ringing sound
sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout;
- // TODO(ronghuawu): Replace PressDTMF with InsertDtmf.
- bool PressDTMF(int digit, bool playout);
// Returns if the telephone-event has been negotiated.
bool CanInsertDtmf();
// Send and/or play a DTMF |event| according to the |flags|.
@@ -365,8 +373,11 @@
// The |ssrc| should be either 0 or a valid send stream ssrc.
// The valid value for the |event| are 0 which corresponding to DTMF
// event 0-9, *, #, A-D.
- bool InsertDtmf(uint32_t ssrc, int event_code, int duration, int flags);
+ bool InsertDtmf(uint32_t ssrc, int event_code, int duration);
bool SetOutputVolume(uint32_t ssrc, double volume);
+ void SetRawAudioSink(uint32_t ssrc,
+ rtc::scoped_ptr<webrtc::AudioSinkInterface> sink);
+
// Get statistics about the current media session.
bool GetStats(VoiceMediaInfo* stats);
@@ -402,12 +413,12 @@
ContentAction action,
std::string* error_desc);
void HandleEarlyMediaTimeout();
- bool InsertDtmf_w(uint32_t ssrc, int event, int duration, int flags);
+ bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
bool SetOutputVolume_w(uint32_t ssrc, double volume);
bool GetStats_w(VoiceMediaInfo* stats);
virtual void OnMessage(rtc::Message* pmsg);
- virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const;
+ virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
virtual void OnConnectionMonitorUpdate(
ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
virtual void OnMediaMonitorUpdate(
@@ -456,8 +467,6 @@
// True if we've added a screencast. Doesn't matter if the capturer
// has been started or not.
bool IsScreencasting();
- int GetScreencastFps(uint32_t ssrc);
- int GetScreencastMaxPixels(uint32_t ssrc);
// Get statistics about the current media session.
bool GetStats(VideoMediaInfo* stats);
@@ -476,7 +485,6 @@
private:
typedef std::map<uint32_t, VideoCapturer*> ScreencastMap;
- struct ScreencastDetailsData;
// overrides from BaseChannel
virtual void ChangeState();
@@ -493,11 +501,10 @@
bool RemoveScreencast_w(uint32_t ssrc);
void OnScreencastWindowEvent_s(uint32_t ssrc, rtc::WindowEvent we);
bool IsScreencasting_w() const;
- void GetScreencastDetails_w(ScreencastDetailsData* d) const;
bool GetStats_w(VideoMediaInfo* stats);
virtual void OnMessage(rtc::Message* pmsg);
- virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const;
+ virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
virtual void OnConnectionMonitorUpdate(
ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
virtual void OnMediaMonitorUpdate(
@@ -614,7 +621,7 @@
virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet);
virtual void OnMessage(rtc::Message* pmsg);
- virtual void GetSrtpCryptoSuiteNames(std::vector<std::string>* ciphers) const;
+ virtual void GetSrtpCryptoSuites(std::vector<int>* crypto_suites) const;
virtual void OnConnectionMonitorUpdate(
ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos);
virtual void OnMediaMonitorUpdate(
diff --git a/talk/session/media/channel_unittest.cc b/talk/session/media/channel_unittest.cc
index 1823320..6b1d66f 100644
--- a/talk/session/media/channel_unittest.cc
+++ b/talk/session/media/channel_unittest.cc
@@ -33,8 +33,8 @@
#include "talk/media/base/rtpdump.h"
#include "talk/media/base/screencastid.h"
#include "talk/media/base/testutils.h"
-#include "webrtc/p2p/base/faketransportcontroller.h"
#include "talk/session/media/channel.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
@@ -44,6 +44,7 @@
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/sslidentity.h"
#include "webrtc/base/window.h"
+#include "webrtc/p2p/base/faketransportcontroller.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(rtc::SSLStreamAdapter::feature())) { \
@@ -174,17 +175,15 @@
if (flags1 & DTLS) {
// Confirmed to work with KT_RSA and KT_ECDSA.
- transport_controller1_.SetLocalCertificate(rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT))
- .Pass()));
+ transport_controller1_.SetLocalCertificate(
+ rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+ rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT))));
}
if (flags2 & DTLS) {
// Confirmed to work with KT_RSA and KT_ECDSA.
- transport_controller2_.SetLocalCertificate(rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT))
- .Pass()));
+ transport_controller2_.SetLocalCertificate(
+ rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+ rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT))));
}
// Add stream information (SSRC) to the local content but not to the remote
@@ -1473,12 +1472,6 @@
EXPECT_TRUE(channel2_->bundle_filter()->FindPayloadType(pl_type1));
EXPECT_FALSE(channel1_->bundle_filter()->FindPayloadType(pl_type2));
EXPECT_FALSE(channel2_->bundle_filter()->FindPayloadType(pl_type2));
- // channel1 - should only have media_content2 as remote. i.e. kSsrc2
- EXPECT_TRUE(channel1_->bundle_filter()->FindStream(kSsrc2));
- EXPECT_FALSE(channel1_->bundle_filter()->FindStream(kSsrc1));
- // channel2 - should only have media_content1 as remote. i.e. kSsrc1
- EXPECT_TRUE(channel2_->bundle_filter()->FindStream(kSsrc1));
- EXPECT_FALSE(channel2_->bundle_filter()->FindStream(kSsrc2));
// Both channels can receive pl_type1 only.
EXPECT_TRUE(SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type1));
@@ -1503,8 +1496,9 @@
EXPECT_TRUE(SendCustomRtcp1(kSsrc2));
EXPECT_TRUE(SendCustomRtcp2(kSsrc1));
- EXPECT_FALSE(CheckCustomRtcp1(kSsrc1));
- EXPECT_FALSE(CheckCustomRtcp2(kSsrc2));
+ // Bundle filter shouldn't filter out any RTCP.
+ EXPECT_TRUE(CheckCustomRtcp1(kSsrc1));
+ EXPECT_TRUE(CheckCustomRtcp2(kSsrc2));
}
// Test that the media monitor can be run and gives timely callbacks.
@@ -2116,23 +2110,6 @@
Base::TestMediaMonitor();
}
-// Test that PressDTMF properly forwards to the media channel.
-TEST_F(VoiceChannelTest, TestDtmf) {
- CreateChannels(0, 0);
- EXPECT_TRUE(SendInitiate());
- EXPECT_TRUE(SendAccept());
- EXPECT_EQ(0U, media_channel1_->dtmf_info_queue().size());
-
- EXPECT_TRUE(channel1_->PressDTMF(1, true));
- EXPECT_TRUE(channel1_->PressDTMF(8, false));
-
- ASSERT_EQ(2U, media_channel1_->dtmf_info_queue().size());
- EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[0],
- 0, 1, 160, cricket::DF_PLAY | cricket::DF_SEND));
- EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[1],
- 0, 8, 160, cricket::DF_SEND));
-}
-
// Test that InsertDtmf properly forwards to the media channel.
TEST_F(VoiceChannelTest, TestInsertDtmf) {
CreateChannels(0, 0);
@@ -2140,18 +2117,17 @@
EXPECT_TRUE(SendAccept());
EXPECT_EQ(0U, media_channel1_->dtmf_info_queue().size());
- EXPECT_TRUE(channel1_->InsertDtmf(1, 3, 100, cricket::DF_SEND));
- EXPECT_TRUE(channel1_->InsertDtmf(2, 5, 110, cricket::DF_PLAY));
- EXPECT_TRUE(channel1_->InsertDtmf(3, 7, 120,
- cricket::DF_PLAY | cricket::DF_SEND));
+ EXPECT_TRUE(channel1_->InsertDtmf(1, 3, 100));
+ EXPECT_TRUE(channel1_->InsertDtmf(2, 5, 110));
+ EXPECT_TRUE(channel1_->InsertDtmf(3, 7, 120));
ASSERT_EQ(3U, media_channel1_->dtmf_info_queue().size());
EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[0],
- 1, 3, 100, cricket::DF_SEND));
+ 1, 3, 100));
EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[1],
- 2, 5, 110, cricket::DF_PLAY));
+ 2, 5, 110));
EXPECT_TRUE(CompareDtmfInfo(media_channel1_->dtmf_info_queue()[2],
- 3, 7, 120, cricket::DF_PLAY | cricket::DF_SEND));
+ 3, 7, 120));
}
TEST_F(VoiceChannelTest, TestSetContentFailure) {
@@ -2253,21 +2229,19 @@
}
TEST_F(VoiceChannelTest, SendBundleToBundle) {
- Base::SendBundleToBundle(kAudioPts, ARRAY_SIZE(kAudioPts), false, false);
+ Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, false);
}
TEST_F(VoiceChannelTest, SendBundleToBundleSecure) {
- Base::SendBundleToBundle(kAudioPts, ARRAY_SIZE(kAudioPts), false, true);
+ Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, true);
}
TEST_F(VoiceChannelTest, SendBundleToBundleWithRtcpMux) {
- Base::SendBundleToBundle(
- kAudioPts, ARRAY_SIZE(kAudioPts), true, false);
+ Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, false);
}
TEST_F(VoiceChannelTest, SendBundleToBundleWithRtcpMuxSecure) {
- Base::SendBundleToBundle(
- kAudioPts, ARRAY_SIZE(kAudioPts), true, true);
+ Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, true);
}
// VideoChannelTest
@@ -2501,21 +2475,19 @@
}
TEST_F(VideoChannelTest, SendBundleToBundle) {
- Base::SendBundleToBundle(kVideoPts, ARRAY_SIZE(kVideoPts), false, false);
+ Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), false, false);
}
TEST_F(VideoChannelTest, SendBundleToBundleSecure) {
- Base::SendBundleToBundle(kVideoPts, ARRAY_SIZE(kVideoPts), false, true);
+ Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), false, true);
}
TEST_F(VideoChannelTest, SendBundleToBundleWithRtcpMux) {
- Base::SendBundleToBundle(
- kVideoPts, ARRAY_SIZE(kVideoPts), true, false);
+ Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), true, false);
}
TEST_F(VideoChannelTest, SendBundleToBundleWithRtcpMuxSecure) {
- Base::SendBundleToBundle(
- kVideoPts, ARRAY_SIZE(kVideoPts), true, true);
+ Base::SendBundleToBundle(kVideoPts, arraysize(kVideoPts), true, true);
}
TEST_F(VideoChannelTest, TestSrtpError) {
diff --git a/talk/session/media/channelmanager.cc b/talk/session/media/channelmanager.cc
index e7e1cd4..e7a4b8b 100644
--- a/talk/session/media/channelmanager.cc
+++ b/talk/session/media/channelmanager.cc
@@ -49,6 +49,7 @@
#include "webrtc/base/sigslotrepeater.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
+#include "webrtc/base/trace_event.h"
namespace cricket {
@@ -101,8 +102,6 @@
initialized_ = false;
main_thread_ = rtc::Thread::Current();
worker_thread_ = worker_thread;
- // Get the default audio options from the media engine.
- audio_options_ = media_engine_->GetAudioOptions();
audio_output_volume_ = kNotSetOutputVolume;
local_renderer_ = NULL;
capturing_ = false;
@@ -156,7 +155,7 @@
void ChannelManager::GetSupportedAudioRtpHeaderExtensions(
RtpHeaderExtensions* ext) const {
- *ext = media_engine_->audio_rtp_header_extensions();
+ *ext = media_engine_->GetAudioCapabilities().header_extensions;
}
void ChannelManager::GetSupportedVideoCodecs(
@@ -175,7 +174,7 @@
void ChannelManager::GetSupportedVideoRtpHeaderExtensions(
RtpHeaderExtensions* ext) const {
- *ext = media_engine_->video_rtp_header_extensions();
+ *ext = media_engine_->GetVideoCapabilities().header_extensions;
}
void ChannelManager::GetSupportedDataCodecs(
@@ -205,11 +204,6 @@
return false;
}
- if (!SetAudioOptions(audio_options_)) {
- LOG(LS_WARNING) << "Failed to SetAudioOptions with options: "
- << audio_options_.ToString();
- }
-
// If audio_output_volume_ has been set via SetOutputVolume(), set the
// audio output volume of the engine.
if (kNotSetOutputVolume != audio_output_volume_ &&
@@ -218,11 +212,6 @@
<< audio_output_volume_;
}
- // Now apply the default video codec that has been set earlier.
- if (default_video_encoder_config_.max_codec.id != 0) {
- SetDefaultVideoEncoderConfig(default_video_encoder_config_);
- }
-
return initialized_;
}
@@ -295,6 +284,7 @@
}
void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) {
+ TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel");
if (voice_channel) {
worker_thread_->Invoke<void>(
Bind(&ChannelManager::DestroyVoiceChannel_w, this, voice_channel));
@@ -302,6 +292,7 @@
}
void ChannelManager::DestroyVoiceChannel_w(VoiceChannel* voice_channel) {
+ TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel_w");
// Destroy voice channel.
ASSERT(initialized_);
ASSERT(worker_thread_ == rtc::Thread::Current());
@@ -351,6 +342,7 @@
}
void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) {
+ TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel");
if (video_channel) {
worker_thread_->Invoke<void>(
Bind(&ChannelManager::DestroyVideoChannel_w, this, video_channel));
@@ -358,6 +350,7 @@
}
void ChannelManager::DestroyVideoChannel_w(VideoChannel* video_channel) {
+ TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel_w");
// Destroy video channel.
ASSERT(initialized_);
ASSERT(worker_thread_ == rtc::Thread::Current());
@@ -408,6 +401,7 @@
}
void ChannelManager::DestroyDataChannel(DataChannel* data_channel) {
+ TRACE_EVENT0("webrtc", "ChannelManager::DestroyDataChannel");
if (data_channel) {
worker_thread_->Invoke<void>(
Bind(&ChannelManager::DestroyDataChannel_w, this, data_channel));
@@ -415,6 +409,7 @@
}
void ChannelManager::DestroyDataChannel_w(DataChannel* data_channel) {
+ TRACE_EVENT0("webrtc", "ChannelManager::DestroyDataChannel_w");
// Destroy data channel.
ASSERT(initialized_);
DataChannels::iterator it = std::find(data_channels_.begin(),
@@ -427,43 +422,6 @@
delete data_channel;
}
-bool ChannelManager::SetAudioOptions(const AudioOptions& options) {
- // "Get device ids from DeviceManager" - these are the defaults returned.
- Device in_dev("", -1);
- Device out_dev("", -1);
-
- // If we're initialized, pass the settings to the media engine.
- bool ret = true;
- if (initialized_) {
- ret = worker_thread_->Invoke<bool>(
- Bind(&ChannelManager::SetAudioOptions_w, this,
- options, &in_dev, &out_dev));
- }
-
- // If all worked well, save the values for use in GetAudioOptions.
- if (ret) {
- audio_options_ = options;
- }
- return ret;
-}
-
-bool ChannelManager::SetAudioOptions_w(
- const AudioOptions& options,
- const Device* in_dev, const Device* out_dev) {
- ASSERT(worker_thread_ == rtc::Thread::Current());
- ASSERT(initialized_);
-
- // Set audio options
- bool ret = media_engine_->SetAudioOptions(options);
-
- // Set the audio devices
- if (ret) {
- ret = media_engine_->SetSoundDevices(in_dev, out_dev);
- }
-
- return ret;
-}
-
bool ChannelManager::GetOutputVolume(int* level) {
if (!initialized_) {
return false;
@@ -487,39 +445,6 @@
return ret;
}
-bool ChannelManager::SetDefaultVideoEncoderConfig(const VideoEncoderConfig& c) {
- bool ret = true;
- if (initialized_) {
- ret = worker_thread_->Invoke<bool>(
- Bind(&MediaEngineInterface::SetDefaultVideoEncoderConfig,
- media_engine_.get(), c));
- }
- if (ret) {
- default_video_encoder_config_ = c;
- }
- return ret;
-}
-
-void ChannelManager::SetVoiceLogging(int level, const char* filter) {
- if (initialized_) {
- worker_thread_->Invoke<void>(
- Bind(&MediaEngineInterface::SetVoiceLogging,
- media_engine_.get(), level, filter));
- } else {
- media_engine_->SetVoiceLogging(level, filter);
- }
-}
-
-void ChannelManager::SetVideoLogging(int level, const char* filter) {
- if (initialized_) {
- worker_thread_->Invoke<void>(
- Bind(&MediaEngineInterface::SetVideoLogging,
- media_engine_.get(), level, filter));
- } else {
- media_engine_->SetVideoLogging(level, filter);
- }
-}
-
std::vector<cricket::VideoFormat> ChannelManager::GetSupportedFormats(
VideoCapturer* capturer) const {
ASSERT(capturer != NULL);
diff --git a/talk/session/media/channelmanager.h b/talk/session/media/channelmanager.h
index 6312e61..2bc516b 100644
--- a/talk/session/media/channelmanager.h
+++ b/talk/session/media/channelmanager.h
@@ -129,7 +129,6 @@
bool GetOutputVolume(int* level);
bool SetOutputVolume(int level);
- bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config);
// RTX will be enabled/disabled in engines that support it. The supporting
// engines will start offering an RTX codec. Must be called before Init().
bool SetVideoRtxEnabled(bool enable);
@@ -137,10 +136,6 @@
// Starts/stops the local microphone and enables polling of the input level.
bool capturing() const { return capturing_; }
- // Configures the logging output of the mediaengine(s).
- void SetVoiceLogging(int level, const char* filter);
- void SetVideoLogging(int level, const char* filter);
-
// Gets capturer's supported formats in a thread safe manner
std::vector<cricket::VideoFormat> GetSupportedFormats(
VideoCapturer* capturer) const;
@@ -181,11 +176,6 @@
sigslot::signal2<VideoCapturer*, CaptureState> SignalVideoCaptureStateChange;
- protected:
- // Adds non-transient parameters which can only be changed through the
- // options store.
- bool SetAudioOptions(const AudioOptions& options);
-
private:
typedef std::vector<VoiceChannel*> VoiceChannels;
typedef std::vector<VideoChannel*> VideoChannels;
@@ -217,8 +207,6 @@
bool rtcp,
DataChannelType data_channel_type);
void DestroyDataChannel_w(DataChannel* data_channel);
- bool SetAudioOptions_w(const AudioOptions& options,
- const Device* in_dev, const Device* out_dev);
void OnVideoCaptureStateChange(VideoCapturer* capturer,
CaptureState result);
void GetSupportedFormats_w(
@@ -238,9 +226,7 @@
VideoChannels video_channels_;
DataChannels data_channels_;
- AudioOptions audio_options_;
int audio_output_volume_;
- VideoEncoderConfig default_video_encoder_config_;
VideoRenderer* local_renderer_;
bool enable_rtx_;
diff --git a/talk/session/media/channelmanager_unittest.cc b/talk/session/media/channelmanager_unittest.cc
index fa6aa2c..4740f0f 100644
--- a/talk/session/media/channelmanager_unittest.cc
+++ b/talk/session/media/channelmanager_unittest.cc
@@ -183,38 +183,6 @@
cm_->Terminate();
}
-// Test that SetDefaultVideoCodec passes through the right values.
-TEST_F(ChannelManagerTest, SetDefaultVideoEncoderConfig) {
- cricket::VideoCodec codec(96, "G264", 1280, 720, 60, 0);
- cricket::VideoEncoderConfig config(codec, 1, 2);
- EXPECT_TRUE(cm_->Init());
- EXPECT_TRUE(cm_->SetDefaultVideoEncoderConfig(config));
- EXPECT_EQ(config, fme_->default_video_encoder_config());
-}
-
-struct GetCapturerFrameSize : public sigslot::has_slots<> {
- void OnVideoFrame(VideoCapturer* capturer, const VideoFrame* frame) {
- width = frame->GetWidth();
- height = frame->GetHeight();
- }
- GetCapturerFrameSize(VideoCapturer* capturer) : width(0), height(0) {
- capturer->SignalVideoFrame.connect(this,
- &GetCapturerFrameSize::OnVideoFrame);
- static_cast<FakeVideoCapturer*>(capturer)->CaptureFrame();
- }
- size_t width;
- size_t height;
-};
-
-// Test that SetDefaultVideoCodec passes through the right values.
-TEST_F(ChannelManagerTest, SetDefaultVideoCodecBeforeInit) {
- cricket::VideoCodec codec(96, "G264", 1280, 720, 60, 0);
- cricket::VideoEncoderConfig config(codec, 1, 2);
- EXPECT_TRUE(cm_->SetDefaultVideoEncoderConfig(config));
- EXPECT_TRUE(cm_->Init());
- EXPECT_EQ(config, fme_->default_video_encoder_config());
-}
-
TEST_F(ChannelManagerTest, GetSetOutputVolumeBeforeInit) {
int level;
// Before init, SetOutputVolume() remembers the volume but does not change the
@@ -250,33 +218,6 @@
EXPECT_EQ(60, level);
}
-// Test that logging options set before Init are applied properly,
-// and retained even after Init.
-TEST_F(ChannelManagerTest, SetLoggingBeforeInit) {
- cm_->SetVoiceLogging(rtc::LS_INFO, "test-voice");
- cm_->SetVideoLogging(rtc::LS_VERBOSE, "test-video");
- EXPECT_EQ(rtc::LS_INFO, fme_->voice_loglevel());
- EXPECT_STREQ("test-voice", fme_->voice_logfilter().c_str());
- EXPECT_EQ(rtc::LS_VERBOSE, fme_->video_loglevel());
- EXPECT_STREQ("test-video", fme_->video_logfilter().c_str());
- EXPECT_TRUE(cm_->Init());
- EXPECT_EQ(rtc::LS_INFO, fme_->voice_loglevel());
- EXPECT_STREQ("test-voice", fme_->voice_logfilter().c_str());
- EXPECT_EQ(rtc::LS_VERBOSE, fme_->video_loglevel());
- EXPECT_STREQ("test-video", fme_->video_logfilter().c_str());
-}
-
-// Test that logging options set after Init are applied properly.
-TEST_F(ChannelManagerTest, SetLogging) {
- EXPECT_TRUE(cm_->Init());
- cm_->SetVoiceLogging(rtc::LS_INFO, "test-voice");
- cm_->SetVideoLogging(rtc::LS_VERBOSE, "test-video");
- EXPECT_EQ(rtc::LS_INFO, fme_->voice_loglevel());
- EXPECT_STREQ("test-voice", fme_->voice_logfilter().c_str());
- EXPECT_EQ(rtc::LS_VERBOSE, fme_->video_loglevel());
- EXPECT_STREQ("test-video", fme_->video_logfilter().c_str());
-}
-
TEST_F(ChannelManagerTest, SetVideoRtxEnabled) {
std::vector<VideoCodec> codecs;
const VideoCodec rtx_codec(96, "rtx", 0, 0, 0, 0);
diff --git a/talk/session/media/mediasession.cc b/talk/session/media/mediasession.cc
index 7413026..24f01b4 100644
--- a/talk/session/media/mediasession.cc
+++ b/talk/session/media/mediasession.cc
@@ -50,6 +50,17 @@
namespace {
const char kInline[] = "inline:";
+
+void GetSupportedCryptoSuiteNames(void (*func)(std::vector<int>*),
+ std::vector<std::string>* names) {
+#ifdef HAVE_SRTP
+ std::vector<int> crypto_suites;
+ func(&crypto_suites);
+ for (const auto crypto : crypto_suites) {
+ names->push_back(rtc::SrtpCryptoSuiteToName(crypto));
+ }
+#endif
+}
}
namespace cricket {
@@ -152,30 +163,50 @@
}
// For audio, HMAC 32 is prefered because of the low overhead.
-void GetSupportedAudioCryptoSuites(
- std::vector<std::string>* crypto_suites) {
+void GetSupportedAudioCryptoSuites(std::vector<int>* crypto_suites) {
#ifdef HAVE_SRTP
- crypto_suites->push_back(rtc::CS_AES_CM_128_HMAC_SHA1_32);
- crypto_suites->push_back(rtc::CS_AES_CM_128_HMAC_SHA1_80);
+ crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32);
+ crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
#endif
}
-void GetSupportedVideoCryptoSuites(
- std::vector<std::string>* crypto_suites) {
- GetDefaultSrtpCryptoSuiteNames(crypto_suites);
+void GetSupportedAudioCryptoSuiteNames(
+ std::vector<std::string>* crypto_suite_names) {
+ GetSupportedCryptoSuiteNames(GetSupportedAudioCryptoSuites,
+ crypto_suite_names);
}
-void GetSupportedDataCryptoSuites(
- std::vector<std::string>* crypto_suites) {
- GetDefaultSrtpCryptoSuiteNames(crypto_suites);
+void GetSupportedVideoCryptoSuites(std::vector<int>* crypto_suites) {
+ GetDefaultSrtpCryptoSuites(crypto_suites);
}
-void GetDefaultSrtpCryptoSuiteNames(std::vector<std::string>* crypto_suites) {
+void GetSupportedVideoCryptoSuiteNames(
+ std::vector<std::string>* crypto_suite_names) {
+ GetSupportedCryptoSuiteNames(GetSupportedVideoCryptoSuites,
+ crypto_suite_names);
+}
+
+void GetSupportedDataCryptoSuites(std::vector<int>* crypto_suites) {
+ GetDefaultSrtpCryptoSuites(crypto_suites);
+}
+
+void GetSupportedDataCryptoSuiteNames(
+ std::vector<std::string>* crypto_suite_names) {
+ GetSupportedCryptoSuiteNames(GetSupportedDataCryptoSuites,
+ crypto_suite_names);
+}
+
+void GetDefaultSrtpCryptoSuites(std::vector<int>* crypto_suites) {
#ifdef HAVE_SRTP
- crypto_suites->push_back(rtc::CS_AES_CM_128_HMAC_SHA1_80);
+ crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
#endif
}
+void GetDefaultSrtpCryptoSuiteNames(
+ std::vector<std::string>* crypto_suite_names) {
+ GetSupportedCryptoSuiteNames(GetDefaultSrtpCryptoSuites, crypto_suite_names);
+}
+
// For video support only 80-bit SHA1 HMAC. For audio 32-bit HMAC is
// tolerated unless bundle is enabled because it is low overhead. Pick the
// crypto in the list that is supported.
@@ -518,8 +549,8 @@
// Updates the transport infos of the |sdesc| according to the given
// |bundle_group|. The transport infos of the content names within the
-// |bundle_group| should be updated to use the ufrag and pwd of the first
-// content within the |bundle_group|.
+// |bundle_group| should be updated to use the ufrag, pwd and DTLS role of the
+// first content within the |bundle_group|.
static bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group,
SessionDescription* sdesc) {
// The bundle should not be empty.
@@ -540,6 +571,8 @@
selected_transport_info->description.ice_ufrag;
const std::string& selected_pwd =
selected_transport_info->description.ice_pwd;
+ ConnectionRole selected_connection_role =
+ selected_transport_info->description.connection_role;
for (TransportInfos::iterator it =
sdesc->transport_infos().begin();
it != sdesc->transport_infos().end(); ++it) {
@@ -547,6 +580,7 @@
it->content_name != selected_content_name) {
it->description.ice_ufrag = selected_ufrag;
it->description.ice_pwd = selected_pwd;
+ it->description.connection_role = selected_connection_role;
}
}
return true;
@@ -602,6 +636,11 @@
target_cryptos->end());
}
+static bool IsRtpProtocol(const std::string& protocol) {
+ return protocol.empty() ||
+ (protocol.find(cricket::kMediaProtocolRtpPrefix) != std::string::npos);
+}
+
static bool IsRtpContent(SessionDescription* sdesc,
const std::string& content_name) {
bool is_rtp = false;
@@ -612,9 +651,7 @@
if (!media_desc) {
return false;
}
- is_rtp = media_desc->protocol().empty() ||
- (media_desc->protocol().find(cricket::kMediaProtocolRtpPrefix) !=
- std::string::npos);
+ is_rtp = IsRtpProtocol(media_desc->protocol());
}
return is_rtp;
}
@@ -726,6 +763,11 @@
offer->set_crypto_required(CT_SDES);
}
offer->set_rtcp_mux(options.rtcp_mux_enabled);
+ // TODO(deadbeef): Once we're sure this works correctly, enable it in
+ // CreateOffer.
+ // if (offer->type() == cricket::MEDIA_TYPE_VIDEO) {
+ // offer->set_rtcp_reduced_size(true);
+ // }
offer->set_multistream(options.is_muc);
offer->set_rtp_header_extensions(rtp_extensions);
@@ -1004,6 +1046,11 @@
answer->set_rtp_header_extensions(negotiated_rtp_extensions);
answer->set_rtcp_mux(options.rtcp_mux_enabled && offer->rtcp_mux());
+ // TODO(deadbeef): Once we're sure this works correctly, enable it in
+ // CreateAnswer.
+ // if (answer->type() == cricket::MEDIA_TYPE_VIDEO) {
+ // answer->set_rtcp_reduced_size(offer->rtcp_reduced_size());
+ // }
if (sdes_policy != SEC_DISABLED) {
CryptoParams crypto;
@@ -1036,12 +1083,16 @@
answer->set_direction(MD_RECVONLY);
break;
case MD_RECVONLY:
- answer->set_direction(answer->streams().empty() ? MD_INACTIVE
- : MD_SENDONLY);
+ answer->set_direction(IsRtpProtocol(answer->protocol()) &&
+ answer->streams().empty()
+ ? MD_INACTIVE
+ : MD_SENDONLY);
break;
case MD_SENDRECV:
- answer->set_direction(answer->streams().empty() ? MD_RECVONLY
- : MD_SENDRECV);
+ answer->set_direction(IsRtpProtocol(answer->protocol()) &&
+ answer->streams().empty()
+ ? MD_RECVONLY
+ : MD_SENDRECV);
break;
default:
RTC_DCHECK(false && "MediaContentDescription has unexpected direction.");
@@ -1508,13 +1559,18 @@
const AudioCodecs& audio_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc) const {
+ const ContentInfo* current_audio_content =
+ GetFirstAudioContent(current_description);
+ std::string content_name =
+ current_audio_content ? current_audio_content->name : CN_AUDIO;
+
cricket::SecurePolicy sdes_policy =
- IsDtlsActive(CN_AUDIO, current_description) ?
- cricket::SEC_DISABLED : secure();
+ IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
+ : secure();
scoped_ptr<AudioContentDescription> audio(new AudioContentDescription());
std::vector<std::string> crypto_suites;
- GetSupportedAudioCryptoSuites(&crypto_suites);
+ GetSupportedAudioCryptoSuiteNames(&crypto_suites);
if (!CreateMediaContentOffer(
options,
audio_codecs,
@@ -1546,8 +1602,8 @@
}
}
- desc->AddContent(CN_AUDIO, NS_JINGLE_RTP, audio.release());
- if (!AddTransportOffer(CN_AUDIO, options.transport_options,
+ desc->AddContent(content_name, NS_JINGLE_RTP, audio.release());
+ if (!AddTransportOffer(content_name, options.audio_transport_options,
current_description, desc)) {
return false;
}
@@ -1562,13 +1618,18 @@
const VideoCodecs& video_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc) const {
+ const ContentInfo* current_video_content =
+ GetFirstVideoContent(current_description);
+ std::string content_name =
+ current_video_content ? current_video_content->name : CN_VIDEO;
+
cricket::SecurePolicy sdes_policy =
- IsDtlsActive(CN_VIDEO, current_description) ?
- cricket::SEC_DISABLED : secure();
+ IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
+ : secure();
scoped_ptr<VideoContentDescription> video(new VideoContentDescription());
std::vector<std::string> crypto_suites;
- GetSupportedVideoCryptoSuites(&crypto_suites);
+ GetSupportedVideoCryptoSuiteNames(&crypto_suites);
if (!CreateMediaContentOffer(
options,
video_codecs,
@@ -1601,8 +1662,8 @@
}
}
- desc->AddContent(CN_VIDEO, NS_JINGLE_RTP, video.release());
- if (!AddTransportOffer(CN_VIDEO, options.transport_options,
+ desc->AddContent(content_name, NS_JINGLE_RTP, video.release());
+ if (!AddTransportOffer(content_name, options.video_transport_options,
current_description, desc)) {
return false;
}
@@ -1623,9 +1684,14 @@
FilterDataCodecs(data_codecs, is_sctp);
+ const ContentInfo* current_data_content =
+ GetFirstDataContent(current_description);
+ std::string content_name =
+ current_data_content ? current_data_content->name : CN_DATA;
+
cricket::SecurePolicy sdes_policy =
- IsDtlsActive(CN_DATA, current_description) ?
- cricket::SEC_DISABLED : secure();
+ IsDtlsActive(content_name, current_description) ? cricket::SEC_DISABLED
+ : secure();
std::vector<std::string> crypto_suites;
if (is_sctp) {
// SDES doesn't make sense for SCTP, so we disable it, and we only
@@ -1638,7 +1704,7 @@
data->set_protocol(
secure_transport ? kMediaProtocolDtlsSctp : kMediaProtocolSctp);
} else {
- GetSupportedDataCryptoSuites(&crypto_suites);
+ GetSupportedDataCryptoSuiteNames(&crypto_suites);
}
if (!CreateMediaContentOffer(
@@ -1655,13 +1721,13 @@
}
if (is_sctp) {
- desc->AddContent(CN_DATA, NS_JINGLE_DRAFT_SCTP, data.release());
+ desc->AddContent(content_name, NS_JINGLE_DRAFT_SCTP, data.release());
} else {
data->set_bandwidth(options.data_bandwidth);
SetMediaProtocol(secure_transport, data.get());
- desc->AddContent(CN_DATA, NS_JINGLE_RTP, data.release());
+ desc->AddContent(content_name, NS_JINGLE_RTP, data.release());
}
- if (!AddTransportOffer(CN_DATA, options.transport_options,
+ if (!AddTransportOffer(content_name, options.data_transport_options,
current_description, desc)) {
return false;
}
@@ -1676,10 +1742,9 @@
SessionDescription* answer) const {
const ContentInfo* audio_content = GetFirstAudioContent(offer);
- scoped_ptr<TransportDescription> audio_transport(
- CreateTransportAnswer(audio_content->name, offer,
- options.transport_options,
- current_description));
+ scoped_ptr<TransportDescription> audio_transport(CreateTransportAnswer(
+ audio_content->name, offer, options.audio_transport_options,
+ current_description));
if (!audio_transport) {
return false;
}
@@ -1735,10 +1800,9 @@
StreamParamsVec* current_streams,
SessionDescription* answer) const {
const ContentInfo* video_content = GetFirstVideoContent(offer);
- scoped_ptr<TransportDescription> video_transport(
- CreateTransportAnswer(video_content->name, offer,
- options.transport_options,
- current_description));
+ scoped_ptr<TransportDescription> video_transport(CreateTransportAnswer(
+ video_content->name, offer, options.video_transport_options,
+ current_description));
if (!video_transport) {
return false;
}
@@ -1791,10 +1855,9 @@
StreamParamsVec* current_streams,
SessionDescription* answer) const {
const ContentInfo* data_content = GetFirstDataContent(offer);
- scoped_ptr<TransportDescription> data_transport(
- CreateTransportAnswer(data_content->name, offer,
- options.transport_options,
- current_description));
+ scoped_ptr<TransportDescription> data_transport(CreateTransportAnswer(
+ data_content->name, offer, options.data_transport_options,
+ current_description));
if (!data_transport) {
return false;
}
diff --git a/talk/session/media/mediasession.h b/talk/session/media/mediasession.h
index e92628e..1540274 100644
--- a/talk/session/media/mediasession.h
+++ b/talk/session/media/mediasession.h
@@ -134,6 +134,10 @@
bool HasSendMediaStream(MediaType type) const;
+ // TODO(deadbeef): Put all the audio/video/data-specific options into a map
+ // structure (content name -> options).
+ // MediaSessionDescriptionFactory assumes there will never be more than one
+ // audio/video/data content, but this will change with unified plan.
bool recv_audio;
bool recv_video;
DataChannelType data_channel_type;
@@ -144,7 +148,9 @@
// bps. -1 == auto.
int video_bandwidth;
int data_bandwidth;
- TransportOptions transport_options;
+ TransportOptions audio_transport_options;
+ TransportOptions video_transport_options;
+ TransportOptions data_transport_options;
struct Stream {
Stream(MediaType type,
@@ -167,17 +173,7 @@
// "content" (as used in XEP-0166) descriptions for voice and video.
class MediaContentDescription : public ContentDescription {
public:
- MediaContentDescription()
- : rtcp_mux_(false),
- bandwidth_(kAutoBandwidth),
- crypto_required_(CT_NONE),
- rtp_header_extensions_set_(false),
- multistream_(false),
- conference_mode_(false),
- partial_(false),
- buffered_mode_latency_(kBufferedModeDisabled),
- direction_(MD_SENDRECV) {
- }
+ MediaContentDescription() {}
virtual MediaType type() const = 0;
virtual bool has_codecs() const = 0;
@@ -195,6 +191,11 @@
bool rtcp_mux() const { return rtcp_mux_; }
void set_rtcp_mux(bool mux) { rtcp_mux_ = mux; }
+ bool rtcp_reduced_size() const { return rtcp_reduced_size_; }
+ void set_rtcp_reduced_size(bool reduced_size) {
+ rtcp_reduced_size_ = reduced_size;
+ }
+
int bandwidth() const { return bandwidth_; }
void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
@@ -291,19 +292,20 @@
int buffered_mode_latency() const { return buffered_mode_latency_; }
protected:
- bool rtcp_mux_;
- int bandwidth_;
+ bool rtcp_mux_ = false;
+ bool rtcp_reduced_size_ = false;
+ int bandwidth_ = kAutoBandwidth;
std::string protocol_;
std::vector<CryptoParams> cryptos_;
- CryptoType crypto_required_;
+ CryptoType crypto_required_ = CT_NONE;
std::vector<RtpHeaderExtension> rtp_header_extensions_;
- bool rtp_header_extensions_set_;
- bool multistream_;
+ bool rtp_header_extensions_set_ = false;
+ bool multistream_ = false;
StreamParamsVec streams_;
- bool conference_mode_;
- bool partial_;
- int buffered_mode_latency_;
- MediaContentDirection direction_;
+ bool conference_mode_ = false;
+ bool partial_ = false;
+ int buffered_mode_latency_ = kBufferedModeDisabled;
+ MediaContentDirection direction_ = MD_SENDRECV;
};
template <class C>
@@ -547,10 +549,19 @@
const DataContentDescription* GetFirstDataContentDescription(
const SessionDescription* sdesc);
-void GetSupportedAudioCryptoSuites(std::vector<std::string>* crypto_suites);
-void GetSupportedVideoCryptoSuites(std::vector<std::string>* crypto_suites);
-void GetSupportedDataCryptoSuites(std::vector<std::string>* crypto_suites);
-void GetDefaultSrtpCryptoSuiteNames(std::vector<std::string>* crypto_suites);
+void GetSupportedAudioCryptoSuites(std::vector<int>* crypto_suites);
+void GetSupportedVideoCryptoSuites(std::vector<int>* crypto_suites);
+void GetSupportedDataCryptoSuites(std::vector<int>* crypto_suites);
+void GetDefaultSrtpCryptoSuites(std::vector<int>* crypto_suites);
+void GetSupportedAudioCryptoSuiteNames(
+ std::vector<std::string>* crypto_suite_names);
+void GetSupportedVideoCryptoSuiteNames(
+ std::vector<std::string>* crypto_suite_names);
+void GetSupportedDataCryptoSuiteNames(
+ std::vector<std::string>* crypto_suite_names);
+void GetDefaultSrtpCryptoSuiteNames(
+ std::vector<std::string>* crypto_suite_names);
+
} // namespace cricket
#endif // TALK_SESSION_MEDIA_MEDIASESSION_H_
diff --git a/talk/session/media/mediasession_unittest.cc b/talk/session/media/mediasession_unittest.cc
index 72aefc8..20b72e9 100644
--- a/talk/session/media/mediasession_unittest.cc
+++ b/talk/session/media/mediasession_unittest.cc
@@ -69,6 +69,9 @@
using cricket::AudioContentDescription;
using cricket::VideoContentDescription;
using cricket::DataContentDescription;
+using cricket::GetFirstAudioContent;
+using cricket::GetFirstVideoContent;
+using cricket::GetFirstDataContent;
using cricket::GetFirstAudioContentDescription;
using cricket::GetFirstVideoContentDescription;
using cricket::GetFirstDataContentDescription;
@@ -235,11 +238,9 @@
f2_.set_video_codecs(MAKE_VECTOR(kVideoCodecs2));
f2_.set_data_codecs(MAKE_VECTOR(kDataCodecs2));
tdf1_.set_certificate(rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- new rtc::FakeSSLIdentity("id1")).Pass()));
+ rtc::scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id1"))));
tdf2_.set_certificate(rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- new rtc::FakeSSLIdentity("id2")).Pass()));
+ rtc::scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id2"))));
}
// Create a video StreamParamsVec object with:
@@ -607,6 +608,7 @@
ASSERT_CRYPTO(dcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), dcd->protocol());
}
+
// Create a RTP data offer, and ensure it matches what we expect.
TEST_F(MediaSessionDescriptionFactoryTest, TestCreateRtpDataOffer) {
MediaSessionOptions opts;
@@ -2313,3 +2315,30 @@
audio_content = answer->GetContentByName("audio");
EXPECT_TRUE(VerifyNoCNCodecs(audio_content));
}
+
+// Test that the content name ("mid" in SDP) is unchanged when creating a
+// new offer.
+TEST_F(MediaSessionDescriptionFactoryTest,
+ TestContentNameNotChangedInSubsequentOffers) {
+ MediaSessionOptions opts;
+ opts.recv_audio = true;
+ opts.recv_video = true;
+ opts.data_channel_type = cricket::DCT_SCTP;
+ // Create offer and modify the default content names.
+ rtc::scoped_ptr<SessionDescription> offer(f1_.CreateOffer(opts, nullptr));
+ for (ContentInfo& content : offer->contents()) {
+ content.name.append("_modified");
+ }
+
+ rtc::scoped_ptr<SessionDescription> updated_offer(
+ f1_.CreateOffer(opts, offer.get()));
+ const ContentInfo* audio_content = GetFirstAudioContent(updated_offer.get());
+ const ContentInfo* video_content = GetFirstVideoContent(updated_offer.get());
+ const ContentInfo* data_content = GetFirstDataContent(updated_offer.get());
+ ASSERT_TRUE(audio_content != nullptr);
+ ASSERT_TRUE(video_content != nullptr);
+ ASSERT_TRUE(data_content != nullptr);
+ EXPECT_EQ("audio_modified", audio_content->name);
+ EXPECT_EQ("video_modified", video_content->name);
+ EXPECT_EQ("data_modified", data_content->name);
+}
diff --git a/talk/session/media/srtpfilter.cc b/talk/session/media/srtpfilter.cc
index 079ddfb..a200a3c 100644
--- a/talk/session/media/srtpfilter.cc
+++ b/talk/session/media/srtpfilter.cc
@@ -146,10 +146,10 @@
return DoSetAnswer(answer_params, source, false);
}
-bool SrtpFilter::SetRtpParams(const std::string& send_cs,
+bool SrtpFilter::SetRtpParams(int send_cs,
const uint8_t* send_key,
int send_key_len,
- const std::string& recv_cs,
+ int recv_cs,
const uint8_t* recv_key,
int recv_key_len) {
if (IsActive()) {
@@ -179,10 +179,10 @@
// SrtpSession.
// - In the muxed case, they are keyed with the same keys, so
// this function is not needed
-bool SrtpFilter::SetRtcpParams(const std::string& send_cs,
+bool SrtpFilter::SetRtcpParams(int send_cs,
const uint8_t* send_key,
int send_key_len,
- const std::string& recv_cs,
+ int recv_cs,
const uint8_t* recv_key,
int recv_key_len) {
// This can only be called once, but can be safely called after
@@ -428,10 +428,12 @@
ParseKeyParams(recv_params.key_params, recv_key, sizeof(recv_key)));
if (ret) {
CreateSrtpSessions();
- ret = (send_session_->SetSend(send_params.cipher_suite,
- send_key, sizeof(send_key)) &&
- recv_session_->SetRecv(recv_params.cipher_suite,
- recv_key, sizeof(recv_key)));
+ ret = (send_session_->SetSend(
+ rtc::SrtpCryptoSuiteFromName(send_params.cipher_suite), send_key,
+ sizeof(send_key)) &&
+ recv_session_->SetRecv(
+ rtc::SrtpCryptoSuiteFromName(recv_params.cipher_suite), recv_key,
+ sizeof(recv_key)));
}
if (ret) {
LOG(LS_INFO) << "SRTP activated with negotiated parameters:"
@@ -448,6 +450,10 @@
bool SrtpFilter::ResetParams() {
offer_params_.clear();
state_ = ST_INIT;
+ send_session_ = nullptr;
+ recv_session_ = nullptr;
+ send_rtcp_session_ = nullptr;
+ recv_rtcp_session_ = nullptr;
LOG(LS_INFO) << "SRTP reset to init state";
return true;
}
@@ -507,11 +513,11 @@
}
}
-bool SrtpSession::SetSend(const std::string& cs, const uint8_t* key, int len) {
+bool SrtpSession::SetSend(int cs, const uint8_t* key, int len) {
return SetKey(ssrc_any_outbound, cs, key, len);
}
-bool SrtpSession::SetRecv(const std::string& cs, const uint8_t* key, int len) {
+bool SrtpSession::SetRecv(int cs, const uint8_t* key, int len) {
return SetKey(ssrc_any_inbound, cs, key, len);
}
@@ -658,10 +664,7 @@
srtp_stat_->set_signal_silent_time(signal_silent_time_in_ms);
}
-bool SrtpSession::SetKey(int type,
- const std::string& cs,
- const uint8_t* key,
- int len) {
+bool SrtpSession::SetKey(int type, int cs, const uint8_t* key, int len) {
if (session_) {
LOG(LS_ERROR) << "Failed to create SRTP session: "
<< "SRTP session already created";
@@ -675,15 +678,15 @@
srtp_policy_t policy;
memset(&policy, 0, sizeof(policy));
- if (cs == rtc::CS_AES_CM_128_HMAC_SHA1_80) {
+ if (cs == rtc::SRTP_AES128_CM_SHA1_80) {
crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtp);
crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp);
- } else if (cs == rtc::CS_AES_CM_128_HMAC_SHA1_32) {
+ } else if (cs == rtc::SRTP_AES128_CM_SHA1_32) {
crypto_policy_set_aes_cm_128_hmac_sha1_32(&policy.rtp); // rtp is 32,
crypto_policy_set_aes_cm_128_hmac_sha1_80(&policy.rtcp); // rtcp still 80
} else {
LOG(LS_WARNING) << "Failed to create SRTP session: unsupported"
- << " cipher_suite " << cs.c_str();
+ << " cipher_suite " << cs;
return false;
}
diff --git a/talk/session/media/srtpfilter.h b/talk/session/media/srtpfilter.h
index 3c3a8e8..6b941f3 100644
--- a/talk/session/media/srtpfilter.h
+++ b/talk/session/media/srtpfilter.h
@@ -104,16 +104,16 @@
// Just set up both sets of keys directly.
// Used with DTLS-SRTP.
- bool SetRtpParams(const std::string& send_cs,
+ bool SetRtpParams(int send_cs,
const uint8_t* send_key,
int send_key_len,
- const std::string& recv_cs,
+ int recv_cs,
const uint8_t* recv_key,
int recv_key_len);
- bool SetRtcpParams(const std::string& send_cs,
+ bool SetRtcpParams(int send_cs,
const uint8_t* send_key,
int send_key_len,
- const std::string& recv_cs,
+ int recv_cs,
const uint8_t* recv_key,
int recv_key_len);
@@ -138,6 +138,8 @@
// Update the silent threshold (in ms) for signaling errors.
void set_signal_silent_time(uint32_t signal_silent_time_in_ms);
+ bool ResetParams();
+
sigslot::repeater3<uint32_t, Mode, Error> SignalSrtpError;
protected:
@@ -153,7 +155,6 @@
CryptoParams* selected_params);
bool ApplyParams(const CryptoParams& send_params,
const CryptoParams& recv_params);
- bool ResetParams();
static bool ParseKeyParams(const std::string& params, uint8_t* key, int len);
private:
@@ -199,10 +200,10 @@
// Configures the session for sending data using the specified
// cipher-suite and key. Receiving must be done by a separate session.
- bool SetSend(const std::string& cs, const uint8_t* key, int len);
+ bool SetSend(int cs, const uint8_t* key, int len);
// Configures the session for receiving data using the specified
// cipher-suite and key. Sending must be done by a separate session.
- bool SetRecv(const std::string& cs, const uint8_t* key, int len);
+ bool SetRecv(int cs, const uint8_t* key, int len);
// Encrypts/signs an individual RTP/RTCP packet, in-place.
// If an HMAC is used, this will increase the packet size.
@@ -232,7 +233,7 @@
SignalSrtpError;
private:
- bool SetKey(int type, const std::string& cs, const uint8_t* key, int len);
+ bool SetKey(int type, int cs, const uint8_t* key, int len);
// Returns send stream current packet index from srtp db.
bool GetSendStreamPacketIndex(void* data, int in_len, int64_t* index);
diff --git a/talk/session/media/srtpfilter_unittest.cc b/talk/session/media/srtpfilter_unittest.cc
index 8122455..1187438 100644
--- a/talk/session/media/srtpfilter_unittest.cc
+++ b/talk/session/media/srtpfilter_unittest.cc
@@ -508,21 +508,17 @@
// Test directly setting the params with AES_CM_128_HMAC_SHA1_80
TEST_F(SrtpFilterTest, TestProtect_SetParamsDirect_AES_CM_128_HMAC_SHA1_80) {
- EXPECT_TRUE(f1_.SetRtpParams(CS_AES_CM_128_HMAC_SHA1_80,
- kTestKey1, kTestKeyLen,
- CS_AES_CM_128_HMAC_SHA1_80,
+ EXPECT_TRUE(f1_.SetRtpParams(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1,
+ kTestKeyLen, rtc::SRTP_AES128_CM_SHA1_80,
kTestKey2, kTestKeyLen));
- EXPECT_TRUE(f2_.SetRtpParams(CS_AES_CM_128_HMAC_SHA1_80,
- kTestKey2, kTestKeyLen,
- CS_AES_CM_128_HMAC_SHA1_80,
+ EXPECT_TRUE(f2_.SetRtpParams(rtc::SRTP_AES128_CM_SHA1_80, kTestKey2,
+ kTestKeyLen, rtc::SRTP_AES128_CM_SHA1_80,
kTestKey1, kTestKeyLen));
- EXPECT_TRUE(f1_.SetRtcpParams(CS_AES_CM_128_HMAC_SHA1_80,
- kTestKey1, kTestKeyLen,
- CS_AES_CM_128_HMAC_SHA1_80,
+ EXPECT_TRUE(f1_.SetRtcpParams(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1,
+ kTestKeyLen, rtc::SRTP_AES128_CM_SHA1_80,
kTestKey2, kTestKeyLen));
- EXPECT_TRUE(f2_.SetRtcpParams(CS_AES_CM_128_HMAC_SHA1_80,
- kTestKey2, kTestKeyLen,
- CS_AES_CM_128_HMAC_SHA1_80,
+ EXPECT_TRUE(f2_.SetRtcpParams(rtc::SRTP_AES128_CM_SHA1_80, kTestKey2,
+ kTestKeyLen, rtc::SRTP_AES128_CM_SHA1_80,
kTestKey1, kTestKeyLen));
EXPECT_TRUE(f1_.IsActive());
EXPECT_TRUE(f2_.IsActive());
@@ -531,21 +527,17 @@
// Test directly setting the params with AES_CM_128_HMAC_SHA1_32
TEST_F(SrtpFilterTest, TestProtect_SetParamsDirect_AES_CM_128_HMAC_SHA1_32) {
- EXPECT_TRUE(f1_.SetRtpParams(CS_AES_CM_128_HMAC_SHA1_32,
- kTestKey1, kTestKeyLen,
- CS_AES_CM_128_HMAC_SHA1_32,
+ EXPECT_TRUE(f1_.SetRtpParams(rtc::SRTP_AES128_CM_SHA1_32, kTestKey1,
+ kTestKeyLen, rtc::SRTP_AES128_CM_SHA1_32,
kTestKey2, kTestKeyLen));
- EXPECT_TRUE(f2_.SetRtpParams(CS_AES_CM_128_HMAC_SHA1_32,
- kTestKey2, kTestKeyLen,
- CS_AES_CM_128_HMAC_SHA1_32,
+ EXPECT_TRUE(f2_.SetRtpParams(rtc::SRTP_AES128_CM_SHA1_32, kTestKey2,
+ kTestKeyLen, rtc::SRTP_AES128_CM_SHA1_32,
kTestKey1, kTestKeyLen));
- EXPECT_TRUE(f1_.SetRtcpParams(CS_AES_CM_128_HMAC_SHA1_32,
- kTestKey1, kTestKeyLen,
- CS_AES_CM_128_HMAC_SHA1_32,
+ EXPECT_TRUE(f1_.SetRtcpParams(rtc::SRTP_AES128_CM_SHA1_32, kTestKey1,
+ kTestKeyLen, rtc::SRTP_AES128_CM_SHA1_32,
kTestKey2, kTestKeyLen));
- EXPECT_TRUE(f2_.SetRtcpParams(CS_AES_CM_128_HMAC_SHA1_32,
- kTestKey2, kTestKeyLen,
- CS_AES_CM_128_HMAC_SHA1_32,
+ EXPECT_TRUE(f2_.SetRtcpParams(rtc::SRTP_AES128_CM_SHA1_32, kTestKey2,
+ kTestKeyLen, rtc::SRTP_AES128_CM_SHA1_32,
kTestKey1, kTestKeyLen));
EXPECT_TRUE(f1_.IsActive());
EXPECT_TRUE(f2_.IsActive());
@@ -554,25 +546,21 @@
// Test directly setting the params with bogus keys
TEST_F(SrtpFilterTest, TestSetParamsKeyTooShort) {
- EXPECT_FALSE(f1_.SetRtpParams(CS_AES_CM_128_HMAC_SHA1_80,
- kTestKey1, kTestKeyLen - 1,
- CS_AES_CM_128_HMAC_SHA1_80,
+ EXPECT_FALSE(f1_.SetRtpParams(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1,
+ kTestKeyLen - 1, rtc::SRTP_AES128_CM_SHA1_80,
kTestKey1, kTestKeyLen - 1));
- EXPECT_FALSE(f1_.SetRtcpParams(CS_AES_CM_128_HMAC_SHA1_80,
- kTestKey1, kTestKeyLen - 1,
- CS_AES_CM_128_HMAC_SHA1_80,
+ EXPECT_FALSE(f1_.SetRtcpParams(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1,
+ kTestKeyLen - 1, rtc::SRTP_AES128_CM_SHA1_80,
kTestKey1, kTestKeyLen - 1));
}
#if defined(ENABLE_EXTERNAL_AUTH)
TEST_F(SrtpFilterTest, TestGetSendAuthParams) {
- EXPECT_TRUE(f1_.SetRtpParams(CS_AES_CM_128_HMAC_SHA1_32,
- kTestKey1, kTestKeyLen,
- CS_AES_CM_128_HMAC_SHA1_32,
+ EXPECT_TRUE(f1_.SetRtpParams(rtc::SRTP_AES128_CM_SHA1_32, kTestKey1,
+ kTestKeyLen, rtc::SRTP_AES128_CM_SHA1_32,
kTestKey2, kTestKeyLen));
- EXPECT_TRUE(f1_.SetRtcpParams(CS_AES_CM_128_HMAC_SHA1_32,
- kTestKey1, kTestKeyLen,
- CS_AES_CM_128_HMAC_SHA1_32,
+ EXPECT_TRUE(f1_.SetRtcpParams(rtc::SRTP_AES128_CM_SHA1_32, kTestKey1,
+ kTestKeyLen, rtc::SRTP_AES128_CM_SHA1_32,
kTestKey2, kTestKeyLen));
uint8_t* auth_key = NULL;
int auth_key_len = 0, auth_tag_len = 0;
@@ -629,28 +617,30 @@
// Test that we can set up the session and keys properly.
TEST_F(SrtpSessionTest, TestGoodSetup) {
- EXPECT_TRUE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
- EXPECT_TRUE(s2_.SetRecv(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s2_.SetRecv(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
}
// Test that we can't change the keys once set.
TEST_F(SrtpSessionTest, TestBadSetup) {
- EXPECT_TRUE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
- EXPECT_TRUE(s2_.SetRecv(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
- EXPECT_FALSE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_80, kTestKey2, kTestKeyLen));
- EXPECT_FALSE(s2_.SetRecv(CS_AES_CM_128_HMAC_SHA1_80, kTestKey2, kTestKeyLen));
+ EXPECT_TRUE(s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s2_.SetRecv(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_FALSE(
+ s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_80, kTestKey2, kTestKeyLen));
+ EXPECT_FALSE(
+ s2_.SetRecv(rtc::SRTP_AES128_CM_SHA1_80, kTestKey2, kTestKeyLen));
}
// Test that we fail keys of the wrong length.
TEST_F(SrtpSessionTest, TestKeysTooShort) {
- EXPECT_FALSE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, 1));
- EXPECT_FALSE(s2_.SetRecv(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, 1));
+ EXPECT_FALSE(s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, 1));
+ EXPECT_FALSE(s2_.SetRecv(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, 1));
}
// Test that we can encrypt and decrypt RTP/RTCP using AES_CM_128_HMAC_SHA1_80.
TEST_F(SrtpSessionTest, TestProtect_AES_CM_128_HMAC_SHA1_80) {
- EXPECT_TRUE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
- EXPECT_TRUE(s2_.SetRecv(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s2_.SetRecv(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
TestProtectRtp(CS_AES_CM_128_HMAC_SHA1_80);
TestProtectRtcp(CS_AES_CM_128_HMAC_SHA1_80);
TestUnprotectRtp(CS_AES_CM_128_HMAC_SHA1_80);
@@ -659,8 +649,8 @@
// Test that we can encrypt and decrypt RTP/RTCP using AES_CM_128_HMAC_SHA1_32.
TEST_F(SrtpSessionTest, TestProtect_AES_CM_128_HMAC_SHA1_32) {
- EXPECT_TRUE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_32, kTestKey1, kTestKeyLen));
- EXPECT_TRUE(s2_.SetRecv(CS_AES_CM_128_HMAC_SHA1_32, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_32, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s2_.SetRecv(rtc::SRTP_AES128_CM_SHA1_32, kTestKey1, kTestKeyLen));
TestProtectRtp(CS_AES_CM_128_HMAC_SHA1_32);
TestProtectRtcp(CS_AES_CM_128_HMAC_SHA1_32);
TestUnprotectRtp(CS_AES_CM_128_HMAC_SHA1_32);
@@ -668,7 +658,7 @@
}
TEST_F(SrtpSessionTest, TestGetSendStreamPacketIndex) {
- EXPECT_TRUE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_32, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_32, kTestKey1, kTestKeyLen));
int64_t index;
int out_len = 0;
EXPECT_TRUE(s1_.ProtectRtp(rtp_packet_, rtp_len_,
@@ -681,8 +671,8 @@
// Test that we fail to unprotect if someone tampers with the RTP/RTCP paylaods.
TEST_F(SrtpSessionTest, TestTamperReject) {
int out_len;
- EXPECT_TRUE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
- EXPECT_TRUE(s2_.SetRecv(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s2_.SetRecv(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
TestProtectRtp(CS_AES_CM_128_HMAC_SHA1_80);
TestProtectRtcp(CS_AES_CM_128_HMAC_SHA1_80);
rtp_packet_[0] = 0x12;
@@ -694,8 +684,8 @@
// Test that we fail to unprotect if the payloads are not authenticated.
TEST_F(SrtpSessionTest, TestUnencryptReject) {
int out_len;
- EXPECT_TRUE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
- EXPECT_TRUE(s2_.SetRecv(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s2_.SetRecv(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
}
@@ -703,7 +693,7 @@
// Test that we fail when using buffers that are too small.
TEST_F(SrtpSessionTest, TestBuffersTooSmall) {
int out_len;
- EXPECT_TRUE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
EXPECT_FALSE(s1_.ProtectRtp(rtp_packet_, rtp_len_,
sizeof(rtp_packet_) - 10, &out_len));
EXPECT_FALSE(s1_.ProtectRtcp(rtcp_packet_, rtcp_len_,
@@ -717,8 +707,8 @@
static const uint16_t replay_window = 1024;
int out_len;
- EXPECT_TRUE(s1_.SetSend(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
- EXPECT_TRUE(s2_.SetRecv(CS_AES_CM_128_HMAC_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s1_.SetSend(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
+ EXPECT_TRUE(s2_.SetRecv(rtc::SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen));
// Initial sequence number.
rtc::SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_) + 2, seqnum_big);
diff --git a/third_party/gflags/BUILD.gn b/third_party/gflags/BUILD.gn
index a2f1c3d..e8a5a13 100644
--- a/third_party/gflags/BUILD.gn
+++ b/third_party/gflags/BUILD.gn
@@ -13,7 +13,7 @@
}
config("gflags_config") {
- include_dirs = [
+ include_dirs = [
"$gflags_gen_arch_root/include", # For configured files.
"src", # For everything else.
]
@@ -25,6 +25,13 @@
"GFLAGS_DLL_DECLARE_FLAG=",
"GFLAGS_DLL_DEFINE_FLAG=",
]
+
+ # GN orders flags on a target before flags from configs. The default config
+ # adds -Wall, and this flag have to be after -Wall -- so they need to
+ # come from a config and can't be on the target directly.
+ if (is_clang) {
+ cflags = [ "-Wno-unused-local-typedef" ]
+ }
}
source_set("gflags") {
@@ -42,9 +49,7 @@
]
}
- include_dirs = [
- "$gflags_gen_arch_root/include/private", # For config.h
- ]
+ include_dirs = [ "$gflags_gen_arch_root/include/private" ] # For config.h
public_configs = [ ":gflags_config" ]
diff --git a/third_party/gflags/gflags.gyp b/third_party/gflags/gflags.gyp
index 76d2448..d3f2788 100644
--- a/third_party/gflags/gflags.gyp
+++ b/third_party/gflags/gflags.gyp
@@ -79,8 +79,10 @@
},
}],
['clang==1', {
+ 'cflags': ['-Wno-unused-local-typedef',],
'cflags!': ['-Wheader-hygiene',],
'xcode_settings': {
+ 'WARNING_CFLAGS': ['-Wno-unused-local-typedef',],
'WARNING_CFLAGS!': ['-Wheader-hygiene',],
},
}],
diff --git a/third_party/gtest-parallel/README.webrtc b/third_party/gtest-parallel/README.webrtc
index acea04f..7e7fdda 100644
--- a/third_party/gtest-parallel/README.webrtc
+++ b/third_party/gtest-parallel/README.webrtc
@@ -1,5 +1,5 @@
URL: https://github.com/google/gtest-parallel
-Version: c0f8ded77566c657ccc7f745fd9cb070750cccf8
+Version: 92eb6adf9df6eee34bb768b40af984e68e86d7cf
License: Apache 2.0
License File: LICENSE
diff --git a/third_party/gtest-parallel/gtest-parallel b/third_party/gtest-parallel/gtest-parallel
index 0be59e4..3e2fdb4 100755
--- a/third_party/gtest-parallel/gtest-parallel
+++ b/third_party/gtest-parallel/gtest-parallel
@@ -13,16 +13,67 @@
# See the License for the specific language governing permissions and
# limitations under the License.
import cPickle
+import errno
import gzip
import multiprocessing
import optparse
import os
+import signal
import subprocess
import sys
+import tempfile
+import thread
import threading
import time
import zlib
+# An object that catches SIGINT sent to the Python process and notices
+# if processes passed to wait() die by SIGINT (we need to look for
+# both of those cases, because pressing Ctrl+C can result in either
+# the main process or one of the subprocesses getting the signal).
+#
+# Before a SIGINT is seen, wait(p) will simply call p.wait() and
+# return the result. Once a SIGINT has been seen (in the main process
+# or a subprocess, including the one the current call is waiting for),
+# wait(p) will call p.terminate() and raise ProcessWasInterrupted.
+class SigintHandler(object):
+ class ProcessWasInterrupted(Exception): pass
+ sigint_returncodes = {-signal.SIGINT, # Unix
+ -1073741510, # Windows
+ }
+ def __init__(self):
+ self.__lock = threading.Lock()
+ self.__processes = set()
+ self.__got_sigint = False
+ signal.signal(signal.SIGINT, self.__sigint_handler)
+ def __on_sigint(self):
+ self.__got_sigint = True
+ while self.__processes:
+ try:
+ self.__processes.pop().terminate()
+ except OSError:
+ pass
+ def __sigint_handler(self, signal_num, frame):
+ with self.__lock:
+ self.__on_sigint()
+ def got_sigint(self):
+ with self.__lock:
+ return self.__got_sigint
+ def wait(self, p):
+ with self.__lock:
+ if self.__got_sigint:
+ p.terminate()
+ self.__processes.add(p)
+ code = p.wait()
+ with self.__lock:
+ self.__processes.discard(p)
+ if code in self.sigint_returncodes:
+ self.__on_sigint()
+ if self.__got_sigint:
+ raise self.ProcessWasInterrupted
+ return code
+sigint_handler = SigintHandler()
+
# Return the width of the terminal, or None if it couldn't be
# determined (e.g. because we're not being run interactively).
def term_width(out):
@@ -53,15 +104,21 @@
else:
self.__out_file.write("\r" + msg[:self.__width].ljust(self.__width))
self.__previous_line_was_transient = True
- def permanent_line(self, msg):
+ def flush_transient_output(self):
if self.__previous_line_was_transient:
self.__out_file.write("\n")
self.__previous_line_was_transient = False
+ def permanent_line(self, msg):
+ self.flush_transient_output()
self.__out_file.write(msg + "\n")
stdout_lock = threading.Lock()
class FilterFormat:
+ if sys.stdout.isatty():
+ # stdout needs to be unbuffered since the output is interactive.
+ sys.stdout = os.fdopen(sys.stdout.fileno(), 'w', 0)
+
out = Outputter(sys.stdout)
total_tests = 0
finished_tests = 0
@@ -80,7 +137,6 @@
if command == "TEST":
(binary, test) = arg.split(' ', 1)
self.tests[job_id] = (binary, test.strip())
- self.outputs[job_id] = []
elif command == "EXIT":
(exit_code, time_ms) = [int(x) for x in arg.split(' ', 1)]
self.finished_tests += 1
@@ -88,8 +144,9 @@
self.print_test_status(test, time_ms)
if exit_code != 0:
self.failures.append(self.tests[job_id])
- for line in self.outputs[job_id]:
- self.out.permanent_line(line)
+ with open(self.outputs[job_id]) as f:
+ for line in f.readlines():
+ self.out.permanent_line(line.rstrip())
self.out.permanent_line(
"[%d/%d] %s returned/aborted with exit code %d (%d ms)"
% (self.finished_tests, self.total_tests, test, exit_code, time_ms))
@@ -97,17 +154,15 @@
self.total_tests = int(arg.split(' ', 1)[1])
self.out.transient_line("[0/%d] Running tests..." % self.total_tests)
- def add_stdout(self, job_id, output):
- self.outputs[job_id].append(output)
+ def logfile(self, job_id, name):
+ self.outputs[job_id] = name
def log(self, line):
stdout_lock.acquire()
(prefix, output) = line.split(' ', 1)
- if prefix[-1] == ':':
- self.handle_meta(int(prefix[:-1]), output)
- else:
- self.add_stdout(int(prefix[:-1]), output)
+ assert prefix[-1] == ':'
+ self.handle_meta(int(prefix[:-1]), output)
stdout_lock.release()
def end(self):
@@ -116,6 +171,7 @@
% (len(self.failures), self.total_tests))
for (binary, test) in self.failures:
self.out.permanent_line(" " + binary + ": " + test)
+ self.out.flush_transient_output()
class RawFormat:
def log(self, line):
@@ -123,6 +179,10 @@
sys.stdout.write(line + "\n")
sys.stdout.flush()
stdout_lock.release()
+ def logfile(self, job_id, name):
+ with open(self.outputs[job_id]) as f:
+ for line in f.readlines():
+ self.log(str(job_id) + '> ' + line.rstrip())
def end(self):
pass
@@ -149,17 +209,19 @@
return
for ((test_binary, test_name), runtime) in times.items():
if (type(test_binary) is not str or type(test_name) is not str
- or type(runtime) not in {int, long}):
+ or type(runtime) not in {int, long, type(None)}):
return
self.__times = times
def get_test_time(self, binary, testname):
- "Return the last duration for the given test, or 0 if there's no record."
- return self.__times.get((binary, testname), 0)
+ """Return the last duration for the given test as an integer number of
+ milliseconds, or None if the test failed or if there's no record for it."""
+ return self.__times.get((binary, testname), None)
def record_test_time(self, binary, testname, runtime_ms):
- "Record that the given test ran in the specified number of milliseconds."
+ """Record that the given test ran in the specified number of
+ milliseconds. If the test failed, runtime_ms should be None."""
with self.__lock:
self.__times[(binary, testname)] = runtime_ms
@@ -184,6 +246,9 @@
parser = optparse.OptionParser(
usage = 'usage: %prog [options] binary [binary ...] -- [additional args]')
+parser.add_option('-d', '--output_dir', type='string',
+ default=os.path.join(tempfile.gettempdir(), "gtest-parallel"),
+ help='output directory for test logs')
parser.add_option('-r', '--repeat', type='int', default=1,
help='repeat tests')
parser.add_option('-w', '--workers', type='int',
@@ -197,6 +262,8 @@
default=False, help='run disabled tests too')
parser.add_option('--format', type='string', default='filter',
help='output format (raw,filter)')
+parser.add_option('--print_test_times', action='store_true', default=False,
+ help='When done, list the run time of each test')
(options, binaries) = parser.parse_args()
@@ -240,17 +307,21 @@
if line[0] != " ":
test_group = line.strip()
continue
- line = line.strip()
- if not options.gtest_also_run_disabled_tests and 'DISABLED' in line:
- continue
+ # Remove comments for parameterized tests and strip whitespace.
line = line.split('#')[0].strip()
if not line:
continue
test = test_group + line
+ if not options.gtest_also_run_disabled_tests and 'DISABLED_' in test:
+ continue
tests.append((times.get_test_time(test_binary, test),
test_binary, test, command))
-tests.sort(reverse=True)
+
+# Sort tests by falling runtime (with None, which is what we get for
+# new and failing tests, being considered larger than any real
+# runtime).
+tests.sort(reverse=True, key=lambda x: ((1 if x[0] is None else 0), x))
# Repeat tests (-r flag).
tests *= options.repeat
@@ -260,31 +331,41 @@
exit_code = 0
+# Create directory for test log output.
+try:
+ os.makedirs(options.output_dir)
+except OSError as e:
+ # Ignore errors if this directory already exists.
+ if e.errno != errno.EEXIST or not os.path.isdir(options.output_dir):
+ raise e
+# Remove files from old test runs.
+for logfile in os.listdir(options.output_dir):
+ os.remove(os.path.join(options.output_dir, logfile))
+
# Run the specified job. Return the elapsed time in milliseconds if
-# the job succeeds, or a very large number (larger than any reasonable
-# elapsed time) if the job fails. (This ensures that failing tests
-# will run first the next time.)
+# the job succeeds, or None if the job fails. (This ensures that
+# failing tests will run first the next time.)
def run_job((command, job_id, test)):
begin = time.time()
- sub = subprocess.Popen(command + ['--gtest_filter=' + test] +
- ['--gtest_color=' + options.gtest_color],
- stdout = subprocess.PIPE,
- stderr = subprocess.STDOUT)
- while True:
- line = sub.stdout.readline()
- if line == '':
- break
- logger.log(str(job_id) + '> ' + line.rstrip())
+ with tempfile.NamedTemporaryFile(dir=options.output_dir, delete=False) as log:
+ sub = subprocess.Popen(command + ['--gtest_filter=' + test] +
+ ['--gtest_color=' + options.gtest_color],
+ stdout=log.file,
+ stderr=log.file)
+ try:
+ code = sigint_handler.wait(sub)
+ except sigint_handler.ProcessWasInterrupted:
+ thread.exit()
+ runtime_ms = int(1000 * (time.time() - begin))
+ logger.logfile(job_id, log.name)
- code = sub.wait()
- runtime_ms = int(1000 * (time.time() - begin))
logger.log("%s: EXIT %s %d" % (job_id, code, runtime_ms))
if code == 0:
return runtime_ms
global exit_code
exit_code = code
- return sys.maxint
+ return None
def worker():
global job_id
@@ -312,4 +393,10 @@
[t.join() for t in workers]
logger.end()
times.write_to_file(save_file)
-sys.exit(exit_code)
+if options.print_test_times:
+ ts = sorted((times.get_test_time(test_binary, test), test_binary, test)
+ for (_, test_binary, test, _) in tests
+ if times.get_test_time(test_binary, test) is not None)
+ for (time_ms, test_binary, test) in ts:
+ print "%8s %s" % ("%dms" % time_ms, test)
+sys.exit(-signal.SIGINT if sigint_handler.got_sigint() else exit_code)
diff --git a/third_party/winsdk_samples/winsdk_samples.gyp b/third_party/winsdk_samples/winsdk_samples.gyp
index a9b8598..12bc265 100644
--- a/third_party/winsdk_samples/winsdk_samples.gyp
+++ b/third_party/winsdk_samples/winsdk_samples.gyp
@@ -85,6 +85,32 @@
'<(baseclasses_dir)/wxutil.cpp',
'<(baseclasses_dir)/wxutil.h',
],
+ 'conditions': [
+ ['clang==1', {
+ 'msvs_settings': {
+ 'VCCLCompilerTool': {
+ 'AdditionalOptions': [
+ # Disable warnings failing when compiling with Clang on Windows.
+ # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+ '-Wno-comment',
+ '-Wno-delete-non-virtual-dtor',
+ '-Wno-ignored-attributes',
+ '-Wno-logical-op-parentheses',
+ '-Wno-non-pod-varargs',
+ '-Wno-microsoft-extra-qualification',
+ '-Wno-missing-braces',
+ '-Wno-overloaded-virtual',
+ '-Wno-parentheses',
+ '-Wno-reorder',
+ '-Wno-string-conversion',
+ '-Wno-tautological-constant-out-of-range-compare',
+ '-Wno-unused-private-field',
+ '-Wno-writable-strings',
+ ],
+ },
+ },
+ },],
+ ], # conditions.
},
],
}
diff --git a/tools/OWNERS b/tools/OWNERS
index 965de1e..3a3f60c 100644
--- a/tools/OWNERS
+++ b/tools/OWNERS
@@ -1,4 +1,3 @@
kjellander@webrtc.org
phoglund@webrtc.org
niklas.enbom@webrtc.org
-andrew@webrtc.org
diff --git a/tools/autoroller/roll_chromium_revision.py b/tools/autoroller/roll_chromium_revision.py
index 917ecd6..1f974e0 100755
--- a/tools/autoroller/roll_chromium_revision.py
+++ b/tools/autoroller/roll_chromium_revision.py
@@ -25,9 +25,6 @@
CHROMIUM_LOG_TEMPLATE = CHROMIUM_SRC_URL + '/+log/%s'
CHROMIUM_FILE_TEMPLATE = CHROMIUM_SRC_URL + '/+/%s/%s'
-# Run these CQ trybots in addition to the default ones in infra/config/cq.cfg.
-EXTRA_TRYBOTS = 'tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal'
-
COMMIT_POSITION_RE = re.compile('^Cr-Commit-Position: .*#([0-9]+).*$')
CLANG_REVISION_RE = re.compile(r'^CLANG_REVISION=(\d+)$')
ROLL_BRANCH_NAME = 'roll_chromium_revision'
@@ -38,14 +35,14 @@
sys.path.append(CHECKOUT_ROOT_DIR)
import setup_links
-sys.path.append(os.path.join(CHECKOUT_ROOT_DIR, 'tools'))
+sys.path.append(os.path.join(CHECKOUT_ROOT_DIR, 'build'))
import find_depot_tools
find_depot_tools.add_depot_tools_to_path()
from gclient import GClientKeywords
-CLANG_UPDATE_SCRIPT_URL_PATH = 'tools/clang/scripts/update.sh'
+CLANG_UPDATE_SCRIPT_URL_PATH = 'tools/clang/scripts/update.py'
CLANG_UPDATE_SCRIPT_LOCAL_PATH = os.path.join('tools', 'clang', 'scripts',
- 'update.sh')
+ 'update.py')
DepsEntry = collections.namedtuple('DepsEntry', 'path url revision')
ChangedDep = collections.namedtuple('ChangedDep',
@@ -292,7 +289,6 @@
commit_msg.append('No update to Clang.\n')
commit_msg.append('TBR=%s' % tbr_authors)
- commit_msg.append('CQ_EXTRA_TRYBOTS=%s' % EXTRA_TRYBOTS)
return '\n'.join(commit_msg)
diff --git a/tools/refactoring/addfileheader.py b/tools/refactoring/addfileheader.py
deleted file mode 100644
index 01c8a8b..0000000
--- a/tools/refactoring/addfileheader.py
+++ /dev/null
@@ -1,163 +0,0 @@
-#!/usr/bin/env python
-
-import stringmanipulation
-import filemanagement
-import sys
-
-extensions = ['.h','.cc','.c','.cpp']
-
-ignore_these = ['my_ignore_header.h']
-
-if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
- print 'parameters are: directory [--commit]'
- quit()
-
-directory = sys.argv[1];
-if(not filemanagement.pathexist(directory)):
- print 'path ' + directory + ' does not exist'
- quit()
-
-if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
- print 'parameters are: parent directory extension new extension [--commit]'
- quit()
-
-commit = False
-if(len(sys.argv) == 3):
- commit = True
-
-files_to_fix = []
-for extension in extensions:
- files_to_fix.extend(filemanagement.listallfilesinfolder(directory,\
- extension))
-
-# Just steal the header from the template
-def fileheaderasstring():
- template_file_name = 'license_template.txt'
- if (not filemanagement.fileexist(template_file_name)):
- print 'File ' + template_file_name + ' not found!'
- quit()
- template_file = open(template_file_name,'r')
- return_string = ''
- for line in template_file:
- return_string += line
- return return_string
-
-# Just steal the header from the template
-def fileheaderasarray():
- template_file_name = 'license_template.txt'
- if (not filemanagement.fileexist(template_file_name)):
- print 'File ' + template_file_name + ' not found!'
- quit()
- template_file = open(template_file_name,'r')
- return_value = []
- for line in template_file:
- return_value.append(line)
- return return_value
-
-
-def findheader(path, file_name):
- full_file_name = path + file_name
- if (not filemanagement.fileexist(full_file_name)):
- print 'File ' + file_name + ' not found!'
- print 'Unexpected error!'
- quit()
- file_handle = open(full_file_name)
- template_file_content = fileheaderasarray()
- compare_content = []
- # load the same number of lines from file as the fileheader
- for index in range(len(template_file_content)):
- line = file_handle.readline()
- if (line == ''):
- return False
- compare_content.append(line)
-
- while (True):
- found = True
- for index in range(len(template_file_content)):
- line1 = template_file_content[index]
- line2 = compare_content[index]
- if(line1 != line2):
- found = False
- break
- if (found):
- return True
- compare_content = compare_content[1:len(compare_content)]
- line = file_handle.readline()
- if (line == ''):
- return False
- compare_content.append(line)
- return False
-
-# Used to store temporary result before flushing to real file when finished
-def temporaryfilename(old_file_name):
- return old_file_name + '.deleteme'
-
-def updatefile(path, old_file_name):
- full_old_file_name = path + old_file_name
- if (not filemanagement.fileexist(full_old_file_name)):
- print 'File ' + full_old_file_name + ' is not found.'
- print 'Should not happen! Ever!'
- quit()
-
- full_temporary_file_name = path + temporaryfilename(old_file_name)
-
- # Make sure that the files are closed by putting them out of scope
- old_file = open(full_old_file_name,'r')
- temporary_file = open(full_temporary_file_name,'w')
-
- temporary_file.writelines(fileheaderasstring())
- remove_whitespaces = True
- for line in old_file:
- if (remove_whitespaces and (len(line.split()) == 0)):
- continue
- else:
- remove_whitespaces = False
- temporary_file.writelines(line)
- old_file.close()
- temporary_file.close()
-
- filemanagement.copyfile(full_old_file_name,full_temporary_file_name)
- filemanagement.deletefile(full_temporary_file_name)
-
-
-failed_files = []
-skipped_files = []
-for index in range(len(files_to_fix)):
- if(commit):
- print (100*index)/len(files_to_fix)
- path_dir = files_to_fix[index][0]
- filename = files_to_fix[index][1]
- is_ignore = False
- for ignore_names in ignore_these:
- if(filename == ignore_names):
- is_ignore = True
- break
- if(is_ignore):
- continue
-
-# Let the word copyright be our sanity, i.e. make sure there is only one
-# copy right occurance or report that there will be no change
- if(filemanagement.findstringinfile(path_dir,filename,'Copyright') or
- filemanagement.findstringinfile(path_dir,filename,'copyright') or
- filemanagement.findstringinfile(path_dir,filename,'COPYRIGHT')):
- if(findheader(path_dir,filename)):
- skipped_files.append(path_dir + filename)
- else:
- failed_files.append(path_dir + filename)
- continue
-
- if (not commit):
- print 'File ' + path_dir + filename + ' will be updated'
- continue
- updatefile(path_dir,filename)
-
-tense = 'will be'
-if (commit):
- tense = 'has been'
-if (len(skipped_files) > 0):
- print str(len(skipped_files)) + ' file(s) ' + tense + ' skipped since they already have the correct header'
-
-if (len(failed_files) > 0):
- print 'Following files seem to have an invalid file header:'
-for line in failed_files:
- print line
diff --git a/tools/refactoring/filemanagement.py b/tools/refactoring/filemanagement.py
deleted file mode 100644
index 4ff64ce..0000000
--- a/tools/refactoring/filemanagement.py
+++ /dev/null
@@ -1,72 +0,0 @@
-import fnmatch
-import os
-import stringmanipulation
-
-def fileexist( file_name ):
- return os.path.isfile(file_name)
-
-def pathexist( path ):
- return os.path.exists(path)
-
-def fixpath( path ):
- return_value = path
- if( return_value[len(return_value) - 1] != '/'):
- return_value = return_value + '/'
- return return_value
-
-def listallfilesinfolder( path, extension ):
- matches = []
- signature = '*' + extension
- for root, dirnames, filenames in os.walk(path):
- for filename in fnmatch.filter(filenames, signature):
- matches.append([fixpath(root), filename])
- return matches
-
-def copyfile(to_file, from_file):
- if(not fileexist(from_file)):
- return
- command = 'cp -f ' + from_file + ' ' + to_file
- os.system(command)
- #print command
-
-def deletefile(file_to_delete):
- if(not fileexist(file_to_delete)):
- return
- os.system('rm ' + file_to_delete)
-
-# very ugly but works, so keep for now
-def findstringinfile(path,file_name,search_string):
- command = 'grep \'' + search_string + '\' ' + path + file_name + ' > deleteme.txt'
- return_value = os.system(command)
-# print command
- return (return_value == 0)
-
-def replacestringinfolder( path, old_string, new_string, extension ):
- if(not stringmanipulation.isextension(extension)):
- print 'failed to search and replace'
- return
- if(len(old_string) == 0):
- print 'failed to search and replace'
- return
- find_command = 'ls '+ path + '/*' + extension
- sed_command = 'sed -i \'s/' + old_string + '/' + new_string +\
- '/g\' *' + extension
- command_string = find_command + ' | xargs ' + sed_command + ' 2> deleteme.txt'
- os.system(command_string)
- #print command_string
-
-#find ./ -name "*.h" -type f | xargs -P 0 sed -i 's/process_thread_wrapper.h/process_thread.h/g' *.h deleteme.txt
-def replacestringinallsubfolders( old_string, new_string, extension):
- if(not stringmanipulation.isextension(extension)):
- print 'failed to search and replace'
- return
- if(len(old_string) == 0):
- print 'failed to search and replace'
- return
-
- find_command = 'find ./ -name \"*' + extension + '\" -type f'
- sed_command = 'sed -i \'s/' + old_string + '/' + new_string +\
- '/g\' *' + extension
- command_string = find_command + ' | xargs -P 0 ' + sed_command + ' 2> deleteme.txt'
- os.system(command_string)
- #print command_string
diff --git a/tools/refactoring/fixincludeguards.py b/tools/refactoring/fixincludeguards.py
deleted file mode 100644
index 0b56355..0000000
--- a/tools/refactoring/fixincludeguards.py
+++ /dev/null
@@ -1,145 +0,0 @@
-#!/usr/bin/env python
-
-import stringmanipulation
-import filemanagement
-import sys
-
-extensions = ['.h']
-
-ignore_these = ['my_ignore_header.h']
-
-if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
- print 'parameters are: directory [--commit]'
- quit()
-
-directory = sys.argv[1];
-if(not filemanagement.pathexist(directory)):
- print 'path ' + directory + ' does not exist'
- quit()
-
-if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
- print 'parameters are: parent directory extension new extension [--commit]'
- quit()
-
-commit = False
-if(len(sys.argv) == 3):
- commit = True
-
-for extension in extensions:
- files_to_fix = filemanagement.listallfilesinfolder(directory,\
- extension)
-
-def buildincludeguardname(path,filename):
- full_file_name = 'WEBRTC_' + path + filename
- full_file_name = full_file_name.upper()
- full_file_name = stringmanipulation.replaceoccurances(full_file_name, '/', '_')
- full_file_name = stringmanipulation.replaceoccurances(full_file_name, '\\', '_')
- full_file_name = stringmanipulation.replaceoccurances(full_file_name, '.', '_')
- full_file_name += '_'
- return full_file_name
-
-def buildnewincludeguardset(path,filename):
- include_guard_name = buildincludeguardname(path,filename)
- if(include_guard_name == ''):
- return []
- return_value = []
- return_value.append('#ifndef ' + include_guard_name)
- return_value.append('#define ' + include_guard_name)
- return_value.append(include_guard_name)
- return return_value
-
-def printincludeguardset(include_guard_set):
- print 'First line: ' + include_guard_set[0]
- print 'Second line: ' + include_guard_set[1]
- print 'Last line: ' + include_guard_set[2]
- return
-
-include_guard_begin_identifier = ['#ifndef', '#if !defined']
-include_guard_second_identifier = ['#define']
-def findincludeguardidentifier(line):
- for begin_identifier in include_guard_begin_identifier:
- line = stringmanipulation.removealloccurances(line,begin_identifier)
- for second_identifier in include_guard_begin_identifier:
- line = stringmanipulation.removealloccurances(line,second_identifier)
- removed_prefix = [True,'']
- line = stringmanipulation.whitespacestoonespace(line)
- while(removed_prefix[0]):
- removed_prefix = stringmanipulation.removeprefix(line,' ')
- line = removed_prefix[1]
- line = stringmanipulation.removealloccurances(line,'(')
- if(line == ''):
- return ''
- word_pos = stringmanipulation.getword(line,0)
- return_value = line[0:word_pos[1]]
- return_value = return_value.rstrip('\r\n')
- return return_value
-
-def findoldincludeguardset(path,filename):
- return_value = []
- full_file_name = path + filename
- file_pointer = open(full_file_name,'r')
- include_guard_name = ''
- for line in file_pointer:
- if (include_guard_name == ''):
- for compare_string in include_guard_begin_identifier:
- if (stringmanipulation.issubstring(compare_string, line) != -1):
- include_guard_name = findincludeguardidentifier(line)
- if (include_guard_name == ''):
- break
- line = line.rstrip('\r\n')
- return_value.append(line)
- break
- else:
- for compare_string in include_guard_second_identifier:
- if (stringmanipulation.issubstring(compare_string, line) != -1):
- if (stringmanipulation.issubstring(include_guard_name, line) != -1):
- line = line.rstrip('\r\n')
- return_value.append(line)
- return_value.append(include_guard_name)
- return return_value
- include_guard_name = ''
- return_value = []
- return []
-
-failed_files = []
-for index in range(len(files_to_fix)):
- if(commit):
- print (100*index)/len(files_to_fix)
- path_dir = files_to_fix[index][0]
- filename = files_to_fix[index][1]
- is_ignore = False
- for ignore_names in ignore_these:
- if(filename == ignore_names):
- is_ignore = True
- break
- if(is_ignore):
- continue
- old_include_guard_set = findoldincludeguardset(path_dir,filename)
- if (len(old_include_guard_set) != 3) :
- failed_files.append('unable to figure out the include guards for ' + filename)
- continue
-
- new_include_guard_set = buildnewincludeguardset(path_dir,filename)
- if (len(new_include_guard_set) != 3) :
- failed_files.append('unable to figure out new the include guards for ' + filename)
- continue
-
- if(not commit):
- print 'old guard: ' + old_include_guard_set[2]
- print 'new guard: ' + new_include_guard_set[2]
- continue
-
- for index in range(2):
- # enough to only replace for file. However, no function for that
- for extension in extensions:
- filemanagement.replacestringinfolder(path_dir,old_include_guard_set[index],new_include_guard_set[index],extension)
- # special case for last to avoid complications
- for extension in extensions:
- filemanagement.replacestringinfolder(path_dir,' ' + old_include_guard_set[2],' ' + new_include_guard_set[2],extension)
- filemanagement.replacestringinfolder(path_dir,'\\/\\/' + old_include_guard_set[2],'\\/\\/ ' + new_include_guard_set[2],extension)
-
-
-if(len(failed_files) > 0):
- print 'Following failures should be investigated manually:'
-for line in failed_files:
- print line
diff --git a/tools/refactoring/fixnames.py b/tools/refactoring/fixnames.py
deleted file mode 100644
index 15381e3..0000000
--- a/tools/refactoring/fixnames.py
+++ /dev/null
@@ -1,387 +0,0 @@
-#!/usr/bin/env python
-
-import stringmanipulation
-import filemanagement
-import p4commands
-import sys
-
-name_space_to_ignore = 'GIPS::'
-#only allow one prefix to be removed since allowing multiple will complicate
-# things
-prefix_to_filter = 'gips'
-#words_to_filter = ['Module']
-# it might be dangerous to remove GIPS but keep it default
-words_to_filter = ['Module','GIPS']
-
-# This script finds all the words that should be replaced in an h-file. Once
-# all words that should be replaced are found it does a global search and
-# replace.
-
-extensions_to_edit = ['.cpp','.cc','.h']
-
-#line = ' ~hiGIPSCriticalSectionScoped()'
-#print line
-#position = stringmanipulation.getword(line,11)
-#old_word = line[position[0]:position[0]+position[1]]
-#result = stringmanipulation.removealloccurances(old_word,'gips')
-#new_word = result
-#print old_word
-#print position[0]
-#print position[0]+position[1]
-#print new_word
-#quit()
-
-# Ignore whole line if any item in this table is a substring of the line
-do_not_replace_line_table = []
-do_not_replace_line_table.append('namespace GIPS')
-
-# [old_string,new_string]
-# List of things to remove that are static:
-manual_replace_table = []
-#manual_replace_table.append(['using namespace GIPS;',''])
-#manual_replace_table.append(['CreateGipsEvent','CreateEvent'])
-#manual_replace_table.append(['CreateGIPSTrace','CreateTrace'])
-#manual_replace_table.append(['ReturnGIPSTrace','ReturnTrace'])
-#manual_replace_table.append(['CreateGIPSFile','CreateFile'])
-replace_table = manual_replace_table
-#replace_table.append(['GIPS::','webrtc::'])
-# List of things to not remove that are static, i.e. exceptions:
-# don't replace any of the GIPS_Words since that will affect all files
-# do that in a separate script!
-do_not_replace_table = []
-do_not_replace_table.append('GIPS_CipherTypes')
-do_not_replace_table.append('GIPS_AuthenticationTypes')
-do_not_replace_table.append('GIPS_SecurityLevels')
-do_not_replace_table.append('GIPS_encryption')
-do_not_replace_table.append('~GIPS_encryption')
-do_not_replace_table.append('GIPS_transport')
-do_not_replace_table.append('~GIPS_transport')
-do_not_replace_table.append('GIPSTraceCallback')
-do_not_replace_table.append('~GIPSTraceCallback')
-do_not_replace_table.append('GIPS_RTP_CSRC_SIZE')
-do_not_replace_table.append('GIPS_RTPDirections')
-do_not_replace_table.append('GIPS_RTP_INCOMING')
-do_not_replace_table.append('GIPS_RTP_OUTGOING')
-do_not_replace_table.append('GIPSFrameType')
-do_not_replace_table.append('GIPS_FRAME_EMPTY')
-do_not_replace_table.append('GIPS_AUDIO_FRAME_SPEECH')
-do_not_replace_table.append('GIPS_AUDIO_FRAME_CN')
-do_not_replace_table.append('GIPS_VIDEO_FRAME_KEY')
-do_not_replace_table.append('GIPS_VIDEO_FRAME_DELTA')
-do_not_replace_table.append('GIPS_VIDEO_FRAME_GOLDEN')
-do_not_replace_table.append('GIPS_VIDEO_FRAME_DELTA_KEY')
-do_not_replace_table.append('GIPS_PacketType')
-do_not_replace_table.append('GIPS_PACKET_TYPE_RTP')
-do_not_replace_table.append('GIPS_PACKET_TYPE_KEEP_ALIVE')
-do_not_replace_table.append('GIPS_AudioLayers')
-do_not_replace_table.append('GIPS_AUDIO_PLATFORM_DEFAULT')
-do_not_replace_table.append('GIPS_AUDIO_WINDOWS_WAVE')
-do_not_replace_table.append('GIPS_AUDIO_WINDOWS_CORE')
-do_not_replace_table.append('GIPS_AUDIO_LINUX_ALSA')
-do_not_replace_table.append('GIPS_AUDIO_LINUX_PULSE')
-do_not_replace_table.append('GIPS_AUDIO_FORMAT')
-do_not_replace_table.append('GIPS_PCM_16_16KHZ')
-do_not_replace_table.append('GIPS_PCM_16_8KHZ')
-do_not_replace_table.append('GIPS_G729')
-do_not_replace_table.append('GIPSAMRmode')
-do_not_replace_table.append('GIPS_RFC3267_BWEFFICIENT')
-do_not_replace_table.append('GIPS_RFC3267_OCTETALIGNED')
-do_not_replace_table.append('GIPS_RFC3267_FILESTORAGE')
-do_not_replace_table.append('GIPS_NCModes')
-do_not_replace_table.append('GIPS_NC_OFF')
-do_not_replace_table.append('GIPS_NC_MILD')
-do_not_replace_table.append('GIPS_NC_MODERATE')
-do_not_replace_table.append('GIPS_NC_AGGRESSIVE')
-do_not_replace_table.append('GIPS_NC_VERY_AGGRESSIVE')
-do_not_replace_table.append('GIPS_AGCModes')
-do_not_replace_table.append('GIPS_AGC_OFF')
-do_not_replace_table.append('GIPS_AGC_ANALOG')
-do_not_replace_table.append('GIPS_AGC_DIGITAL')
-do_not_replace_table.append('GIPS_AGC_STANDALONE_DIG')
-do_not_replace_table.append('GIPS_ECModes')
-do_not_replace_table.append('GIPS_EC_UNCHANGED')
-do_not_replace_table.append('GIPS_EC_DEFAULT')
-do_not_replace_table.append('GIPS_EC_CONFERENCE')
-do_not_replace_table.append('GIPS_EC_AEC')
-do_not_replace_table.append('GIPS_EC_AES')
-do_not_replace_table.append('GIPS_EC_AECM')
-do_not_replace_table.append('GIPS_EC_NEC_IAD')
-do_not_replace_table.append('GIPS_AESModes')
-do_not_replace_table.append('GIPS_AES_DEFAULT')
-do_not_replace_table.append('GIPS_AES_NORMAL')
-do_not_replace_table.append('GIPS_AES_HIGH')
-do_not_replace_table.append('GIPS_AES_ATTENUATE')
-do_not_replace_table.append('GIPS_AES_NORMAL_SOFT_TRANS')
-do_not_replace_table.append('GIPS_AES_HIGH_SOFT_TRANS')
-do_not_replace_table.append('GIPS_AES_ATTENUATE_SOFT_TRANS')
-do_not_replace_table.append('GIPS_AECMModes')
-do_not_replace_table.append('GIPS_AECM_QUIET_EARPIECE_OR_HEADSET')
-do_not_replace_table.append('GIPS_AECM_EARPIECE')
-do_not_replace_table.append('GIPS_AECM_LOUD_EARPIECE')
-do_not_replace_table.append('GIPS_AECM_SPEAKERPHONE')
-do_not_replace_table.append('GIPS_AECM_LOUD_SPEAKERPHONE')
-do_not_replace_table.append('AECM_LOUD_SPEAKERPHONE')
-do_not_replace_table.append('GIPS_VAD_CONVENTIONAL')
-do_not_replace_table.append('GIPS_VAD_AGGRESSIVE_LOW')
-do_not_replace_table.append('GIPS_VAD_AGGRESSIVE_MID')
-do_not_replace_table.append('GIPS_VAD_AGGRESSIVE_HIGH')
-do_not_replace_table.append('GIPS_NetEQModes')
-do_not_replace_table.append('GIPS_NETEQ_DEFAULT')
-do_not_replace_table.append('GIPS_NETEQ_STREAMING')
-do_not_replace_table.append('GIPS_NETEQ_FAX')
-do_not_replace_table.append('GIPS_NetEQBGNModes')
-do_not_replace_table.append('GIPS_BGN_ON')
-do_not_replace_table.append('GIPS_BGN_FADE')
-do_not_replace_table.append('GIPS_BGN_OFF')
-do_not_replace_table.append('GIPS_OnHoldModes')
-do_not_replace_table.append('GIPS_HOLD_SEND_AND_PLAY')
-do_not_replace_table.append('GIPS_HOLD_SEND_ONLY')
-do_not_replace_table.append('GIPS_HOLD_PLAY_ONLY')
-do_not_replace_table.append('GIPS_PayloadFrequencies')
-do_not_replace_table.append('GIPS_FREQ_8000_HZ')
-do_not_replace_table.append('GIPS_FREQ_16000_HZ')
-do_not_replace_table.append('GIPS_FREQ_32000_HZ')
-do_not_replace_table.append('GIPS_TelephoneEventDetectionMethods')
-do_not_replace_table.append('GIPS_IN_BAND')
-do_not_replace_table.append('GIPS_OUT_OF_BAND')
-do_not_replace_table.append('GIPS_IN_AND_OUT_OF_BAND')
-do_not_replace_table.append('GIPS_ProcessingTypes')
-do_not_replace_table.append('GIPS_PLAYBACK_PER_CHANNEL')
-do_not_replace_table.append('GIPS_PLAYBACK_ALL_CHANNELS_MIXED')
-do_not_replace_table.append('GIPS_RECORDING_PER_CHANNEL')
-do_not_replace_table.append('GIPS_RECORDING_ALL_CHANNELS_MIXED')
-do_not_replace_table.append('GIPS_StereoChannel')
-do_not_replace_table.append('GIPS_StereoLeft')
-do_not_replace_table.append('GIPS_StereoRight')
-do_not_replace_table.append('GIPS_StereoBoth')
-do_not_replace_table.append('GIPS_stat_val')
-do_not_replace_table.append('GIPS_P56_statistics')
-do_not_replace_table.append('GIPS_echo_statistics')
-do_not_replace_table.append('GIPS_NetworkStatistics')
-do_not_replace_table.append('GIPS_JitterStatistics')
-do_not_replace_table.append('GIPSVideoRawType')
-do_not_replace_table.append('GIPS_VIDEO_I420')
-do_not_replace_table.append('GIPS_VIDEO_YV12')
-do_not_replace_table.append('GIPS_VIDEO_YUY2')
-do_not_replace_table.append('GIPS_VIDEO_UYVY')
-do_not_replace_table.append('GIPS_VIDEO_IYUV')
-do_not_replace_table.append('GIPS_VIDEO_ARGB')
-do_not_replace_table.append('GIPS_VIDEO_RGB24')
-do_not_replace_table.append('GIPS_VIDEO_RGB565')
-do_not_replace_table.append('GIPS_VIDEO_ARGB4444')
-do_not_replace_table.append('GIPS_VIDEO_ARGB1555')
-do_not_replace_table.append('GIPS_VIDEO_MJPG')
-do_not_replace_table.append('GIPS_VIDEO_NV12')
-do_not_replace_table.append('GIPS_VIDEO_NV21')
-do_not_replace_table.append('GIPS_VIDEO_Unknown')
-do_not_replace_table.append('GIPSVideoLayouts')
-do_not_replace_table.append('GIPS_LAYOUT_NONE')
-do_not_replace_table.append('GIPS_LAYOUT_DEFAULT')
-do_not_replace_table.append('GIPS_LAYOUT_ADVANCED1')
-do_not_replace_table.append('GIPS_LAYOUT_ADVANCED2')
-do_not_replace_table.append('GIPS_LAYOUT_ADVANCED3')
-do_not_replace_table.append('GIPS_LAYOUT_ADVANCED4')
-do_not_replace_table.append('GIPS_LAYOUT_FULL')
-do_not_replace_table.append('KGIPSConfigParameterSize')
-do_not_replace_table.append('KGIPSPayloadNameSize')
-do_not_replace_table.append('GIPSVideoCodecH263')
-do_not_replace_table.append('GIPSVideoH264Packetization')
-do_not_replace_table.append('GIPS_H264_SingleMode')
-do_not_replace_table.append('GIPS_H264_NonInterleavedMode')
-do_not_replace_table.append('GIPSVideoCodecComplexity')
-do_not_replace_table.append('GIPSVideoCodec_Complexity_Normal')
-do_not_replace_table.append('GIPSVideoCodec_Comlexity_High')
-do_not_replace_table.append('GIPSVideoCodec_Comlexity_Higher')
-do_not_replace_table.append('GIPSVideoCodec_Comlexity_Max')
-do_not_replace_table.append('GIPSVideoCodecH264')
-do_not_replace_table.append('GIPSVideoH264Packetization')
-do_not_replace_table.append('GIPSVideoCodecComplexity')
-do_not_replace_table.append('GIPSVideoCodecProfile')
-do_not_replace_table.append('KGIPSConfigParameterSize')
-do_not_replace_table.append('KGIPSMaxSVCLayers')
-do_not_replace_table.append('GIPSVideoH264LayerTypes')
-do_not_replace_table.append('GIPS_H264SVC_Base')
-do_not_replace_table.append('GIPS_H264SVC_Extend_2X2')
-do_not_replace_table.append('GIPS_H264SVC_Extend_1X1')
-do_not_replace_table.append('GIPS_H264SVC_Extend_MGS')
-do_not_replace_table.append('GIPS_H264SVC_Extend_1_5')
-do_not_replace_table.append('GIPS_H264SVC_Extend_Custom')
-do_not_replace_table.append('GIPSVideoH264LayersProperties')
-do_not_replace_table.append('GIPSVideoH264LayerTypes')
-do_not_replace_table.append('GIPSVideoH264Layers')
-do_not_replace_table.append('GIPSVideoH264LayersProperties')
-do_not_replace_table.append('GIPSVideoCodecH264SVC')
-do_not_replace_table.append('GIPSVideoCodecComplexity')
-do_not_replace_table.append('GIPSVideoCodecProfile')
-do_not_replace_table.append('GIPSVideoH264Layers')
-do_not_replace_table.append('GIPSVideoCodecVP8')
-do_not_replace_table.append('GIPSVideoCodecComplexity')
-do_not_replace_table.append('GIPSVideoCodecMPEG')
-do_not_replace_table.append('GIPSVideoCodecGeneric')
-do_not_replace_table.append('GIPSVideoCodecType')
-do_not_replace_table.append('GIPSVideoCodec_H263')
-do_not_replace_table.append('GIPSVideoCodec_H264')
-do_not_replace_table.append('GIPSVideoCodec_H264SVC')
-do_not_replace_table.append('GIPSVideoCodec_VP8')
-do_not_replace_table.append('GIPSVideoCodec_MPEG4')
-do_not_replace_table.append('GIPSVideoCodec_I420')
-do_not_replace_table.append('GIPSVideoCodec_RED')
-do_not_replace_table.append('GIPSVideoCodec_ULPFEC')
-do_not_replace_table.append('GIPSVideoCodec_Unknown')
-do_not_replace_table.append('GIPSVideoCodecUnion')
-do_not_replace_table.append('GIPSVideoCodecH263')
-do_not_replace_table.append('GIPSVideoCodecH264')
-do_not_replace_table.append('GIPSVideoCodecH264SVC')
-do_not_replace_table.append('GIPSVideoCodecVP8')
-do_not_replace_table.append('GIPSVideoCodecMPEG4')
-do_not_replace_table.append('GIPSVideoCodecGeneric')
-do_not_replace_table.append('GIPSVideoCodec')
-do_not_replace_table.append('GIPSVideoCodecType')
-do_not_replace_table.append('GIPSVideoCodecUnion')
-do_not_replace_table.append('GIPSAudioFrame')
-do_not_replace_table.append('GIPS_CodecInst')
-do_not_replace_table.append('GIPS_FileFormats')
-do_not_replace_table.append('GIPSTickTime')
-do_not_replace_table.append('GIPS_Word64')
-do_not_replace_table.append('GIPS_UWord64')
-do_not_replace_table.append('GIPS_Word32')
-do_not_replace_table.append('GIPS_UWord32')
-do_not_replace_table.append('GIPS_Word16')
-do_not_replace_table.append('GIPS_UWord16')
-do_not_replace_table.append('GIPS_Word8')
-do_not_replace_table.append('GIPS_UWord8')
-
-if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
- print 'parameters are: parent directory [--commit]'
- quit()
-
-if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
- print 'parameters are: parent directory [--commit]'
- quit()
-
-commit = (len(sys.argv) == 3)
-
-directory = sys.argv[1];
-if(not filemanagement.pathexist(directory)):
- print 'path ' + directory + ' does not exist'
- quit()
-
-# APIs are all in h-files
-extension = '.h'
-
-# All h-files
-files_to_modify = filemanagement.listallfilesinfolder(directory,\
- extension)
-
-def isinmanualremovetable( compare_word ):
- for old_word, new_word in manual_replace_table:
- if(old_word == compare_word):
- return True
- return False
-
-# Begin
-# This function looks at each line and decides which words should be replaced
-# that is this is the only part of the script that you will ever want to change!
-def findstringstoreplace(line):
- original_line = line
-# Dont replace compiler directives
- if(line[0] == '#'):
- return []
-# Dont allow global removal of namespace gips since it is very intrusive
- for sub_string_compare in do_not_replace_line_table:
- index = stringmanipulation.issubstring(line,sub_string_compare)
- if(index != -1):
- return []
-
- return_value = []
-
- line = stringmanipulation.removeccomment(line)
- line = stringmanipulation.whitespacestoonespace(line)
- if(len(line) == 0):
- return []
- if(line[0] == '*'):
- return []
- index = stringmanipulation.issubstring(line,prefix_to_filter)
- while index >= 0:
- dont_store_hit = False
- word_position = stringmanipulation.getword(line, index)
- start_of_word = word_position[0]
- size_of_word = word_position[1]
- end_of_word = start_of_word + size_of_word
- old_word = line[start_of_word:end_of_word]
- if(isinmanualremovetable(old_word)):
- dont_store_hit = True
- if((end_of_word + 2 < len(line)) and\
- name_space_to_ignore == line[start_of_word:end_of_word+2]):
- dont_store_hit = True
-
- result = stringmanipulation.removeprefix(old_word,prefix_to_filter)
- new_word = result[1]
- for word_to_filter in words_to_filter:
- new_word = stringmanipulation.removealloccurances(new_word,word_to_filter)
- result = stringmanipulation.removeprefix(new_word,'_')
- new_word = result[1]
- new_word = stringmanipulation.fixabbreviations(new_word)
- new_word = stringmanipulation.removealloccurances(new_word,'_')
- if(not dont_store_hit):
- return_value.append([old_word,new_word])
-# remove the word we found from the string so we dont find it again
- line = line[0:start_of_word] + line[end_of_word:len(line)]
- index = stringmanipulation.issubstring(line,'GIPS')
-
- return return_value
-# End
-
-# loop through all files
-for path, file_name in files_to_modify:
-# if(file_name != 'GIPSTickUtil.h'):
-# continue
- full_file_name = path + file_name
- file_pointer = open(full_file_name,'r')
-# print file_name
-#loop through all lines
- for line in file_pointer:
-# print line
- local_replace_string = findstringstoreplace(line)
- #print local_replace_string
- if(len(local_replace_string) != 0):
- replace_table.extend(local_replace_string)
-
-
-# we have built our replace table now
-replace_table = stringmanipulation.removeduplicates( replace_table )
-replace_table = stringmanipulation.ordertablesizefirst( replace_table )
-replace_table = stringmanipulation.complement(replace_table,\
- do_not_replace_table)
-
-def replaceoriginal( path,my_table ):
- size_of_table = len(my_table)
- for index in range(len(my_table)):
- old_name = my_table[index][0]
- new_name = my_table[index][1]
- filemanagement.replacestringinfolder(path, old_name, new_name,\
- ".h")
- print (100*index) / (size_of_table*2)
-
-def replaceall( my_table, extension_list ):
- size_of_table = len(my_table)
- for index in range(len(my_table)):
- old_name = my_table[index][0]
- new_name = my_table[index][1]
- new_name = new_name
- for extension in extensions_to_edit:
- filemanagement.replacestringinallsubfolders(old_name, new_name,
- extension)
- print 100*(size_of_table + index) / (size_of_table*2)
-
-
-if(commit):
- print 'commiting'
- replace_table = stringmanipulation.removenochange(replace_table)
- p4commands.checkoutallfiles()
- replaceoriginal(directory,replace_table)
- replaceall(replace_table,extensions_to_edit)
- p4commands.revertunchangedfiles()
-else:
- for old_name, new_name in replace_table:
- print 'Going to replace [' + old_name + '] with [' + new_name + ']'
diff --git a/tools/refactoring/integratefiles.py b/tools/refactoring/integratefiles.py
deleted file mode 100644
index c5cc892..0000000
--- a/tools/refactoring/integratefiles.py
+++ /dev/null
@@ -1,100 +0,0 @@
-#!/usr/bin/env python
-
-import stringmanipulation
-import filemanagement
-import p4commands
-import sys
-
-extensions = ['.h', '.cpp', '.cc', '.gyp']
-
-ignore_these = ['list_no_stl.h','map_no_stl.h','constructor_magic.h']
-
-exceptions = [
-['GIPSRWLock.h','rw_lock.h'],
-['GIPSCriticalsection.h','critical_section.h'],
-]
-
-if((len(sys.argv) != 4) and (len(sys.argv) != 5)):
- print 'parameters are: parent directory extension new extension [--commit]'
- quit()
-
-directory = sys.argv[1];
-if(not filemanagement.pathexist(directory)):
- print 'path ' + directory + ' does not exist'
- quit()
-
-old_extension = sys.argv[2]
-if(not stringmanipulation.isextension(old_extension)):
- print old_extension + ' is not a valid extension'
- quit()
-
-new_extension = sys.argv[3]
-if(not stringmanipulation.isextension(new_extension)):
- print new_extension + ' is not a valid extension'
- quit()
-
-if((len(sys.argv) == 5) and (sys.argv[4] != '--commit')):
- print 'parameters are: parent directory extension new extension [--commit]'
- quit()
-
-commit = False
-if(len(sys.argv) == 5):
- commit = True
-
-files_to_integrate = filemanagement.listallfilesinfolder(directory,\
- old_extension)
-
-if(commit):
- p4commands.checkoutallfiles()
-for index in range(len(files_to_integrate)):
- if(commit):
- print (100*index)/len(files_to_integrate)
- path_dir = files_to_integrate[index][0]
- filename = files_to_integrate[index][1]
- is_ignore = False
- for ignore_names in ignore_these:
- if(filename == ignore_names):
- is_ignore = True
- break
- if(is_ignore):
- continue
-
- new_file_name = ''
- is_exception = False
- for exception_name,exception_name_new in exceptions:
- if(filename == exception_name):
- is_exception = True
- new_file_name = exception_name_new
- break
-
- if(not is_exception):
- new_file_name = filename
-
- new_file_name = stringmanipulation.removeallprefix(new_file_name,\
- 'gips')
- new_file_name = stringmanipulation.removealloccurances(new_file_name,\
- 'module')
- new_file_name = stringmanipulation.changeextension(new_file_name,\
- old_extension,\
- new_extension)
- new_file_name = stringmanipulation.fixabbreviations( new_file_name )
- new_file_name = stringmanipulation.lowercasewithunderscore(new_file_name)
- if(not commit):
- print 'File ' + filename + ' will be replaced with ' + new_file_name
- continue
- full_new_file_name = path_dir + new_file_name
- full_old_file_name = path_dir + filename
- if(full_new_file_name != full_old_file_name):
- p4commands.integratefile(full_old_file_name,full_new_file_name)
- else:
- print 'skipping ' + new_file_name + ' due to no change'
- for extension in extensions:
- print 'replacing ' + filename
- if (extension == ".gyp"):
- filemanagement.replacestringinallsubfolders(
- filename,new_file_name,extension)
- else:
- filemanagement.replacestringinallsubfolders(
- '\"' + filename + '\"', '\"' + new_file_name + '\"', extension)
-if(commit):
- p4commands.revertunchangedfiles()
diff --git a/tools/refactoring/p4commands.py b/tools/refactoring/p4commands.py
deleted file mode 100644
index 71ac31b..0000000
--- a/tools/refactoring/p4commands.py
+++ /dev/null
@@ -1,31 +0,0 @@
-import os
-import filemanagement
-
-# checks out entire p4 repository
-def checkoutallfiles():
- os.system('p4 edit //depotGoogle/...')
- return
-
-# reverts all unchanged files, this is completely innoculus
-def revertunchangedfiles():
- os.system('p4 revert -a //depotGoogle/...')
- return
-
-def integratefile( old_name, new_name):
- if(old_name == new_name):
- return
- if(not filemanagement.fileexist(old_name)):
- return
- integrate_command = 'p4 integrate -o -f ' +\
- old_name +\
- ' ' +\
- new_name +\
- ' > p4summary.txt 2> error.txt'
- os.system(integrate_command)
- #print integrate_command
- delete_command = 'p4 delete -c default ' +\
- old_name +\
- ' > p4summary.txt 2> error.txt'
- os.system(delete_command)
- #print delete_command
- return
diff --git a/tools/refactoring/removetrace.py b/tools/refactoring/removetrace.py
deleted file mode 100644
index 43c622d..0000000
--- a/tools/refactoring/removetrace.py
+++ /dev/null
@@ -1,161 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-# NOTE: This is a hack which disobeys a number of conventions and best
-# practices. It's here just to be easily shared. If it's to remain in the
-# repository it should be refactored.
-
-#!/usr/bin/env python
-
-import stringmanipulation
-import filemanagement
-import sys
-
-trace_remove_key_word = 'kTraceModuleCall'
-
-if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
- print 'parameters are: parent directory [--commit]'
- quit()
-
-if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
- print 'parameters are: parent directory [--commit]'
- quit()
-
-commit = (len(sys.argv) == 3)
-
-directory = sys.argv[1];
-occurances = []
-
-trace_identifier = 'WEBRTC_TRACE('
-extensions = ['.h','.cc','.c','.cpp']
-files_to_fix = []
-for extension in extensions:
- files_to_fix.extend(filemanagement.listallfilesinfolder(directory,\
- extension))
-
-# This function identifies the begining of a trace statement
-def istracebegining(line):
- return stringmanipulation.issubstring(line, trace_identifier) != -1
-
-def endofstatement(line):
- return stringmanipulation.issubstring(line, ';') != -1
-
-def removekeywordfound(line):
- return stringmanipulation.issubstring(line, trace_remove_key_word) != -1
-
-# Used to store temporary result before flushing to real file when finished
-def temporaryfilename():
- return 'deleteme.txt'
-
-
-def find_occurances(path, file_name):
- full_filename = path + file_name
- file_handle = open(full_filename,'r')
- line_is_trace = False
- last_trace_line = -1
- for line_nr, line in enumerate(file_handle):
- if(istracebegining(line)):
- line_is_trace = True;
- last_trace_line = line_nr
-
- if(line_is_trace):
- if(removekeywordfound(line)):
- occurances.append(last_trace_line)
-
- if(endofstatement(line)):
- line_is_trace = False;
-
-def remove_occurances(path, file_name):
- full_file_name = path + file_name
- if (not filemanagement.fileexist(full_file_name)):
- print 'File ' + full_file_name + ' is not found.'
- print 'Should not happen! Ever!'
- quit()
-
- full_temporary_file_name = path + temporaryfilename()
- temporary_file = open(full_temporary_file_name,'w')
- original_file = open(full_file_name,'r')
- next_occurance_id = 0;
- removing_statement = False
- if(len(occurances) == next_occurance_id):
- return
- next_occurance = occurances[next_occurance_id]
- next_occurance_id += 1
- for line_nr, line in enumerate(original_file):
- if(line_nr == next_occurance):
- removing_statement = True
- if(len(occurances) == next_occurance_id):
- next_occurance_id = -1
- else:
- next_occurance = occurances[next_occurance_id]
- next_occurance_id += 1
-
- if (not removing_statement):
- temporary_file.writelines(line)
-
- if(endofstatement(line)):
- removing_statement = False;
-
- temporary_file.close()
- original_file.close()
- filemanagement.copyfile(full_file_name,full_temporary_file_name)
- filemanagement.deletefile(full_temporary_file_name)
-
-def nextoccurance():
- if (len(occurances) == 0):
- return -1
- return_value = occurances[0]
- occurances = occurances[1:len(occurances)]
- return return_value
-
-def would_be_removed_occurances(path, file_name):
- full_file_name = path + file_name
- if (not filemanagement.fileexist(full_file_name)):
- print 'File ' + full_file_name + ' is not found.'
- print 'Should not happen! Ever!'
- quit()
-
- original_file = open(full_file_name,'r')
- removing_statement = False
- next_occurance_id = 0;
- if(len(occurances) == next_occurance_id):
- return
- next_occurance = occurances[next_occurance_id]
- next_occurance_id += 1
- for line_nr, line in enumerate(original_file):
- if(line_nr == next_occurance):
- removing_statement = True
- if(len(occurances) == next_occurance_id):
- return
- next_occurance = occurances[next_occurance_id]
- next_occurance_id += 1
-
- if (removing_statement):
- print line_nr
-
- if(endofstatement(line)):
- removing_statement = False;
- if(next_occurance == -1):
- break
- original_file.close()
-
-for index in range(len(files_to_fix)):
- if(commit):
- print (100*index)/len(files_to_fix)
-
- path_dir = files_to_fix[index][0]
- filename = files_to_fix[index][1]
-
- #print path_dir + filename
- occurances = []
- find_occurances(path_dir, filename)
-
- if(not commit):
- would_be_removed_occurances(path_dir, filename)
- continue
- remove_occurances(path_dir, filename)
diff --git a/tools/refactoring/stringmanipulation.py b/tools/refactoring/stringmanipulation.py
deleted file mode 100644
index 0d9e0ff..0000000
--- a/tools/refactoring/stringmanipulation.py
+++ /dev/null
@@ -1,303 +0,0 @@
-import string
-
-# returns tuple, [success,updated_string] where the updated string has
-# has one less (the first) occurance of match string
-def removefirstoccurance( remove_string, match_string ):
- lowercase_string = remove_string.lower()
- lowercase_match_string = match_string.lower()
- lowest_index = lowercase_string.find(lowercase_match_string)
- if(lowest_index == -1):
- return [False,remove_string]
- past_match_index = lowest_index + len(lowercase_match_string)
- highest_index = len(remove_string)
- remove_string = remove_string[0:lowest_index] + remove_string[past_match_index: highest_index]
- return [True,remove_string]
-
-# returns a string with all occurances of match_string removed
-def removealloccurances( remove_string, match_string ):
- return_value = [True, remove_string]
- while(return_value[0]):
- return_value = removefirstoccurance(return_value[1],match_string)
- return return_value[1]
-
-# removes an occurance of match_string only if it's first in the string
-# returns tuple [succes, new_string]
-def removeprefix( remove_string, match_string ):
- lowercase_string = remove_string.lower()
- lowercase_match_string = match_string.lower()
- lowest_index = lowercase_string.find(lowercase_match_string)
- if(lowest_index == -1):
- return [False,remove_string]
- if(lowest_index != 0):
- return [False,remove_string]
- past_match_index = lowest_index + len(lowercase_match_string)
- highest_index = len(remove_string)
- remove_string = remove_string[0:lowest_index] + remove_string[past_match_index: highest_index]
-# print lowest_index
-# print past_match_index
- return [True,remove_string]
-
-# removes multiple occurances of match string as long as they are first in
-# the string
-def removeallprefix( remove_string, match_string ):
- return_value = [True, remove_string]
- while(return_value[0]):
- return_value = removeprefix(return_value[1],match_string)
- return return_value[1]
-
-# returns true if extensionstring is a correct extension
-def isextension( extensionstring ):
- if(len(extensionstring) < 2):
- return False
- if(extensionstring[0] != '.'):
- return False
- if(extensionstring[1:len(extensionstring)-1].find('.') != -1):
- return False
- return True
-
-# returns the index of start of the last occurance of match_string
-def findlastoccurance( original_string, match_string ):
- search_index = original_string.find(match_string)
- found_index = search_index
- last_index = len(original_string) - 1
- while((search_index != -1) and (search_index < last_index)):
- search_index = original_string[search_index+1:last_index].find(match_string)
- if(search_index != -1):
- found_index = search_index
- return found_index
-
-# changes extension from original_extension to new_extension
-def changeextension( original_string, original_extension, new_extension):
- if(not isextension(original_extension)):
- return original_string
- if(not isextension(new_extension)):
- return original_string
- index = findlastoccurance(original_string, original_extension)
- if(index == -1):
- return original_string
- return_value = original_string[0:index] + new_extension
- return return_value
-
-# wanted to do this with str.find however didnt seem to work so do it manually
-# returns the index of the first capital letter
-def findfirstcapitalletter( original_string ):
- for index in range(len(original_string)):
- if(original_string[index].lower() != original_string[index]):
- return index
- return -1
-
-
-# replaces capital letters with underscore and lower case letter (except very
-# first
-def lowercasewithunderscore( original_string ):
-# ignore the first letter since there should be no underscore in front of it
- if(len(original_string) < 2):
- return original_string
- return_value = original_string[1:len(original_string)]
- index = findfirstcapitalletter(return_value)
- while(index != -1):
- return_value = return_value[0:index] + \
- '_' + \
- return_value[index].lower() + \
- return_value[index+1:len(return_value)]
- index = findfirstcapitalletter(return_value)
- return_value = original_string[0].lower() + return_value
- return return_value
-
-# my table is a duplicate of strings
-def removeduplicates( my_table ):
- new_table = []
- for old_string1, new_string1 in my_table:
- found = 0
- for old_string2, new_string2 in new_table:
- if(old_string1 == old_string2):
- found += 1
- if(new_string1 == new_string2):
- if(new_string1 == ''):
- found += found
- else:
- found += 1
- if(found == 1):
- print 'missmatching set, terminating program'
- print old_string1
- print new_string1
- print old_string2
- print new_string2
- quit()
- if(found == 2):
- break
- if(found == 0):
- new_table.append([old_string1,new_string1])
- return new_table
-
-def removenochange( my_table ):
- new_table = []
- for old_string, new_string in my_table:
- if(old_string != new_string):
- new_table.append([old_string,new_string])
- return new_table
-
-# order table after size of the string (can be used to replace bigger strings
-# first which is useful since smaller strings can be inside the bigger string)
-# E.g. GIPS is a sub string of GIPSVE if we remove GIPS first GIPSVE will never
-# be removed. N is small so no need for fancy sort algorithm. Use selection sort
-def ordertablesizefirst( my_table ):
- for current_index in range(len(my_table)):
- biggest_string = 0
- biggest_string_index = -1
- for search_index in range(len(my_table)):
- if(search_index < current_index):
- continue
- length_of_string = len(my_table[search_index][0])
- if(length_of_string > biggest_string):
- biggest_string = length_of_string
- biggest_string_index = search_index
- if(biggest_string_index == -1):
- print 'sorting algorithm failed, program exit'
- quit()
- old_value = my_table[current_index]
- my_table[current_index] = my_table[biggest_string_index]
- my_table[biggest_string_index] = old_value
- return my_table
-
-# returns true if string 1 or 2 is a substring of the other, assuming neither
-# has whitespaces
-def issubstring( string1, string2 ):
- if(len(string1) == 0):
- return -1
- if(len(string2) == 0):
- return -1
- large_string = string1
- small_string = string2
- if(len(string1) < len(string2)):
- large_string = string2
- small_string = string1
-
- for index in range(len(large_string)):
- large_sub_string = large_string[index:index+len(small_string)].lower()
- if(large_sub_string ==\
- small_string.lower()):
- return index
- return -1
-
-#not_part_of_word_table = [' ','(',')','{','}',':','\t','*','&','/','[',']','.',',','\n']
-#def ispartofword( char ):
-# for item in not_part_of_word_table:
-# if(char == item):
-# return False
-# return True
-
-# must be numerical,_ or charachter
-def ispartofword( char ):
- if(char.isalpha()):
- return True
- if(char.isalnum()):
- return True
- if(char == '_'):
- return True
- return False
-
-# returns the index of the first letter in the word that the current_index
-# is pointing to and the size of the word
-def getword( line, current_index):
- if(current_index < 0):
- return []
- line = line.rstrip()
- if(len(line) <= current_index):
- return []
- if(line[current_index] == ' '):
- return []
- start_pos = current_index
- while start_pos >= 0:
- if(not ispartofword(line[start_pos])):
- start_pos += 1
- break
- start_pos -= 1
- if(start_pos == -1):
- start_pos = 0
- end_pos = current_index
- while end_pos < len(line):
- if(not ispartofword(line[end_pos])):
- break
- end_pos += 1
- return [start_pos,end_pos - start_pos]
-
-# my table is a tuple [string1,string2] complement_to_table is just a list
-# of strings to compare to string1
-def complement( my_table, complement_to_table ):
- new_table = []
- for index in range(len(my_table)):
- found = False;
- for compare_string in complement_to_table:
- if(my_table[index][0].lower() == compare_string.lower()):
- found = True
- if(not found):
- new_table.append(my_table[index])
- return new_table
-
-def removestringfromhead( line, remove_string):
- for index in range(len(line)):
- if(line[index:index+len(remove_string)] != remove_string):
- return line[index:index+len(line)]
- return ''
-
-def removeccomment( line ):
- comment_string = '//'
- for index in range(len(line)):
- if(line[index:index+len(comment_string)] == comment_string):
- return line[0:index]
- return line
-
-def whitespacestoonespace( line ):
- return ' '.join(line.split())
-
-def fixabbreviations( original_string ):
- previouswascapital = (original_string[0].upper() == original_string[0])
- new_string = ''
- for index in range(len(original_string)):
- if(index == 0):
- new_string += original_string[index]
- continue
- if(original_string[index] == '_'):
- new_string += original_string[index]
- previouswascapital = False
- continue
- if(original_string[index].isdigit()):
- new_string += original_string[index]
- previouswascapital = False
- continue
- currentiscapital = (original_string[index].upper() == original_string[index])
- letter_to_add = original_string[index]
- if(previouswascapital and currentiscapital):
- letter_to_add = letter_to_add.lower()
- if(previouswascapital and (not currentiscapital)):
- old_letter = new_string[len(new_string)-1]
- new_string = new_string[0:len(new_string)-1]
- new_string += old_letter.upper()
- previouswascapital = currentiscapital
- new_string += letter_to_add
- return new_string
-
-def replaceoccurances(old_string, replace_string, replace_with_string):
- if (len(replace_string) == 0):
- return old_string
- if (len(old_string) < len(replace_string)):
- return old_string
- # Simple implementation, could proably be done smarter
- new_string = ''
- for index in range(len(old_string)):
- #print new_string
- if(len(replace_string) > (len(old_string) - index)):
- new_string += old_string[index:index + len(old_string)]
- break
- match = (len(replace_string) > 0)
- for replace_index in range(len(replace_string)):
- if (replace_string[replace_index] != old_string[index + replace_index]):
- match = False
- break
- if (match):
- new_string += replace_with_string
- index =+ len(replace_string)
- else:
- new_string += old_string[index]
- return new_string
diff --git a/tools/refactoring/trim.py b/tools/refactoring/trim.py
deleted file mode 100644
index 5539f5f..0000000
--- a/tools/refactoring/trim.py
+++ /dev/null
@@ -1,29 +0,0 @@
-#!/usr/bin/env python
-
-import sys
-import fileinput
-
-# Defaults
-TABSIZE = 4
-
-usage = """
-Replaces all TAB characters with %(TABSIZE)d space characters.
-In addition, all trailing space characters are removed.
-usage: trim file ...
-file ... : files are changed in place without taking any backup.
-""" % vars()
-
-def main():
-
- if len(sys.argv) == 1:
- sys.stderr.write(usage)
- sys.exit(2)
-
- # Iterate over the lines of all files listed in sys.argv[1:]
- for line in fileinput.input(sys.argv[1:], inplace=True):
- line = line.replace('\t',' '*TABSIZE); # replace TABs
- line = line.rstrip(None) # remove trailing whitespaces
- print line # modify the file
-
-if __name__ == '__main__':
- main()
diff --git a/tools/refactoring/trimall.py b/tools/refactoring/trimall.py
deleted file mode 100644
index 7a1c458..0000000
--- a/tools/refactoring/trimall.py
+++ /dev/null
@@ -1,59 +0,0 @@
-#!/usr/bin/env python
-
-import sys
-import fileinput
-import filemanagement
-import p4commands
-
-# Defaults
-TABSIZE = 4
-
-extensions = ['.h','.cc','.c','.cpp']
-
-ignore_these = ['my_ignore_header.h']
-
-usage = """
-Replaces all TAB characters with %(TABSIZE)d space characters.
-In addition, all trailing space characters are removed.
-usage: trim directory
-""" % vars()
-
-if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
- sys.stderr.write(usage)
- sys.exit(2)
-
-directory = sys.argv[1];
-if(not filemanagement.pathexist(directory)):
- sys.stderr.write(usage)
- sys.exit(2)
-
-if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
- sys.stderr.write(usage)
- sys.exit(2)
-
-commit = False
-if(len(sys.argv) == 3):
- commit = True
-
-files_to_fix = []
-for extension in extensions:
- files_to_fix.extend(filemanagement.listallfilesinfolder(directory,\
- extension))
-
-def main():
- if (commit):
- p4commands.checkoutallfiles()
- for path,file_name in files_to_fix:
- full_file_name = path + file_name
- if (not commit):
- print full_file_name + ' will be edited'
- continue
- for line in fileinput.input(full_file_name, inplace=True):
- line = line.replace('\t',' '*TABSIZE); # replace TABs
- line = line.rstrip(None) # remove trailing whitespaces
- print line # modify the file
- if (commit):
- p4commands.revertunchangedfiles()
-
-if __name__ == '__main__':
- main()
diff --git a/tools/refactoring/webrtc_reformat.py b/tools/refactoring/webrtc_reformat.py
deleted file mode 100755
index 269d1c3..0000000
--- a/tools/refactoring/webrtc_reformat.py
+++ /dev/null
@@ -1,212 +0,0 @@
-#!/usr/bin/python
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""WebRTC reformat script.
-
-This script is used to reformat WebRTC code from the old code style to Google
-C++ code style. This script does not indent code; use clang-reformat-chrome.py
-as described in go/webrtc/engineering/reformatting-gips---google.
-"""
-
-__author__ = 'mflodman@webrtc.org (Magnus Flodman)'
-
-import fnmatch
-import os
-import re
-import subprocess
-import sys
-
-
-def LowerWord(obj):
- """Helper for DeCamelCase."""
- optional_last_letters = obj.group(3) or ''
- return obj.group(1) + '_' + obj.group(2).lower() + optional_last_letters
-
-
-def DeCamelCase(text):
- """De-camelize variable names.
-
- This function will look at any stringLikeThis and format it in steps. The
- sequence will be stringLikeThis -> string_likeThis -> string_like_this.
- """
- possible_tokens_before_vars = '[ _*\(\&\!\[]'
- pattern = re.compile(r'(?<=' + possible_tokens_before_vars + ')' +
- # Match some lower-case characters
- '([a-z]+)' +
- # Don't match kFoo, !kFoo, [kFoo], etc
- '(?<!' + possible_tokens_before_vars + 'k)' +
- # Match some upper-case characters
- '([A-Z]+)([a-z])?')
- while re.search(pattern, text):
- text = re.sub(pattern, LowerWord, text)
- return text
-
-
-def MoveUnderScore(text):
- """Moves the underscore from beginning of variable name to the end."""
- # TODO(mflodman) Replace \1 with ?-expression.
- # We don't want to change macros and #defines though, so don't do anything
- # if the first character is uppercase (normal variables shouldn't have that).
- pattern = r'([ \*\!\&\(\[\]])_(?!_)(?![A-Z])(\w+)'
- return re.sub(pattern, r'\1\2_', text)
-
-
-def PostfixToPrefixInForLoops(text):
- """Converts x++ to ++x in the increment part of a for loop."""
- pattern = r'(for \(.*;.*;) (\w+)\+\+\)'
- return re.sub(pattern, r'\1++\2)', text)
-
-
-def SortIncludeHeaders(text, filename):
- """Sorts all include headers in alphabetic order.
-
- The file's own header goes first, followed by system headers and then
- project headers. This function will exit if we detect any fancy #ifdef logic
- among the includes - that's a lot harder to sort.
-
- Args:
- text: The file text.
- filename: The file we are reformatting.
-
- Returns:
- The text with includes sorted.
- """
- # Get all includes in file.
- include_pattern = re.compile('#include.+\n')
- includes = re.findall(include_pattern, text)
-
- # Sort system headers and project headers separately.
- sys_includes = []
- project_includes = []
- self_include = ''
- sys_pattern = re.compile('#include <')
- h_filename, _ = os.path.splitext(os.path.basename(filename))
-
- for item in includes:
- if re.search(h_filename + '\.', item):
- self_include = item
- elif re.search(sys_pattern, item):
- sys_includes.append(item)
- else:
- project_includes.append(item)
-
- sys_includes = sorted(sys_includes)
- project_includes = sorted(project_includes)
- headers = (self_include + '\n' + ''.join(sys_includes) + '\n' +
- ''.join(project_includes))
-
- # Replace existing headers with the sorted string.
- text_no_hdrs = re.sub(include_pattern, r'???', text)
-
- # Insert sorted headers unless we detect #ifdefs right next to the headers.
- if re.search(r'(#ifdef|#ifndef|#if).*\s*\?{3,}\s*#endif', text_no_hdrs):
- print 'WARNING: Include headers not sorted in ' + filename
- return text
-
- return_text = re.sub(r'\?{3,}', headers, text_no_hdrs, 1)
- if re.search(r'\?{3,}', text_no_hdrs):
- # Remove possible remaining ???.
- return_text = re.sub(r'\?{3,}', r'', return_text)
-
- return return_text
-
-
-def AddPath(match):
- """Helper for adding file path for WebRTC header files, ignoring other."""
- file_to_examine = match.group(1) + '.h'
- # TODO(mflodman) Use current directory and find webrtc/.
- for path, _, files in os.walk('./webrtc'):
- for filename in files:
- if fnmatch.fnmatch(filename, file_to_examine):
- path_name = os.path.join(path, filename).replace('./', '')
- return '#include "%s"\n' % path_name
-
- # No path found, return original string.
- return '#include "'+ file_to_examine + '"\n'
-
-
-def AddHeaderPath(text):
- """Add path to all included header files that have no path yet."""
- headers = re.compile('#include "(.+).h"\n')
- return re.sub(headers, AddPath, text)
-
-
-def AddWebrtcToOldSrcRelativePath(match):
- file_to_examine = match.group(1) + '.h'
- path, filename = os.path.split(file_to_examine)
- dirs_in_webrtc = [name for name in os.listdir('./webrtc')
- if os.path.isdir(os.path.join('./webrtc', name))]
- for dir_in_webrtc in dirs_in_webrtc:
- if path.startswith(dir_in_webrtc):
- return '#include "%s"\n' % os.path.join('webrtc', path, filename)
- return '#include "%s"\n' % file_to_examine
-
-def AddWebrtcPrefixToOldSrcRelativePaths(text):
- """For all paths starting with for instance video_engine, add webrtc/."""
- headers = re.compile('#include "(.+).h"\n')
- return re.sub(headers, AddWebrtcToOldSrcRelativePath, text)
-
-
-def FixIncludeGuards(text, file_name):
- """Change include guard according to the stantard."""
- # Remove a possible webrtc/ from the path.
- file_name = re.sub(r'(webrtc\/)(.+)', r'\2', file_name)
- new_guard = 'WEBRTC_' + file_name
- new_guard = new_guard.upper()
- new_guard = re.sub(r'([/\.])', r'_', new_guard)
- new_guard += '_'
-
- text = re.sub(r'#ifndef WEBRTC_.+\n', r'#ifndef ' + new_guard + '\n', text, 1)
- text = re.sub(r'#define WEBRTC_.+\n', r'#define ' + new_guard + '\n', text, 1)
- text = re.sub(r'#endif *\/\/ *WEBRTC_.+\n', r'#endif // ' + new_guard + '\n',
- text, 1)
-
- return text
-
-
-def SaveFile(filename, text):
- os.remove(filename)
- f = open(filename, 'w')
- f.write(text)
- f.close()
-
-
-def main():
- args = sys.argv[1:]
- if not args:
- print 'Usage: %s <filename>' % sys.argv[0]
- sys.exit(1)
-
- for filename in args:
- f = open(filename)
- text = f.read()
- f.close()
-
- text = DeCamelCase(text)
- text = MoveUnderScore(text)
- text = PostfixToPrefixInForLoops(text)
- text = AddHeaderPath(text)
- text = AddWebrtcPrefixToOldSrcRelativePaths(text)
- text = SortIncludeHeaders(text, filename)
-
- # Remove the original file and re-create it with the reformatted content.
- SaveFile(filename, text)
-
- if filename.endswith('.h'):
- f = open(filename)
- text = f.read()
- f.close()
- text = FixIncludeGuards(text, filename)
- SaveFile(filename, text)
-
- print filename + ' done.'
-
-
-if __name__ == '__main__':
- main()
diff --git a/tools/sslroots/generate_sslroots.py b/tools/sslroots/generate_sslroots.py
new file mode 100644
index 0000000..65751f1
--- /dev/null
+++ b/tools/sslroots/generate_sslroots.py
@@ -0,0 +1,190 @@
+# -*- coding:utf-8 -*-
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+
+"""This is a tool to transform a crt file into a C/C++ header.
+
+Usage:
+generate_sslroots.py cert_file.crt [--verbose | -v] [--full_cert | -f]
+
+Arguments:
+ -v Print output while running.
+ -f Add public key and certificate name. Default is to skip and reduce
+ generated file size.
+"""
+
+import commands
+from optparse import OptionParser
+import os
+import re
+import string
+
+_GENERATED_FILE = 'sslroots.h'
+_PREFIX = '__generated__'
+_EXTENSION = '.crt'
+_SUBJECT_NAME_ARRAY = 'subject_name'
+_SUBJECT_NAME_VARIABLE = 'SubjectName'
+_PUBLIC_KEY_ARRAY = 'public_key'
+_PUBLIC_KEY_VARIABLE = 'PublicKey'
+_CERTIFICATE_ARRAY = 'certificate'
+_CERTIFICATE_VARIABLE = 'Certificate'
+_CERTIFICATE_SIZE_VARIABLE = 'CertificateSize'
+_INT_TYPE = 'size_t'
+_CHAR_TYPE = 'const unsigned char*'
+_VERBOSE = 'verbose'
+
+
+def main():
+ """The main entrypoint."""
+ parser = OptionParser('usage %prog FILE')
+ parser.add_option('-v', '--verbose', dest='verbose', action='store_true')
+ parser.add_option('-f', '--full_cert', dest='full_cert', action='store_true')
+ options, args = parser.parse_args()
+ if len(args) < 1:
+ parser.error('No crt file specified.')
+ return
+ root_dir = _SplitCrt(args[0], options)
+ _GenCFiles(root_dir, options)
+ _Cleanup(root_dir)
+
+
+def _SplitCrt(source_file, options):
+ sub_file_blocks = []
+ label_name = ''
+ root_dir = os.path.dirname(os.path.abspath(source_file)) + '/'
+ _PrintOutput(root_dir, options)
+ f = open(source_file)
+ for line in f:
+ if line.startswith('# Label: '):
+ sub_file_blocks.append(line)
+ label = re.search(r'\".*\"', line)
+ temp_label = label.group(0)
+ end = len(temp_label)-1
+ label_name = _SafeName(temp_label[1:end])
+ elif line.startswith('-----END CERTIFICATE-----'):
+ sub_file_blocks.append(line)
+ new_file_name = root_dir + _PREFIX + label_name + _EXTENSION
+ _PrintOutput('Generating: ' + new_file_name, options)
+ new_file = open(new_file_name, 'w')
+ for out_line in sub_file_blocks:
+ new_file.write(out_line)
+ new_file.close()
+ sub_file_blocks = []
+ else:
+ sub_file_blocks.append(line)
+ f.close()
+ return root_dir
+
+
+def _GenCFiles(root_dir, options):
+ output_header_file = open(root_dir + _GENERATED_FILE, 'w')
+ output_header_file.write(_CreateOutputHeader())
+ if options.full_cert:
+ subject_name_list = _CreateArraySectionHeader(_SUBJECT_NAME_VARIABLE,
+ _CHAR_TYPE, options)
+ public_key_list = _CreateArraySectionHeader(_PUBLIC_KEY_VARIABLE,
+ _CHAR_TYPE, options)
+ certificate_list = _CreateArraySectionHeader(_CERTIFICATE_VARIABLE,
+ _CHAR_TYPE, options)
+ certificate_size_list = _CreateArraySectionHeader(_CERTIFICATE_SIZE_VARIABLE,
+ _INT_TYPE, options)
+
+ for _, _, files in os.walk(root_dir):
+ for current_file in files:
+ if current_file.startswith(_PREFIX):
+ prefix_length = len(_PREFIX)
+ length = len(current_file) - len(_EXTENSION)
+ label = current_file[prefix_length:length]
+ filtered_output, cert_size = _CreateCertSection(root_dir, current_file,
+ label, options)
+ output_header_file.write(filtered_output + '\n\n\n')
+ if options.full_cert:
+ subject_name_list += _AddLabelToArray(label, _SUBJECT_NAME_ARRAY)
+ public_key_list += _AddLabelToArray(label, _PUBLIC_KEY_ARRAY)
+ certificate_list += _AddLabelToArray(label, _CERTIFICATE_ARRAY)
+ certificate_size_list += (' %s,\n') %(cert_size)
+
+ if options.full_cert:
+ subject_name_list += _CreateArraySectionFooter()
+ output_header_file.write(subject_name_list)
+ public_key_list += _CreateArraySectionFooter()
+ output_header_file.write(public_key_list)
+ certificate_list += _CreateArraySectionFooter()
+ output_header_file.write(certificate_list)
+ certificate_size_list += _CreateArraySectionFooter()
+ output_header_file.write(certificate_size_list)
+ output_header_file.close()
+
+
+def _Cleanup(root_dir):
+ for f in os.listdir(root_dir):
+ if f.startswith(_PREFIX):
+ os.remove(root_dir + f)
+
+
+def _CreateCertSection(root_dir, source_file, label, options):
+ command = 'openssl x509 -in %s%s -noout -C' %(root_dir, source_file)
+ _PrintOutput(command, options)
+ output = commands.getstatusoutput(command)[1]
+ renamed_output = output.replace('unsigned char XXX_',
+ 'const unsigned char ' + label + '_')
+ filtered_output = ''
+ cert_block = '^const unsigned char.*?};$'
+ prog = re.compile(cert_block, re.IGNORECASE | re.MULTILINE | re.DOTALL)
+ if not options.full_cert:
+ filtered_output = prog.sub('', renamed_output, count=2)
+ else:
+ filtered_output = renamed_output
+
+ cert_size_block = r'\d\d\d+'
+ prog2 = re.compile(cert_size_block, re.MULTILINE | re.VERBOSE)
+ result = prog2.findall(renamed_output)
+ cert_size = result[len(result) - 1]
+
+ return filtered_output, cert_size
+
+
+def _CreateOutputHeader():
+ output = ('// This file is the root certificates in C form that are needed to'
+ ' connect to\n// Google.\n\n'
+ '// It was generated with the following command line:\n'
+ '// > python tools/certs/generate_sslroots.py'
+ '\n// https://pki.google.com/roots.pem\n\n')
+ return output
+
+
+def _CreateArraySectionHeader(type_name, type_type, options):
+ output = ('const %s kSSLCert%sList[] = {\n') %(type_type, type_name)
+ _PrintOutput(output, options)
+ return output
+
+
+def _AddLabelToArray(label, type_name):
+ return ' %s_%s,\n' %(label, type_name)
+
+
+def _CreateArraySectionFooter():
+ return '};\n\n'
+
+
+def _SafeName(original_file_name):
+ bad_chars = ' -./\\()áéíőú'
+ replacement_chars = ''
+ for _ in bad_chars:
+ replacement_chars += '_'
+ translation_table = string.maketrans(bad_chars, replacement_chars)
+ return original_file_name.translate(translation_table)
+
+
+def _PrintOutput(output, options):
+ if options.verbose:
+ print output
+
+if __name__ == '__main__':
+ main()
diff --git a/tools/valgrind-webrtc/drmemory/suppressions.txt b/tools/valgrind-webrtc/drmemory/suppressions.txt
index 2addea5..1b0626c 100644
--- a/tools/valgrind-webrtc/drmemory/suppressions.txt
+++ b/tools/valgrind-webrtc/drmemory/suppressions.txt
@@ -2,22 +2,6 @@
# It acts as a place holder for future additions for WebRTC.
# It must exist for the Python wrapper script to work properly.
-INVALID HEAP ARGUMENT
-name=https://code.google.com/p/webrtc/issues/detail?id=2321 (1)
-drmemorylib.dll!replace_operator_delete_array
-*!webrtc::scoped_array<short>::~scoped_array<short>
-*!webrtc::PushResampler::~PushResampler
-...
-*!testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void>
-
-INVALID HEAP ARGUMENT
-name=https://code.google.com/p/webrtc/issues/detail?id=2321 (2)
-drmemorylib.dll!replace_operator_delete_array
-*!webrtc::scoped_array<float>::~scoped_array<float>
-*!webrtc::PushSincResampler::~PushSincResampler
-...
-*!testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void>
-
GDI USAGE ERROR
name=https://code.google.com/p/webrtc/issues/detail?id=2323 (1)
system call NtGdiDeleteObjectApp
@@ -90,25 +74,6 @@
*!webrtc::test::UdpSocketManager_AddAndRemoveSocketDoesNotLeakMemory_Test::TestBody
*!testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void>
-INVALID HEAP ARGUMENT
-name=https://code.google.com/p/webrtc/issues/detail?id=2515 (1)
-drmemorylib.dll!replace_operator_delete_nothrow
-*!webrtc::scoped_array<short>::~scoped_array<short>
-*!webrtc::NetEqImpl::~NetEqImpl
-*!webrtc::NetEqImpl::`scalar deleting destructor'
-...
-*!testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void>
-
-INVALID HEAP ARGUMENT
-name=https://code.google.com/p/webrtc/issues/detail?id=2515 (2)
-drmemorylib.dll!replace_operator_delete_nothrow
-*!webrtc::scoped_array<short>::reset
-*!webrtc::NetEqImpl::SetSampleRateAndChannels
-*!webrtc::NetEqImpl::InsertPacketInternal
-*!webrtc::NetEqImpl::InsertPacket
-...
-*!testing::internal::HandleSehExceptionsInMethodIfSupported<testing::Test,void>
-
UNINITIALIZED READ
name=https://code.google.com/p/webrtc/issues/detail?id=2516
system call NtUserGetThreadDesktop parameter value #1
diff --git a/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-drmemory_win32.txt b/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-drmemory_win32.txt
index d41c231..d041dbd 100644
--- a/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-drmemory_win32.txt
+++ b/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-drmemory_win32.txt
@@ -1,7 +1,7 @@
# Flakily fails or crashes on Dr Memory Full.
# https://code.google.com/p/webrtc/issues/detail?id=3158
DtmfSenderTest.*
-JsepPeerConnectionP2PTestClient.*
+P2PTestConductor.*
PeerConnectionEndToEndTest.*
PeerConnectionInterfaceTest.*
# Issue 3453
diff --git a/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-memcheck.txt b/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-memcheck.txt
index 40974a2..9cf29b8 100644
--- a/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-memcheck.txt
+++ b/tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-memcheck.txt
@@ -1,6 +1,6 @@
# Tests that are failing when run under memcheck.
# https://code.google.com/p/webrtc/issues/detail?id=4387
DtmfSenderTest.*
-JsepPeerConnectionP2PTestClient.*
+P2PTestConductor.*
PeerConnectionEndToEndTest.*
PeerConnectionInterfaceTest.*
diff --git a/tools/valgrind-webrtc/gtest_exclude/modules_tests.gtest-drmemory.txt b/tools/valgrind-webrtc/gtest_exclude/modules_tests.gtest-drmemory.txt
index e4f3ae2..1642e3c 100644
--- a/tools/valgrind-webrtc/gtest_exclude/modules_tests.gtest-drmemory.txt
+++ b/tools/valgrind-webrtc/gtest_exclude/modules_tests.gtest-drmemory.txt
@@ -8,6 +8,5 @@
AudioCodingModuleTest.TestVADDTX*
AudioCodingModuleTest.TestOpus*
FecTest.FecTest
-TestVp8Impl.BaseUnitTest
VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP8
VideoProcessorIntegrationTest.*VP9
diff --git a/tools/valgrind-webrtc/gtest_exclude/rtc_unittests.gtest-drmemory.txt b/tools/valgrind-webrtc/gtest_exclude/rtc_unittests.gtest-drmemory.txt
index 092b785..8880627 100644
--- a/tools/valgrind-webrtc/gtest_exclude/rtc_unittests.gtest-drmemory.txt
+++ b/tools/valgrind-webrtc/gtest_exclude/rtc_unittests.gtest-drmemory.txt
@@ -5,3 +5,6 @@
PortTest.*
PseudoTcpTest.TestSendBothUseLargeWindowScale
SharedExclusiveLockTest.TestSharedShared
+# Fails on Dr Memory Light.
+# https://bugs.chromium.org/p/webrtc/issues/detail?id=5199
+ThreadTest.ThreeThreadsInvoke
diff --git a/tools/valgrind-webrtc/gtest_exclude/system_wrappers_unittests.gtest-drmemory_win32.txt b/tools/valgrind-webrtc/gtest_exclude/system_wrappers_unittests.gtest-drmemory_win32.txt
old mode 100755
new mode 100644
index 0bceb97..a3344c7
--- a/tools/valgrind-webrtc/gtest_exclude/system_wrappers_unittests.gtest-drmemory_win32.txt
+++ b/tools/valgrind-webrtc/gtest_exclude/system_wrappers_unittests.gtest-drmemory_win32.txt
@@ -1,3 +1,4 @@
-# https://code.google.com/p/webrtc/issues/detail?id=2330
-ClockTest.NtpTime
-
+# https://code.google.com/p/webrtc/issues/detail?id=2330
+ClockTest.NtpTime
+CritSectTest.ThreadWakesOnce
+CritSectTest.ThreadWakesTwice
diff --git a/tools/valgrind-webrtc/gtest_exclude/video_engine_tests.gtest-drmemory_win32.txt b/tools/valgrind-webrtc/gtest_exclude/video_engine_tests.gtest-drmemory_win32.txt
index 309fd40..c4d953a 100644
--- a/tools/valgrind-webrtc/gtest_exclude/video_engine_tests.gtest-drmemory_win32.txt
+++ b/tools/valgrind-webrtc/gtest_exclude/video_engine_tests.gtest-drmemory_win32.txt
@@ -3,14 +3,23 @@
EndToEndTest.CanSwitchToUseAllSsrcs
EndToEndTest.SendsAndReceivesMultipleStreams
EndToEndTest.ReceivesAndRetransmitsNack
-# https://code.google.com/p/webrtc/issues/detail?id=3471
-VideoSendStreamTest.RetransmitsNackOverRtxWithPacing
+EndToEndTest.ReceivesTransportFeedback
# Flaky: https://code.google.com/p/webrtc/issues/detail?id=3552
EndToEndTest.RestartingSendStreamPreservesRtpState
EndToEndTest.RestartingSendStreamPreservesRtpStatesWithRtx
EndToEndTest.SendsAndReceivesH264
EndToEndTest.SendsAndReceivesVP9
+EndToEndTest.TransportFeedbackNotConfigured
+EndToEndTest.TransportSeqNumOnAudioAndVideo
VideoSendStreamTest.CanReconfigureToUseStartBitrateAbovePreviousMax
VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly
+# https://code.google.com/p/webrtc/issues/detail?id=5417
+VideoSendStreamTest.Vp9NonFlexMode_2Tl2SLayers
+VideoSendStreamTest.Vp9NonFlexMode_3Tl2SLayers
+VideoSendStreamTest.Vp9FlexModeRefCount
# https://code.google.com/p/webrtc/issues/detail?id=4979
EndToEndTest.AssignsTransportSequenceNumbers
+# Flaky: https://bugs.chromium.org/p/webrtc/issues/detail?id=5225
+BitrateEstimatorTest.SwitchesToASTThenBackToTOFForVideo
+# https://bugs.chromium.org/p/webrtc/issues/detail?id=5312
+RtcEventLogTest.DropOldEvents
diff --git a/tools/valgrind-webrtc/gtest_exclude/video_engine_tests.gtest-memcheck.txt b/tools/valgrind-webrtc/gtest_exclude/video_engine_tests.gtest-memcheck.txt
index 57387dc..6faf218 100644
--- a/tools/valgrind-webrtc/gtest_exclude/video_engine_tests.gtest-memcheck.txt
+++ b/tools/valgrind-webrtc/gtest_exclude/video_engine_tests.gtest-memcheck.txt
@@ -6,3 +6,5 @@
# Flaky under memcheck (WebRTC issue 5134)
EndToEndTest.AssignsTransportSequenceNumbers
+# https://bugs.chromium.org/p/webrtc/issues/detail?id=5312
+RtcEventLogTest.DropOldEvents
diff --git a/tools/valgrind-webrtc/memcheck/suppressions.txt b/tools/valgrind-webrtc/memcheck/suppressions.txt
index 6291df2..9919522 100644
--- a/tools/valgrind-webrtc/memcheck/suppressions.txt
+++ b/tools/valgrind-webrtc/memcheck/suppressions.txt
@@ -232,159 +232,6 @@
}
{
- bug_332_1
- Memcheck:Uninitialized
- ...
- fun:_ZN6webrtc11VoEBaseImpl16NeedMorePlayDataEjhhjPvRj
- fun:_ZN6webrtc17AudioDeviceBuffer18RequestPlayoutDataEj
- fun:_ZN6webrtc21AudioDeviceLinuxPulse17PlayThreadProcessEv
- fun:_ZN6webrtc21AudioDeviceLinuxPulse14PlayThreadFuncEPv
- fun:_ZN6webrtc11ThreadPosix3RunEv
- fun:StartThread
-}
-
-{
- bug_332_2
- Memcheck:Unaddressable
- fun:memcpy@@GLIBC_2.14
- fun:_ZN6webrtc21AudioDeviceLinuxPulse16ReadRecordedDataEPKvm
- fun:_ZN6webrtc21AudioDeviceLinuxPulse16RecThreadProcessEv
- fun:_ZN6webrtc21AudioDeviceLinuxPulse13RecThreadFuncEPv
- fun:_ZN6webrtc11ThreadPosix3RunEv
- fun:StartThread
-}
-
-{
- bug_332_3
- Memcheck:Uninitialized
- fun:_ZN6webrtc11RTCPUtility21RTCPParseCommonHeaderEPKhS2_RNS0_16RTCPCommonHeaderE
- ...
- fun:_ZN6webrtc12RTCPReceiver10HandleSDESERNS_11RTCPUtility12RTCPParserV2E
- fun:_ZN6webrtc12RTCPReceiver18IncomingRTCPPacketERNS_8RTCPHelp21RTCPPacketInformationEPNS_11RTCPUtility12RTCPParserV2E
- fun:_ZN6webrtc17ModuleRtpRtcpImpl14IncomingPacketEPKht
- fun:_ZN6webrtc3voe7Channel18IncomingRTCPPacketEPKaiPKct
- fun:_ZN6webrtc16UdpTransportImpl20IncomingRTCPFunctionEPKaiPKNS_13SocketAddressE
- fun:_ZN6webrtc16UdpTransportImpl20IncomingRTCPCallbackEPvPKaiPKNS_13SocketAddressE
- fun:_ZN6webrtc14UdpSocketPosix11HasIncomingEv
- fun:_ZN6webrtc25UdpSocketManagerPosixImpl7ProcessEv
- fun:_ZN6webrtc25UdpSocketManagerPosixImpl3RunEPv
- fun:_ZN6webrtc11ThreadPosix3RunEv
- fun:StartThread
-}
-
-{
- bug_332_4
- Memcheck:Uninitialized
- ...
- fun:_ZN6webrtc3voe10AudioLevel12ComputeLevelERKNS_10AudioFrameE
- fun:_ZN6webrtc3voe11OutputMixer28DoOperationsOnCombinedSignalEv
- fun:_ZN6webrtc11VoEBaseImpl16NeedMorePlayDataEjhhjPvRj
- fun:_ZN6webrtc17AudioDeviceBuffer18RequestPlayoutDataEj
- fun:_ZN6webrtc21AudioDeviceLinuxPulse17PlayThreadProcessEv
- fun:_ZN6webrtc21AudioDeviceLinuxPulse14PlayThreadFuncEPv
- fun:_ZN6webrtc11ThreadPosix3RunEv
- fun:StartThread
-}
-
-{
- bug_332_5
- Memcheck:Uninitialized
- fun:WebRtcSpl_UpBy2ShortToInt
- fun:WebRtcSpl_Resample8khzTo22khz
- fun:_ZN6webrtc9Resampler4PushEPKsiPsiRi
- fun:_ZN6webrtc3voe16RemixAndResampleERKNS_10AudioFrameEPNS_9ResamplerEPS1_
- fun:_ZN6webrtc3voe11OutputMixer13GetMixedAudioEiiPNS_10AudioFrameE
- fun:_ZN6webrtc11VoEBaseImpl16NeedMorePlayDataEjhhjPvRj
- fun:_ZN6webrtc17AudioDeviceBuffer18RequestPlayoutDataEj
- fun:_ZN6webrtc21AudioDeviceLinuxPulse17PlayThreadProcessEv
- fun:_ZN6webrtc21AudioDeviceLinuxPulse14PlayThreadFuncEPv
- fun:_ZN6webrtc11ThreadPosix3RunEv
- fun:StartThread
-}
-
-{
- bug 332_6
- Memcheck:Param
- socketcall.sendto(msg)
- obj:*libpthread-*.so
- fun:_ZN6webrtc14UdpSocketPosix6SendToEPKaiRKNS_13SocketAddressE
- fun:_ZN6webrtc16UdpTransportImpl14SendRTCPPacketEiPKvi
- fun:_ZN6webrtc3voe7Channel14SendRTCPPacketEiPKvi
- ...
- fun:_ZN6webrtc17ModuleRtpRtcpImpl14IncomingPacketEPKht
- fun:_ZN6webrtc3voe7Channel17IncomingRTPPacketEPKaiPKct
- fun:_ZN6webrtc16UdpTransportImpl19IncomingRTPFunctionEPKaiPKNS_13SocketAddressE
- fun:_ZN6webrtc16UdpTransportImpl19IncomingRTPCallbackEPvPKaiPKNS_13SocketAddressE
- fun:_ZN6webrtc14UdpSocketPosix11HasIncomingEv
- fun:_ZN6webrtc25UdpSocketManagerPosixImpl7ProcessEv
- fun:_ZN6webrtc25UdpSocketManagerPosixImpl3RunEPv
- fun:_ZN6webrtc11ThreadPosix3RunEv
- fun:StartThread
-}
-
-
-{
- bug_332_7
- Memcheck:Param
- socketcall.sendto(msg)
- obj:*libpthread-*.so
- fun:_ZN6webrtc14UdpSocketPosix6SendToEPKaiRKNS_13SocketAddressE
- fun:_ZN6webrtc16UdpTransportImpl10SendPacketEiPKvi
- ...
- fun:_ZN6webrtc21AudioDeviceLinuxPulse16ReadRecordedDataEPKvm
- fun:_ZN6webrtc21AudioDeviceLinuxPulse16RecThreadProcessEv
- fun:_ZN6webrtc21AudioDeviceLinuxPulse13RecThreadFuncEPv
- fun:_ZN6webrtc11ThreadPosix3RunEv
- fun:StartThread
-}
-
-{
- bug_332_8
- Memcheck:Param
- socketcall.sendto(msg)
- obj:*libpthread-*.so
- fun:_ZN6webrtc14UdpSocketPosix6SendToEPKaiRKNS_13SocketAddressE
- fun:_ZN6webrtc16UdpTransportImpl14SendRTCPPacketEiPKvi
- fun:_ZN6webrtc3voe7Channel14SendRTCPPacketEiPKvi
- fun:_ZN6webrtc10RTCPSender13SendToNetworkEPKht
- fun:_ZN6webrtc10RTCPSender8SendRTCPEjiPKtbm
- fun:_ZN6webrtc17ModuleRtpRtcpImpl7ProcessEv
- fun:_ZN6webrtc17ProcessThreadImpl7ProcessEv
- fun:_ZN6webrtc17ProcessThreadImpl3RunEPv
- fun:_ZN6webrtc11ThreadPosix3RunEv
- fun:StartThread
-}
-
-{
- bug_332_9
- Memcheck:Uninitialized
- ...
- fun:_ZN6webrtc17AudioDeviceBuffer19DeliverRecordedDataEv
- fun:_ZN6webrtc21AudioDeviceLinuxPulse19ProcessRecordedDataEPajj
- fun:_ZN6webrtc21AudioDeviceLinuxPulse16ReadRecordedDataEPKvm
- fun:_ZN6webrtc21AudioDeviceLinuxPulse16RecThreadProcessEv
- fun:_ZN6webrtc21AudioDeviceLinuxPulse13RecThreadFuncEPv
- fun:_ZN6webrtc11ThreadPosix3RunEv
- fun:StartThread
-}
-
-{
- bug_332_10
- Memcheck:Uninitialized
- ...
- fun:_ZN6webrtc12RTCPReceiver18IncomingRTCPPacketERNS_8RTCPHelp21RTCPPacketInformationEPNS_11RTCPUtility12RTCPParserV2E
- fun:_ZN6webrtc17ModuleRtpRtcpImpl14IncomingPacketEPKht
- fun:_ZN6webrtc3voe7Channel18IncomingRTCPPacketEPKaiPKct
- fun:_ZN6webrtc16UdpTransportImpl20IncomingRTCPFunctionEPKaiPKNS_13SocketAddressE
- fun:_ZN6webrtc16UdpTransportImpl20IncomingRTCPCallbackEPvPKaiPKNS_13SocketAddressE
- fun:_ZN6webrtc14UdpSocketPosix11HasIncomingEv
- fun:_ZN6webrtc25UdpSocketManagerPosixImpl7ProcessEv
- fun:_ZN6webrtc25UdpSocketManagerPosixImpl3RunEPv
- fun:_ZN6webrtc11ThreadPosix3RunEv
- fun:StartThread
-}
-
-{
bug_891
Memcheck:Unaddressable
fun:XShmPutImage
diff --git a/webrtc/.gitignore b/webrtc/.gitignore
index dade989..89cb6db 100644
--- a/webrtc/.gitignore
+++ b/webrtc/.gitignore
@@ -1 +1,27 @@
-# This file is for projects that checkout webrtc/ directly (e.g. Chromium).
+# This file is for projects that checkout webrtc/ directly (e.g. Chromium). It
+# is a truncated copy of the .gitignore file in the parent directory.
+*.DS_Store
+*.Makefile
+*.ncb
+*.ninja
+*.props
+*.pyc
+*.rules
+*.scons
+*.sdf
+*.sln
+*.suo
+*.targets
+*.user
+*.vcproj
+*.vcxproj
+*.vcxproj.filters
+*.vpj
+*.vpw
+*.vpwhistu
+*.vtg
+*.xcodeproj
+*_proto.xml
+*_proto_cpp.xml
+*~
+.*.sw?
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index ac14d7d..be824b7 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -8,8 +8,8 @@
# TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
-import("//build/config/crypto.gni")
import("//build/config/linux/pkg_config.gni")
+import("//build/config/sanitizers/sanitizers.gni")
import("build/webrtc.gni")
import("//third_party/protobuf/proto_library.gni")
@@ -178,8 +178,8 @@
public_configs = [ ":common_inherited_config" ]
deps = [
- "audio",
":webrtc_common",
+ "audio",
"base:rtc_base",
"call",
"common_audio",
@@ -219,8 +219,8 @@
testonly = true
deps = [
":webrtc",
- "modules/video_render:video_render_internal_impl",
"modules/video_capture:video_capture_internal_impl",
+ "modules/video_render:video_render_internal_impl",
"test",
]
}
@@ -238,6 +238,12 @@
configs += [ ":common_config" ]
public_configs = [ ":common_inherited_config" ]
+
+ if (is_clang && !is_nacl) {
+ # Suppress warnings from Chrome's Clang plugins.
+ # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+ configs -= [ "//build/config/clang:find_bad_constructs" ]
+ }
}
source_set("gtest_prod") {
@@ -279,3 +285,14 @@
configs -= [ "//build/config/clang:find_bad_constructs" ]
}
}
+
+if (use_libfuzzer || use_drfuzz) {
+ # This target is only here for gn to discover fuzzer build targets under
+ # webrtc/test/fuzzers/.
+ group("webrtc_fuzzers_dummy") {
+ testonly = true
+ deps = [
+ "test/fuzzers:webrtc_fuzzer_main",
+ ]
+ }
+}
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
new file mode 100644
index 0000000..7cfa083
--- /dev/null
+++ b/webrtc/api/BUILD.gn
@@ -0,0 +1,76 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../build/webrtc.gni")
+
+config("ios_config") {
+ libs = [
+ "CoreGraphics.framework",
+ "GLKit.framework",
+ "OpenGLES.framework",
+ "QuartzCore.framework",
+ ]
+}
+
+if (is_ios) {
+ source_set("rtc_api_objc") {
+ deps = [
+ "//webrtc/base:rtc_base_objc",
+ #"//talk/libjingle:libjingle_peerconnection",
+ ]
+ cflags = [
+ "-fobjc-arc",
+ "-Wobjc-missing-property-synthesis",
+ ]
+ sources = [
+ # Add these when there's a BUILD.gn for peer connection APIs
+ #"objc/RTCIceCandidate+Private.h",
+ #"objc/RTCIceCandidate.h",
+ #"objc/RTCIceCandidate.mm",
+ #"objc/RTCMediaSource+Private.h",
+ #"objc/RTCMediaSource.h",
+ #"objc/RTCMediaSource.mm",
+ #"objc/RTCMediaStreamTrack+Private.h",
+ #"objc/RTCMediaStreamTrack.h",
+ #"objc/RTCMediaStreamTrack.mm",
+ "objc/RTCIceServer+Private.h",
+ "objc/RTCIceServer.h",
+ "objc/RTCIceServer.mm",
+ "objc/RTCMediaConstraints+Private.h",
+ "objc/RTCMediaConstraints.h",
+ "objc/RTCMediaConstraints.mm",
+ "objc/RTCOpenGLVideoRenderer.h",
+ "objc/RTCOpenGLVideoRenderer.mm",
+ "objc/RTCSessionDescription+Private.h",
+ "objc/RTCSessionDescription.h",
+ "objc/RTCSessionDescription.mm",
+ "objc/RTCStatsReport+Private.h",
+ "objc/RTCStatsReport.h",
+ "objc/RTCStatsReport.mm",
+ "objc/RTCVideoFrame+Private.h",
+ "objc/RTCVideoFrame.h",
+ "objc/RTCVideoFrame.mm",
+ "objc/RTCVideoRenderer.h",
+ "objc/WebRTC-Prefix.pch",
+ ]
+
+ if (is_ios) {
+ sources += [
+ "objc/RTCEAGLVideoView.h",
+ "objc/RTCEAGLVideoView.m",
+ ]
+ }
+
+ if (is_mac) {
+ sources += [
+ "objc/RTCNSGLVideoView.h",
+ "objc/RTCNSGLVideoView.m",
+ ]
+ }
+ }
+}
diff --git a/webrtc/api/OWNERS b/webrtc/api/OWNERS
new file mode 100644
index 0000000..cd06158
--- /dev/null
+++ b/webrtc/api/OWNERS
@@ -0,0 +1 @@
+tkchin@webrtc.org
diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp
new file mode 100644
index 0000000..ba3fe8d
--- /dev/null
+++ b/webrtc/api/api.gyp
@@ -0,0 +1,83 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [ '../build/common.gypi', ],
+ 'conditions': [
+ ['OS=="ios"', {
+ 'targets': [
+ {
+ 'target_name': 'rtc_api_objc',
+ 'type': 'static_library',
+ 'dependencies': [
+ '<(webrtc_root)/base/base.gyp:rtc_base_objc',
+ '../../talk/libjingle.gyp:libjingle_peerconnection',
+ ],
+ 'sources': [
+ 'objc/RTCIceCandidate+Private.h',
+ 'objc/RTCIceCandidate.h',
+ 'objc/RTCIceCandidate.mm',
+ 'objc/RTCIceServer+Private.h',
+ 'objc/RTCIceServer.h',
+ 'objc/RTCIceServer.mm',
+ 'objc/RTCMediaConstraints+Private.h',
+ 'objc/RTCMediaConstraints.h',
+ 'objc/RTCMediaConstraints.mm',
+ 'objc/RTCMediaSource+Private.h',
+ 'objc/RTCMediaSource.h',
+ 'objc/RTCMediaSource.mm',
+ 'objc/RTCMediaStreamTrack+Private.h',
+ 'objc/RTCMediaStreamTrack.h',
+ 'objc/RTCMediaStreamTrack.mm',
+ 'objc/RTCOpenGLVideoRenderer.h',
+ 'objc/RTCOpenGLVideoRenderer.mm',
+ 'objc/RTCSessionDescription+Private.h',
+ 'objc/RTCSessionDescription.h',
+ 'objc/RTCSessionDescription.mm',
+ 'objc/RTCStatsReport+Private.h',
+ 'objc/RTCStatsReport.h',
+ 'objc/RTCStatsReport.mm',
+ 'objc/RTCVideoFrame+Private.h',
+ 'objc/RTCVideoFrame.h',
+ 'objc/RTCVideoFrame.mm',
+ 'objc/RTCVideoRenderer.h',
+ ],
+ 'conditions': [
+ ['OS=="ios"', {
+ 'sources': [
+ 'objc/RTCEAGLVideoView.h',
+ 'objc/RTCEAGLVideoView.m',
+ ],
+ 'all_dependent_settings': {
+ 'xcode_settings': {
+ 'OTHER_LDFLAGS': [
+ '-framework CoreGraphics',
+ '-framework GLKit',
+ '-framework OpenGLES',
+ '-framework QuartzCore',
+ ]
+ }
+ }
+ }],
+ ['OS=="mac"', {
+ 'sources': [
+ 'objc/RTCNSGLVideoView.h',
+ 'objc/RTCNSGLVideoView.m',
+ ],
+ }],
+ ],
+ 'xcode_settings': {
+ 'CLANG_ENABLE_OBJC_ARC': 'YES',
+ 'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'YES',
+ 'GCC_PREFIX_HEADER': 'objc/WebRTC-Prefix.pch',
+ },
+ }
+ ],
+ }], # OS=="ios"
+ ],
+}
diff --git a/webrtc/api/api_tests.gyp b/webrtc/api/api_tests.gyp
new file mode 100644
index 0000000..c2c18bc
--- /dev/null
+++ b/webrtc/api/api_tests.gyp
@@ -0,0 +1,40 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'includes': [ '../build/common.gypi', ],
+ 'conditions': [
+ ['OS=="ios"', {
+ 'targets': [
+ {
+ 'target_name': 'rtc_api_objc_test',
+ 'type': 'executable',
+ 'dependencies': [
+ '<(webrtc_root)/api/api.gyp:rtc_api_objc',
+ '<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
+ ],
+ 'sources': [
+ 'objctests/RTCIceCandidateTest.mm',
+ 'objctests/RTCIceServerTest.mm',
+ 'objctests/RTCMediaConstraintsTest.mm',
+ 'objctests/RTCSessionDescriptionTest.mm',
+ ],
+ 'xcode_settings': {
+ 'CLANG_ENABLE_OBJC_ARC': 'YES',
+ 'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'YES',
+ 'GCC_PREFIX_HEADER': 'objc/WebRTC-Prefix.pch',
+ # |-ObjC| flag needed to make sure category method implementations
+ # are included:
+ # https://developer.apple.com/library/mac/qa/qa1490/_index.html
+ 'OTHER_LDFLAGS': ['-ObjC'],
+ },
+ }
+ ],
+ }], # OS=="ios"
+ ],
+}
diff --git a/webrtc/api/objc/OWNERS b/webrtc/api/objc/OWNERS
new file mode 100644
index 0000000..cd06158
--- /dev/null
+++ b/webrtc/api/objc/OWNERS
@@ -0,0 +1 @@
+tkchin@webrtc.org
diff --git a/webrtc/api/objc/README b/webrtc/api/objc/README
new file mode 100644
index 0000000..bd33e61
--- /dev/null
+++ b/webrtc/api/objc/README
@@ -0,0 +1,3 @@
+This is a work-in-progress to update the Objective-C API according to the W3C
+specification. The Objective-C API located at talk/app/webrtc/objc is
+deprecated, but will remain for the time being.
diff --git a/webrtc/api/objc/RTCEAGLVideoView.h b/webrtc/api/objc/RTCEAGLVideoView.h
new file mode 100644
index 0000000..1a57df7
--- /dev/null
+++ b/webrtc/api/objc/RTCEAGLVideoView.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCVideoRenderer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCEAGLVideoView;
+@protocol RTCEAGLVideoViewDelegate
+
+- (void)videoView:(RTCEAGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
+
+@end
+
+/**
+ * RTCEAGLVideoView is an RTCVideoRenderer which renders video frames in its
+ * bounds using OpenGLES 2.0.
+ */
+@interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
+
+@property(nonatomic, weak) id<RTCEAGLVideoViewDelegate> delegate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCEAGLVideoView.m b/webrtc/api/objc/RTCEAGLVideoView.m
new file mode 100644
index 0000000..e664ede
--- /dev/null
+++ b/webrtc/api/objc/RTCEAGLVideoView.m
@@ -0,0 +1,259 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCEAGLVideoView.h"
+
+#import <GLKit/GLKit.h>
+
+#import "RTCVideoFrame.h"
+#import "RTCOpenGLVideoRenderer.h"
+
+// RTCDisplayLinkTimer wraps a CADisplayLink and is set to fire every two screen
+// refreshes, which should be 30fps. We wrap the display link in order to avoid
+// a retain cycle since CADisplayLink takes a strong reference onto its target.
+// The timer is paused by default.
+@interface RTCDisplayLinkTimer : NSObject
+
+@property(nonatomic) BOOL isPaused;
+
+- (instancetype)initWithTimerHandler:(void (^)(void))timerHandler;
+- (void)invalidate;
+
+@end
+
+@implementation RTCDisplayLinkTimer {
+ CADisplayLink *_displayLink;
+ void (^_timerHandler)(void);
+}
+
+- (instancetype)initWithTimerHandler:(void (^)(void))timerHandler {
+ NSParameterAssert(timerHandler);
+ if (self = [super init]) {
+ _timerHandler = timerHandler;
+ _displayLink =
+ [CADisplayLink displayLinkWithTarget:self
+ selector:@selector(displayLinkDidFire:)];
+ _displayLink.paused = YES;
+ // Set to half of screen refresh, which should be 30fps.
+ [_displayLink setFrameInterval:2];
+ [_displayLink addToRunLoop:[NSRunLoop currentRunLoop]
+ forMode:NSRunLoopCommonModes];
+ }
+ return self;
+}
+
+- (void)dealloc {
+ [self invalidate];
+}
+
+- (BOOL)isPaused {
+ return _displayLink.paused;
+}
+
+- (void)setIsPaused:(BOOL)isPaused {
+ _displayLink.paused = isPaused;
+}
+
+- (void)invalidate {
+ [_displayLink invalidate];
+}
+
+- (void)displayLinkDidFire:(CADisplayLink *)displayLink {
+ _timerHandler();
+}
+
+@end
+
+// RTCEAGLVideoView wraps a GLKView which is setup with
+// enableSetNeedsDisplay = NO for the purpose of gaining control of
+// exactly when to call -[GLKView display]. This need for extra
+// control is required to avoid triggering method calls on GLKView
+// that results in attempting to bind the underlying render buffer
+// when the drawable size would be empty which would result in the
+// error GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT. -[GLKView display] is
+// the method that will trigger the binding of the render
+// buffer. Because the standard behaviour of -[UIView setNeedsDisplay]
+// is disabled for the reasons above, the RTCEAGLVideoView maintains
+// its own |isDirty| flag.
+
+@interface RTCEAGLVideoView () <GLKViewDelegate>
+// |videoFrame| is set when we receive a frame from a worker thread and is read
+// from the display link callback so atomicity is required.
+@property(atomic, strong) RTCVideoFrame *videoFrame;
+@property(nonatomic, readonly) GLKView *glkView;
+@property(nonatomic, readonly) RTCOpenGLVideoRenderer *glRenderer;
+@end
+
+@implementation RTCEAGLVideoView {
+ RTCDisplayLinkTimer *_timer;
+ // This flag should only be set and read on the main thread (e.g. by
+ // setNeedsDisplay)
+ BOOL _isDirty;
+}
+
+@synthesize delegate = _delegate;
+@synthesize videoFrame = _videoFrame;
+@synthesize glkView = _glkView;
+@synthesize glRenderer = _glRenderer;
+
+- (instancetype)initWithFrame:(CGRect)frame {
+ if (self = [super initWithFrame:frame]) {
+ [self configure];
+ }
+ return self;
+}
+
+- (instancetype)initWithCoder:(NSCoder *)aDecoder {
+ if (self = [super initWithCoder:aDecoder]) {
+ [self configure];
+ }
+ return self;
+}
+
+- (void)configure {
+ EAGLContext *glContext =
+ [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES3];
+ if (!glContext) {
+ glContext = [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES2];
+ }
+ _glRenderer = [[RTCOpenGLVideoRenderer alloc] initWithContext:glContext];
+
+ // GLKView manages a framebuffer for us.
+ _glkView = [[GLKView alloc] initWithFrame:CGRectZero
+ context:glContext];
+ _glkView.drawableColorFormat = GLKViewDrawableColorFormatRGBA8888;
+ _glkView.drawableDepthFormat = GLKViewDrawableDepthFormatNone;
+ _glkView.drawableStencilFormat = GLKViewDrawableStencilFormatNone;
+ _glkView.drawableMultisample = GLKViewDrawableMultisampleNone;
+ _glkView.delegate = self;
+ _glkView.layer.masksToBounds = YES;
+ _glkView.enableSetNeedsDisplay = NO;
+ [self addSubview:_glkView];
+
+ // Listen to application state in order to clean up OpenGL before app goes
+ // away.
+ NSNotificationCenter *notificationCenter =
+ [NSNotificationCenter defaultCenter];
+ [notificationCenter addObserver:self
+ selector:@selector(willResignActive)
+ name:UIApplicationWillResignActiveNotification
+ object:nil];
+ [notificationCenter addObserver:self
+ selector:@selector(didBecomeActive)
+ name:UIApplicationDidBecomeActiveNotification
+ object:nil];
+
+ // Frames are received on a separate thread, so we poll for current frame
+ // using a refresh rate proportional to screen refresh frequency. This
+ // occurs on the main thread.
+ __weak RTCEAGLVideoView *weakSelf = self;
+ _timer = [[RTCDisplayLinkTimer alloc] initWithTimerHandler:^{
+ RTCEAGLVideoView *strongSelf = weakSelf;
+ [strongSelf displayLinkTimerDidFire];
+ }];
+ [self setupGL];
+}
+
+- (void)dealloc {
+ [[NSNotificationCenter defaultCenter] removeObserver:self];
+ UIApplicationState appState =
+ [UIApplication sharedApplication].applicationState;
+ if (appState == UIApplicationStateActive) {
+ [self teardownGL];
+ }
+ [_timer invalidate];
+}
+
+#pragma mark - UIView
+
+- (void)setNeedsDisplay {
+ [super setNeedsDisplay];
+ _isDirty = YES;
+}
+
+- (void)setNeedsDisplayInRect:(CGRect)rect {
+ [super setNeedsDisplayInRect:rect];
+ _isDirty = YES;
+}
+
+- (void)layoutSubviews {
+ [super layoutSubviews];
+ _glkView.frame = self.bounds;
+}
+
+#pragma mark - GLKViewDelegate
+
+// This method is called when the GLKView's content is dirty and needs to be
+// redrawn. This occurs on main thread.
+- (void)glkView:(GLKView *)view drawInRect:(CGRect)rect {
+ // The renderer will draw the frame to the framebuffer corresponding to the
+ // one used by |view|.
+ [_glRenderer drawFrame:self.videoFrame];
+}
+
+#pragma mark - RTCVideoRenderer
+
+// These methods may be called on non-main thread.
+- (void)setSize:(CGSize)size {
+ __weak RTCEAGLVideoView *weakSelf = self;
+ dispatch_async(dispatch_get_main_queue(), ^{
+ RTCEAGLVideoView *strongSelf = weakSelf;
+ [strongSelf.delegate videoView:strongSelf didChangeVideoSize:size];
+ });
+}
+
+- (void)renderFrame:(RTCVideoFrame *)frame {
+ self.videoFrame = frame;
+}
+
+#pragma mark - Private
+
+- (void)displayLinkTimerDidFire {
+ // Don't render unless video frame have changed or the view content
+ // has explicitly been marked dirty.
+ if (!_isDirty && _glRenderer.lastDrawnFrame == self.videoFrame) {
+ return;
+ }
+
+ // Always reset isDirty at this point, even if -[GLKView display]
+ // won't be called in the case the drawable size is empty.
+ _isDirty = NO;
+
+ // Only call -[GLKView display] if the drawable size is
+ // non-empty. Calling display will make the GLKView setup its
+ // render buffer if necessary, but that will fail with error
+ // GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT if size is empty.
+ if (self.bounds.size.width > 0 && self.bounds.size.height > 0) {
+ [_glkView display];
+ }
+}
+
+- (void)setupGL {
+ self.videoFrame = nil;
+ [_glRenderer setupGL];
+ _timer.isPaused = NO;
+}
+
+- (void)teardownGL {
+ self.videoFrame = nil;
+ _timer.isPaused = YES;
+ [_glkView deleteDrawable];
+ [_glRenderer teardownGL];
+}
+
+- (void)didBecomeActive {
+ [self setupGL];
+}
+
+- (void)willResignActive {
+ [self teardownGL];
+}
+
+@end
diff --git a/webrtc/api/objc/RTCIceCandidate+Private.h b/webrtc/api/objc/RTCIceCandidate+Private.h
new file mode 100644
index 0000000..ca95a43
--- /dev/null
+++ b/webrtc/api/objc/RTCIceCandidate+Private.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceCandidate.h"
+
+#include "talk/app/webrtc/jsep.h"
+#include "webrtc/base/scoped_ptr.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIceCandidate ()
+
+/**
+ * The native IceCandidateInterface representation of this RTCIceCandidate
+ * object. This is needed to pass to the underlying C++ APIs.
+ */
+@property(nonatomic, readonly)
+ rtc::scoped_ptr<webrtc::IceCandidateInterface> nativeCandidate;
+
+/**
+ * Initialize an RTCIceCandidate from a native IceCandidateInterface. No
+ * ownership is taken of the native candidate.
+ */
+- (instancetype)initWithNativeCandidate:
+ (webrtc::IceCandidateInterface *)candidate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCIceCandidate.h b/webrtc/api/objc/RTCIceCandidate.h
new file mode 100644
index 0000000..41ea69e
--- /dev/null
+++ b/webrtc/api/objc/RTCIceCandidate.h
@@ -0,0 +1,44 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIceCandidate : NSObject
+
+/**
+ * If present, the identifier of the "media stream identification" for the media
+ * component this candidate is associated with.
+ */
+@property(nonatomic, readonly, nullable) NSString *sdpMid;
+
+/**
+ * The index (starting at zero) of the media description this candidate is
+ * associated with in the SDP.
+ */
+@property(nonatomic, readonly) NSInteger sdpMLineIndex;
+
+/** The SDP string for this candidate. */
+@property(nonatomic, readonly) NSString *sdp;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Initialize an RTCIceCandidate from SDP.
+ */
+- (instancetype)initWithSdp:(NSString *)sdp
+ sdpMLineIndex:(NSInteger)sdpMLineIndex
+ sdpMid:(nullable NSString *)sdpMid
+ NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCIceCandidate.mm b/webrtc/api/objc/RTCIceCandidate.mm
new file mode 100644
index 0000000..9e094f6
--- /dev/null
+++ b/webrtc/api/objc/RTCIceCandidate.mm
@@ -0,0 +1,70 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceCandidate.h"
+
+#import "webrtc/api/objc/RTCIceCandidate+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+#import "webrtc/base/objc/RTCLogging.h"
+
+@implementation RTCIceCandidate
+
+@synthesize sdpMid = _sdpMid;
+@synthesize sdpMLineIndex = _sdpMLineIndex;
+@synthesize sdp = _sdp;
+
+- (instancetype)initWithSdp:(NSString *)sdp
+ sdpMLineIndex:(NSInteger)sdpMLineIndex
+ sdpMid:(NSString *)sdpMid {
+ NSParameterAssert(sdp.length);
+ if (self = [super init]) {
+ _sdpMid = [sdpMid copy];
+ _sdpMLineIndex = sdpMLineIndex;
+ _sdp = [sdp copy];
+ }
+ return self;
+}
+
+- (NSString *)description {
+ return [NSString stringWithFormat:@"RTCIceCandidate:\n%@\n%ld\n%@",
+ _sdpMid,
+ (long)_sdpMLineIndex,
+ _sdp];
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeCandidate:
+ (webrtc::IceCandidateInterface *)candidate {
+ NSParameterAssert(candidate);
+ std::string sdp;
+ candidate->ToString(&sdp);
+
+ return [self initWithSdp:[NSString stringForStdString:sdp]
+ sdpMLineIndex:candidate->sdp_mline_index()
+ sdpMid:[NSString stringForStdString:candidate->sdp_mid()]];
+}
+
+- (rtc::scoped_ptr<webrtc::IceCandidateInterface>)nativeCandidate {
+ webrtc::SdpParseError error;
+
+ webrtc::IceCandidateInterface *candidate = webrtc::CreateIceCandidate(
+ _sdpMid.stdString, _sdpMLineIndex, _sdp.stdString, &error);
+
+ if (!candidate) {
+ RTCLog(@"Failed to create ICE candidate: %s\nline: %s",
+ error.description.c_str(),
+ error.line.c_str());
+ }
+
+ return rtc::scoped_ptr<webrtc::IceCandidateInterface>(candidate);
+}
+
+@end
diff --git a/webrtc/api/objc/RTCIceServer+Private.h b/webrtc/api/objc/RTCIceServer+Private.h
new file mode 100644
index 0000000..59f5a92
--- /dev/null
+++ b/webrtc/api/objc/RTCIceServer+Private.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceServer.h"
+
+#include "talk/app/webrtc/peerconnectioninterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIceServer ()
+
+/**
+ * IceServer struct representation of this RTCIceServer object's data.
+ * This is needed to pass to the underlying C++ APIs.
+ */
+@property(nonatomic, readonly)
+ webrtc::PeerConnectionInterface::IceServer iceServer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCIceServer.h b/webrtc/api/objc/RTCIceServer.h
new file mode 100644
index 0000000..be4e0d7
--- /dev/null
+++ b/webrtc/api/objc/RTCIceServer.h
@@ -0,0 +1,42 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCIceServer : NSObject
+
+/** URI(s) for this server represented as NSStrings. */
+@property(nonatomic, copy, readonly) NSArray<NSString *> *urlStrings;
+
+/** Username to use if this RTCIceServer object is a TURN server. */
+@property(nonatomic, copy, readonly, nullable) NSString *username;
+
+/** Credential to use if this RTCIceServer object is a TURN server. */
+@property(nonatomic, copy, readonly, nullable) NSString *credential;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Convenience initializer for a server with no authentication (e.g. STUN). */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, and credentialType.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+ username:(nullable NSString *)username
+ credential:(nullable NSString *)credential
+ NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCIceServer.mm b/webrtc/api/objc/RTCIceServer.mm
new file mode 100644
index 0000000..7a898e0
--- /dev/null
+++ b/webrtc/api/objc/RTCIceServer.mm
@@ -0,0 +1,64 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceServer.h"
+
+#import "webrtc/api/objc/RTCIceServer+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@implementation RTCIceServer
+
+@synthesize urlStrings = _urlStrings;
+@synthesize username = _username;
+@synthesize credential = _credential;
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings {
+ NSParameterAssert(urlStrings.count);
+ return [self initWithURLStrings:urlStrings
+ username:nil
+ credential:nil];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+ username:(NSString *)username
+ credential:(NSString *)credential {
+ NSParameterAssert(urlStrings.count);
+ if (self = [super init]) {
+ _urlStrings = [[NSArray alloc] initWithArray:urlStrings copyItems:YES];
+ _username = [username copy];
+ _credential = [credential copy];
+ }
+ return self;
+}
+
+- (NSString *)description {
+ return [NSString stringWithFormat:@"RTCIceServer:\n%@\n%@\n%@",
+ _urlStrings,
+ _username,
+ _credential];
+}
+
+#pragma mark - Private
+
+- (webrtc::PeerConnectionInterface::IceServer)iceServer {
+ __block webrtc::PeerConnectionInterface::IceServer iceServer;
+
+ iceServer.username = [NSString stdStringForString:_username];
+ iceServer.password = [NSString stdStringForString:_credential];
+
+ [_urlStrings enumerateObjectsUsingBlock:^(NSString *url,
+ NSUInteger idx,
+ BOOL *stop) {
+ iceServer.urls.push_back(url.stdString);
+ }];
+ return iceServer;
+}
+
+@end
diff --git a/webrtc/api/objc/RTCMediaConstraints+Private.h b/webrtc/api/objc/RTCMediaConstraints+Private.h
new file mode 100644
index 0000000..2c4b722
--- /dev/null
+++ b/webrtc/api/objc/RTCMediaConstraints+Private.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaConstraints.h"
+
+#include "talk/app/webrtc/mediaconstraintsinterface.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+
+class MediaConstraints : public MediaConstraintsInterface {
+ public:
+ virtual ~MediaConstraints();
+ MediaConstraints();
+ MediaConstraints(
+ const MediaConstraintsInterface::Constraints& mandatory,
+ const MediaConstraintsInterface::Constraints& optional);
+ virtual const Constraints& GetMandatory() const;
+ virtual const Constraints& GetOptional() const;
+
+ private:
+ MediaConstraintsInterface::Constraints mandatory_;
+ MediaConstraintsInterface::Constraints optional_;
+};
+
+} // namespace webrtc
+
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaConstraints ()
+
+/**
+ * A MediaConstraints representation of this RTCMediaConstraints object. This is
+ * needed to pass to the underlying C++ APIs.
+ */
+- (rtc::scoped_ptr<webrtc::MediaConstraints>)nativeConstraints;
+
+/** Return a native Constraints object representing these constraints */
++ (webrtc::MediaConstraintsInterface::Constraints)
+ nativeConstraintsForConstraints:
+ (NSDictionary<NSString *, NSString *> *)constraints;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCMediaConstraints.h b/webrtc/api/objc/RTCMediaConstraints.h
new file mode 100644
index 0000000..a8ad391
--- /dev/null
+++ b/webrtc/api/objc/RTCMediaConstraints.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaConstraints : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize with mandatory and/or optional constraints. */
+- (instancetype)initWithMandatoryConstraints:
+ (nullable NSDictionary<NSString *, NSString *> *)mandatory
+ optionalConstraints:
+ (nullable NSDictionary<NSString *, NSString *> *)optional
+ NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCMediaConstraints.mm b/webrtc/api/objc/RTCMediaConstraints.mm
new file mode 100644
index 0000000..a53a517
--- /dev/null
+++ b/webrtc/api/objc/RTCMediaConstraints.mm
@@ -0,0 +1,92 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaConstraints.h"
+
+#import "webrtc/api/objc/RTCMediaConstraints+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+namespace webrtc {
+
+MediaConstraints::~MediaConstraints() {}
+
+MediaConstraints::MediaConstraints() {}
+
+MediaConstraints::MediaConstraints(
+ const MediaConstraintsInterface::Constraints& mandatory,
+ const MediaConstraintsInterface::Constraints& optional)
+ : mandatory_(mandatory), optional_(optional) {}
+
+const MediaConstraintsInterface::Constraints&
+MediaConstraints::GetMandatory() const {
+ return mandatory_;
+}
+
+const MediaConstraintsInterface::Constraints&
+MediaConstraints::GetOptional() const {
+ return optional_;
+}
+
+} // namespace webrtc
+
+
+@implementation RTCMediaConstraints {
+ NSDictionary<NSString *, NSString *> *_mandatory;
+ NSDictionary<NSString *, NSString *> *_optional;
+}
+
+- (instancetype)initWithMandatoryConstraints:
+ (NSDictionary<NSString *, NSString *> *)mandatory
+ optionalConstraints:
+ (NSDictionary<NSString *, NSString *> *)optional {
+ if (self = [super init]) {
+ _mandatory = [[NSDictionary alloc] initWithDictionary:mandatory
+ copyItems:YES];
+ _optional = [[NSDictionary alloc] initWithDictionary:optional
+ copyItems:YES];
+ }
+ return self;
+}
+
+- (NSString *)description {
+ return [NSString stringWithFormat:@"RTCMediaConstraints:\n%@\n%@",
+ _mandatory,
+ _optional];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_ptr<webrtc::MediaConstraints>)nativeConstraints {
+ webrtc::MediaConstraintsInterface::Constraints mandatory =
+ [[self class] nativeConstraintsForConstraints:_mandatory];
+ webrtc::MediaConstraintsInterface::Constraints optional =
+ [[self class] nativeConstraintsForConstraints:_optional];
+
+ webrtc::MediaConstraints *nativeConstraints =
+ new webrtc::MediaConstraints(mandatory, optional);
+ return rtc::scoped_ptr<webrtc::MediaConstraints>(nativeConstraints);
+}
+
++ (webrtc::MediaConstraintsInterface::Constraints)
+ nativeConstraintsForConstraints:
+ (NSDictionary<NSString *, NSString *> *)constraints {
+ webrtc::MediaConstraintsInterface::Constraints nativeConstraints;
+ for (NSString *key in constraints) {
+ NSAssert([key isKindOfClass:[NSString class]],
+ @"%@ is not an NSString.", key);
+ NSAssert([constraints[key] isKindOfClass:[NSString class]],
+ @"%@ is not an NSString.", constraints[key]);
+ nativeConstraints.push_back(webrtc::MediaConstraintsInterface::Constraint(
+ key.stdString, constraints[key].stdString));
+ }
+ return nativeConstraints;
+}
+
+@end
diff --git a/webrtc/api/objc/RTCMediaSource+Private.h b/webrtc/api/objc/RTCMediaSource+Private.h
new file mode 100644
index 0000000..fcbaad8
--- /dev/null
+++ b/webrtc/api/objc/RTCMediaSource+Private.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaSource.h"
+
+#include "talk/app/webrtc/mediastreaminterface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaSource ()
+
+/**
+ * The MediaSourceInterface object passed to this RTCMediaSource during
+ * construction.
+ */
+@property(nonatomic, readonly)
+ rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
+
+/** Initialize an RTCMediaSource from a native MediaSourceInterface. */
+- (instancetype)initWithNativeMediaSource:
+ (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+ NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
+ (RTCSourceState)state;
+
++ (RTCSourceState)sourceStateForNativeState:
+ (webrtc::MediaSourceInterface::SourceState)nativeState;
+
++ (NSString *)stringForState:(RTCSourceState)state;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCMediaSource.h b/webrtc/api/objc/RTCMediaSource.h
new file mode 100644
index 0000000..0b36b8d
--- /dev/null
+++ b/webrtc/api/objc/RTCMediaSource.h
@@ -0,0 +1,31 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+typedef NS_ENUM(NSInteger, RTCSourceState) {
+ RTCSourceStateInitializing,
+ RTCSourceStateLive,
+ RTCSourceStateEnded,
+ RTCSourceStateMuted,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaSource : NSObject
+
+/** The current state of the RTCMediaSource. */
+@property(nonatomic, readonly) RTCSourceState state;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCMediaSource.mm b/webrtc/api/objc/RTCMediaSource.mm
new file mode 100644
index 0000000..5f46ab8
--- /dev/null
+++ b/webrtc/api/objc/RTCMediaSource.mm
@@ -0,0 +1,84 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaSource.h"
+
+#import "webrtc/api/objc/RTCMediaSource+Private.h"
+
+@implementation RTCMediaSource {
+ rtc::scoped_refptr<webrtc::MediaSourceInterface> _nativeMediaSource;
+}
+
+- (RTCSourceState)state {
+ return [[self class] sourceStateForNativeState:_nativeMediaSource->state()];
+}
+
+- (NSString *)description {
+ return [NSString stringWithFormat:@"RTCMediaSource:\n%@",
+ [[self class] stringForState:self.state]];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource {
+ return _nativeMediaSource;
+}
+
+- (instancetype)initWithNativeMediaSource:
+ (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource {
+ NSParameterAssert(nativeMediaSource);
+ if (self = [super init]) {
+ _nativeMediaSource = nativeMediaSource;
+ }
+ return self;
+}
+
++ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
+ (RTCSourceState)state {
+ switch (state) {
+ case RTCSourceStateInitializing:
+ return webrtc::MediaSourceInterface::kInitializing;
+ case RTCSourceStateLive:
+ return webrtc::MediaSourceInterface::kLive;
+ case RTCSourceStateEnded:
+ return webrtc::MediaSourceInterface::kEnded;
+ case RTCSourceStateMuted:
+ return webrtc::MediaSourceInterface::kMuted;
+ }
+}
+
++ (RTCSourceState)sourceStateForNativeState:
+ (webrtc::MediaSourceInterface::SourceState)nativeState {
+ switch (nativeState) {
+ case webrtc::MediaSourceInterface::kInitializing:
+ return RTCSourceStateInitializing;
+ case webrtc::MediaSourceInterface::kLive:
+ return RTCSourceStateLive;
+ case webrtc::MediaSourceInterface::kEnded:
+ return RTCSourceStateEnded;
+ case webrtc::MediaSourceInterface::kMuted:
+ return RTCSourceStateMuted;
+ }
+}
+
++ (NSString *)stringForState:(RTCSourceState)state {
+ switch (state) {
+ case RTCSourceStateInitializing:
+ return @"Initializing";
+ case RTCSourceStateLive:
+ return @"Live";
+ case RTCSourceStateEnded:
+ return @"Ended";
+ case RTCSourceStateMuted:
+ return @"Muted";
+ }
+}
+
+@end
diff --git a/webrtc/api/objc/RTCMediaStreamTrack+Private.h b/webrtc/api/objc/RTCMediaStreamTrack+Private.h
new file mode 100644
index 0000000..3e17e63
--- /dev/null
+++ b/webrtc/api/objc/RTCMediaStreamTrack+Private.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaStreamTrack.h"
+
+#include "talk/app/webrtc/mediastreaminterface.h"
+#include "webrtc/base/scoped_ptr.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaStreamTrack ()
+
+/**
+ * The native MediaStreamTrackInterface representation of this
+ * RTCMediaStreamTrack object. This is needed to pass to the underlying C++
+ * APIs.
+ */
+@property(nonatomic, readonly)
+ rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack;
+
+/**
+ * Initialize an RTCMediaStreamTrack from a native MediaStreamTrackInterface.
+ */
+- (instancetype)initWithNativeTrack:
+ (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+ NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
+ (RTCMediaStreamTrackState)state;
+
++ (RTCMediaStreamTrackState)trackStateForNativeState:
+ (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
+
++ (NSString *)stringForState:(RTCMediaStreamTrackState)state;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCMediaStreamTrack.h b/webrtc/api/objc/RTCMediaStreamTrack.h
new file mode 100644
index 0000000..beb48d3
--- /dev/null
+++ b/webrtc/api/objc/RTCMediaStreamTrack.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+/**
+ * Represents the state of the track. This exposes the same states in C++,
+ * which include two more states than are in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCMediaStreamTrackState) {
+ RTCMediaStreamTrackStateInitializing,
+ RTCMediaStreamTrackStateLive,
+ RTCMediaStreamTrackStateEnded,
+ RTCMediaStreamTrackStateFailed,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMediaStreamTrack : NSObject
+
+/**
+ * The kind of track. For example, "audio" if this track represents an audio
+ * track and "video" if this track represents a video track.
+ */
+@property(nonatomic, readonly) NSString *kind;
+
+/** An identifier string. */
+@property(nonatomic, readonly) NSString *trackId;
+
+/** The enabled state of the track. */
+@property(nonatomic) BOOL isEnabled;
+
+/** The state of the track. */
+@property(nonatomic, readonly) RTCMediaStreamTrackState readyState;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCMediaStreamTrack.mm b/webrtc/api/objc/RTCMediaStreamTrack.mm
new file mode 100644
index 0000000..e5751b0
--- /dev/null
+++ b/webrtc/api/objc/RTCMediaStreamTrack.mm
@@ -0,0 +1,105 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaStreamTrack.h"
+
+#import "webrtc/api/objc/RTCMediaStreamTrack+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@implementation RTCMediaStreamTrack {
+ rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> _nativeTrack;
+}
+
+- (NSString *)kind {
+ return [NSString stringForStdString:_nativeTrack->kind()];
+}
+
+- (NSString *)trackId {
+ return [NSString stringForStdString:_nativeTrack->id()];
+}
+
+- (BOOL)isEnabled {
+ return _nativeTrack->enabled();
+}
+
+- (void)setIsEnabled:(BOOL)isEnabled {
+ _nativeTrack->set_enabled(isEnabled);
+}
+
+- (RTCMediaStreamTrackState)readyState {
+ return [[self class] trackStateForNativeState:_nativeTrack->state()];
+}
+
+- (NSString *)description {
+ NSString *readyState = [[self class] stringForState:self.readyState];
+ return [NSString stringWithFormat:@"RTCMediaStreamTrack:\n%@\n%@\n%@\n%@",
+ self.kind,
+ self.trackId,
+ self.isEnabled ? @"enabled" : @"disabled",
+ readyState];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
+ return _nativeTrack;
+}
+
+- (instancetype)initWithNativeTrack:
+ (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
+ NSParameterAssert(nativeTrack);
+ if (self = [super init]) {
+ _nativeTrack = nativeTrack;
+ }
+ return self;
+}
+
++ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
+ (RTCMediaStreamTrackState)state {
+ switch (state) {
+ case RTCMediaStreamTrackStateInitializing:
+ return webrtc::MediaStreamTrackInterface::kInitializing;
+ case RTCMediaStreamTrackStateLive:
+ return webrtc::MediaStreamTrackInterface::kLive;
+ case RTCMediaStreamTrackStateEnded:
+ return webrtc::MediaStreamTrackInterface::kEnded;
+ case RTCMediaStreamTrackStateFailed:
+ return webrtc::MediaStreamTrackInterface::kFailed;
+ }
+}
+
++ (RTCMediaStreamTrackState)trackStateForNativeState:
+ (webrtc::MediaStreamTrackInterface::TrackState)nativeState {
+ switch (nativeState) {
+ case webrtc::MediaStreamTrackInterface::kInitializing:
+ return RTCMediaStreamTrackStateInitializing;
+ case webrtc::MediaStreamTrackInterface::kLive:
+ return RTCMediaStreamTrackStateLive;
+ case webrtc::MediaStreamTrackInterface::kEnded:
+ return RTCMediaStreamTrackStateEnded;
+ case webrtc::MediaStreamTrackInterface::kFailed:
+ return RTCMediaStreamTrackStateFailed;
+ }
+}
+
++ (NSString *)stringForState:(RTCMediaStreamTrackState)state {
+ switch (state) {
+ case RTCMediaStreamTrackStateInitializing:
+ return @"Initializing";
+ case RTCMediaStreamTrackStateLive:
+ return @"Live";
+ case RTCMediaStreamTrackStateEnded:
+ return @"Ended";
+ case RTCMediaStreamTrackStateFailed:
+ return @"Failed";
+ }
+}
+
+@end
diff --git a/webrtc/api/objc/RTCNSGLVideoView.h b/webrtc/api/objc/RTCNSGLVideoView.h
new file mode 100644
index 0000000..27eb31e
--- /dev/null
+++ b/webrtc/api/objc/RTCNSGLVideoView.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if TARGET_OS_IPHONE
+#error "This file targets OSX."
+#endif
+
+#import <AppKit/NSOpenGLView.h>
+
+#import "RTCVideoRenderer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCNSGLVideoView;
+@protocol RTCNSGLVideoViewDelegate
+
+- (void)videoView:(RTCNSGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
+
+@end
+
+@interface RTCNSGLVideoView : NSOpenGLView <RTCVideoRenderer>
+
+@property(nonatomic, weak) id<RTCNSGLVideoViewDelegate> delegate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCNSGLVideoView.m b/webrtc/api/objc/RTCNSGLVideoView.m
new file mode 100644
index 0000000..063e6f1
--- /dev/null
+++ b/webrtc/api/objc/RTCNSGLVideoView.m
@@ -0,0 +1,141 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCNSGLVideoView.h"
+
+#import <CoreVideo/CVDisplayLink.h>
+#import <OpenGL/gl3.h>
+#import "RTCVideoFrame.h"
+#import "RTCOpenGLVideoRenderer.h"
+
+@interface RTCNSGLVideoView ()
+// |videoFrame| is set when we receive a frame from a worker thread and is read
+// from the display link callback so atomicity is required.
+@property(atomic, strong) RTCVideoFrame *videoFrame;
+@property(atomic, strong) RTCOpenGLVideoRenderer *glRenderer;
+- (void)drawFrame;
+@end
+
+static CVReturn OnDisplayLinkFired(CVDisplayLinkRef displayLink,
+ const CVTimeStamp *now,
+ const CVTimeStamp *outputTime,
+ CVOptionFlags flagsIn,
+ CVOptionFlags *flagsOut,
+ void *displayLinkContext) {
+ RTCNSGLVideoView *view = (__bridge RTCNSGLVideoView *)displayLinkContext;
+ [view drawFrame];
+ return kCVReturnSuccess;
+}
+
+@implementation RTCNSGLVideoView {
+ CVDisplayLinkRef _displayLink;
+}
+
+@synthesize delegate = _delegate;
+@synthesize videoFrame = _videoFrame;
+@synthesize glRenderer = _glRenderer;
+
+- (void)dealloc {
+ [self teardownDisplayLink];
+}
+
+- (void)drawRect:(NSRect)rect {
+ [self drawFrame];
+}
+
+- (void)reshape {
+ [super reshape];
+ NSRect frame = [self frame];
+ CGLLockContext([[self openGLContext] CGLContextObj]);
+ glViewport(0, 0, frame.size.width, frame.size.height);
+ CGLUnlockContext([[self openGLContext] CGLContextObj]);
+}
+
+- (void)lockFocus {
+ NSOpenGLContext *context = [self openGLContext];
+ [super lockFocus];
+ if ([context view] != self) {
+ [context setView:self];
+ }
+ [context makeCurrentContext];
+}
+
+- (void)prepareOpenGL {
+ [super prepareOpenGL];
+ if (!self.glRenderer) {
+ self.glRenderer =
+ [[RTCOpenGLVideoRenderer alloc] initWithContext:[self openGLContext]];
+ }
+ [self.glRenderer setupGL];
+ [self setupDisplayLink];
+}
+
+- (void)clearGLContext {
+ [self.glRenderer teardownGL];
+ self.glRenderer = nil;
+ [super clearGLContext];
+}
+
+#pragma mark - RTCVideoRenderer
+
+// These methods may be called on non-main thread.
+- (void)setSize:(CGSize)size {
+ dispatch_async(dispatch_get_main_queue(), ^{
+ [self.delegate videoView:self didChangeVideoSize:size];
+ });
+}
+
+- (void)renderFrame:(RTCVideoFrame *)frame {
+ self.videoFrame = frame;
+}
+
+#pragma mark - Private
+
+- (void)drawFrame {
+ RTCVideoFrame *videoFrame = self.videoFrame;
+ if (self.glRenderer.lastDrawnFrame != videoFrame) {
+ // This method may be called from CVDisplayLink callback which isn't on the
+ // main thread so we have to lock the GL context before drawing.
+ CGLLockContext([[self openGLContext] CGLContextObj]);
+ [self.glRenderer drawFrame:videoFrame];
+ CGLUnlockContext([[self openGLContext] CGLContextObj]);
+ }
+}
+
+- (void)setupDisplayLink {
+ if (_displayLink) {
+ return;
+ }
+ // Synchronize buffer swaps with vertical refresh rate.
+ GLint swapInt = 1;
+ [[self openGLContext] setValues:&swapInt forParameter:NSOpenGLCPSwapInterval];
+
+ // Create display link.
+ CVDisplayLinkCreateWithActiveCGDisplays(&_displayLink);
+ CVDisplayLinkSetOutputCallback(_displayLink,
+ &OnDisplayLinkFired,
+ (__bridge void *)self);
+ // Set the display link for the current renderer.
+ CGLContextObj cglContext = [[self openGLContext] CGLContextObj];
+ CGLPixelFormatObj cglPixelFormat = [[self pixelFormat] CGLPixelFormatObj];
+ CVDisplayLinkSetCurrentCGDisplayFromOpenGLContext(
+ _displayLink, cglContext, cglPixelFormat);
+ CVDisplayLinkStart(_displayLink);
+}
+
+- (void)teardownDisplayLink {
+ if (!_displayLink) {
+ return;
+ }
+ CVDisplayLinkRelease(_displayLink);
+ _displayLink = NULL;
+}
+
+@end
diff --git a/webrtc/api/objc/RTCOpenGLVideoRenderer.h b/webrtc/api/objc/RTCOpenGLVideoRenderer.h
new file mode 100644
index 0000000..729839c
--- /dev/null
+++ b/webrtc/api/objc/RTCOpenGLVideoRenderer.h
@@ -0,0 +1,58 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#if TARGET_OS_IPHONE
+#import <GLKit/GLKit.h>
+#else
+#import <AppKit/NSOpenGL.h>
+#endif
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCVideoFrame;
+
+// RTCOpenGLVideoRenderer issues appropriate OpenGL commands to draw a frame to
+// the currently bound framebuffer. Supports OpenGL 3.2 and OpenGLES 2.0. OpenGL
+// framebuffer creation and management should be handled elsewhere using the
+// same context used to initialize this class.
+@interface RTCOpenGLVideoRenderer : NSObject
+
+// The last successfully drawn frame. Used to avoid drawing frames unnecessarily
+// hence saving battery life by reducing load.
+@property(nonatomic, readonly) RTCVideoFrame *lastDrawnFrame;
+
+#if TARGET_OS_IPHONE
+- (instancetype)initWithContext:(EAGLContext *)context
+ NS_DESIGNATED_INITIALIZER;
+#else
+- (instancetype)initWithContext:(NSOpenGLContext *)context
+ NS_DESIGNATED_INITIALIZER;
+#endif
+
+// Draws |frame| onto the currently bound OpenGL framebuffer. |setupGL| must be
+// called before this function will succeed.
+- (BOOL)drawFrame:(RTCVideoFrame *)frame;
+
+// The following methods are used to manage OpenGL resources. On iOS
+// applications should release resources when placed in background for use in
+// the foreground application. In fact, attempting to call OpenGLES commands
+// while in background will result in application termination.
+
+// Sets up the OpenGL state needed for rendering.
+- (void)setupGL;
+// Tears down the OpenGL state created by |setupGL|.
+- (void)teardownGL;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCOpenGLVideoRenderer.mm b/webrtc/api/objc/RTCOpenGLVideoRenderer.mm
new file mode 100644
index 0000000..56a6431
--- /dev/null
+++ b/webrtc/api/objc/RTCOpenGLVideoRenderer.mm
@@ -0,0 +1,485 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCOpenGLVideoRenderer.h"
+
+#include <string.h>
+
+#include "webrtc/base/scoped_ptr.h"
+
+#if TARGET_OS_IPHONE
+#import <OpenGLES/ES3/gl.h>
+#else
+#import <OpenGL/gl3.h>
+#endif
+
+#import "RTCVideoFrame.h"
+
+// TODO(tkchin): check and log openGL errors. Methods here return BOOLs in
+// anticipation of that happening in the future.
+
+#if TARGET_OS_IPHONE
+#define RTC_PIXEL_FORMAT GL_LUMINANCE
+#define SHADER_VERSION
+#define VERTEX_SHADER_IN "attribute"
+#define VERTEX_SHADER_OUT "varying"
+#define FRAGMENT_SHADER_IN "varying"
+#define FRAGMENT_SHADER_OUT
+#define FRAGMENT_SHADER_COLOR "gl_FragColor"
+#define FRAGMENT_SHADER_TEXTURE "texture2D"
+#else
+#define RTC_PIXEL_FORMAT GL_RED
+#define SHADER_VERSION "#version 150\n"
+#define VERTEX_SHADER_IN "in"
+#define VERTEX_SHADER_OUT "out"
+#define FRAGMENT_SHADER_IN "in"
+#define FRAGMENT_SHADER_OUT "out vec4 fragColor;\n"
+#define FRAGMENT_SHADER_COLOR "fragColor"
+#define FRAGMENT_SHADER_TEXTURE "texture"
+#endif
+
+// Vertex shader doesn't do anything except pass coordinates through.
+static const char kVertexShaderSource[] =
+ SHADER_VERSION
+ VERTEX_SHADER_IN " vec2 position;\n"
+ VERTEX_SHADER_IN " vec2 texcoord;\n"
+ VERTEX_SHADER_OUT " vec2 v_texcoord;\n"
+ "void main() {\n"
+ " gl_Position = vec4(position.x, position.y, 0.0, 1.0);\n"
+ " v_texcoord = texcoord;\n"
+ "}\n";
+
+// Fragment shader converts YUV values from input textures into a final RGB
+// pixel. The conversion formula is from http://www.fourcc.org/fccyvrgb.php.
+static const char kFragmentShaderSource[] =
+ SHADER_VERSION
+ "precision highp float;"
+ FRAGMENT_SHADER_IN " vec2 v_texcoord;\n"
+ "uniform lowp sampler2D s_textureY;\n"
+ "uniform lowp sampler2D s_textureU;\n"
+ "uniform lowp sampler2D s_textureV;\n"
+ FRAGMENT_SHADER_OUT
+ "void main() {\n"
+ " float y, u, v, r, g, b;\n"
+ " y = " FRAGMENT_SHADER_TEXTURE "(s_textureY, v_texcoord).r;\n"
+ " u = " FRAGMENT_SHADER_TEXTURE "(s_textureU, v_texcoord).r;\n"
+ " v = " FRAGMENT_SHADER_TEXTURE "(s_textureV, v_texcoord).r;\n"
+ " u = u - 0.5;\n"
+ " v = v - 0.5;\n"
+ " r = y + 1.403 * v;\n"
+ " g = y - 0.344 * u - 0.714 * v;\n"
+ " b = y + 1.770 * u;\n"
+ " " FRAGMENT_SHADER_COLOR " = vec4(r, g, b, 1.0);\n"
+ " }\n";
+
+// Compiles a shader of the given |type| with GLSL source |source| and returns
+// the shader handle or 0 on error.
+GLuint CreateShader(GLenum type, const GLchar *source) {
+ GLuint shader = glCreateShader(type);
+ if (!shader) {
+ return 0;
+ }
+ glShaderSource(shader, 1, &source, NULL);
+ glCompileShader(shader);
+ GLint compileStatus = GL_FALSE;
+ glGetShaderiv(shader, GL_COMPILE_STATUS, &compileStatus);
+ if (compileStatus == GL_FALSE) {
+ glDeleteShader(shader);
+ shader = 0;
+ }
+ return shader;
+}
+
+// Links a shader program with the given vertex and fragment shaders and
+// returns the program handle or 0 on error.
+GLuint CreateProgram(GLuint vertexShader, GLuint fragmentShader) {
+ if (vertexShader == 0 || fragmentShader == 0) {
+ return 0;
+ }
+ GLuint program = glCreateProgram();
+ if (!program) {
+ return 0;
+ }
+ glAttachShader(program, vertexShader);
+ glAttachShader(program, fragmentShader);
+ glLinkProgram(program);
+ GLint linkStatus = GL_FALSE;
+ glGetProgramiv(program, GL_LINK_STATUS, &linkStatus);
+ if (linkStatus == GL_FALSE) {
+ glDeleteProgram(program);
+ program = 0;
+ }
+ return program;
+}
+
+// When modelview and projection matrices are identity (default) the world is
+// contained in the square around origin with unit size 2. Drawing to these
+// coordinates is equivalent to drawing to the entire screen. The texture is
+// stretched over that square using texture coordinates (u, v) that range
+// from (0, 0) to (1, 1) inclusive. Texture coordinates are flipped vertically
+// here because the incoming frame has origin in upper left hand corner but
+// OpenGL expects origin in bottom left corner.
+const GLfloat gVertices[] = {
+ // X, Y, U, V.
+ -1, -1, 0, 1, // Bottom left.
+ 1, -1, 1, 1, // Bottom right.
+ 1, 1, 1, 0, // Top right.
+ -1, 1, 0, 0, // Top left.
+};
+
+// |kNumTextures| must not exceed 8, which is the limit in OpenGLES2. Two sets
+// of 3 textures are used here, one for each of the Y, U and V planes. Having
+// two sets alleviates CPU blockage in the event that the GPU is asked to render
+// to a texture that is already in use.
+static const GLsizei kNumTextureSets = 2;
+static const GLsizei kNumTextures = 3 * kNumTextureSets;
+
+@implementation RTCOpenGLVideoRenderer {
+#if TARGET_OS_IPHONE
+ EAGLContext *_context;
+#else
+ NSOpenGLContext *_context;
+#endif
+ BOOL _isInitialized;
+ NSUInteger _currentTextureSet;
+ // Handles for OpenGL constructs.
+ GLuint _textures[kNumTextures];
+ GLuint _program;
+#if !TARGET_OS_IPHONE
+ GLuint _vertexArray;
+#endif
+ GLuint _vertexBuffer;
+ GLint _position;
+ GLint _texcoord;
+ GLint _ySampler;
+ GLint _uSampler;
+ GLint _vSampler;
+ // Used to create a non-padded plane for GPU upload when we receive padded
+ // frames.
+ rtc::scoped_ptr<uint8_t[]> _planeBuffer;
+}
+
+@synthesize lastDrawnFrame = _lastDrawnFrame;
+
++ (void)initialize {
+ // Disable dithering for performance.
+ glDisable(GL_DITHER);
+}
+
+#if TARGET_OS_IPHONE
+- (instancetype)initWithContext:(EAGLContext *)context {
+#else
+- (instancetype)initWithContext:(NSOpenGLContext *)context {
+#endif
+ NSAssert(context != nil, @"context cannot be nil");
+ if (self = [super init]) {
+ _context = context;
+ }
+ return self;
+}
+
+- (BOOL)drawFrame:(RTCVideoFrame *)frame {
+ if (!_isInitialized) {
+ return NO;
+ }
+ if (_lastDrawnFrame == frame) {
+ return NO;
+ }
+ [self ensureGLContext];
+ glClear(GL_COLOR_BUFFER_BIT);
+ if (frame) {
+ if (![self updateTextureSizesForFrame:frame] ||
+ ![self updateTextureDataForFrame:frame]) {
+ return NO;
+ }
+#if !TARGET_OS_IPHONE
+ glBindVertexArray(_vertexArray);
+#endif
+ glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer);
+ glDrawArrays(GL_TRIANGLE_FAN, 0, 4);
+ }
+#if !TARGET_OS_IPHONE
+ [_context flushBuffer];
+#endif
+ _lastDrawnFrame = frame;
+ return YES;
+}
+
+- (void)setupGL {
+ if (_isInitialized) {
+ return;
+ }
+ [self ensureGLContext];
+ if (![self setupProgram]) {
+ return;
+ }
+ if (![self setupTextures]) {
+ return;
+ }
+ if (![self setupVertices]) {
+ return;
+ }
+ glUseProgram(_program);
+ glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
+ _isInitialized = YES;
+}
+
+- (void)teardownGL {
+ if (!_isInitialized) {
+ return;
+ }
+ [self ensureGLContext];
+ glDeleteProgram(_program);
+ _program = 0;
+ glDeleteTextures(kNumTextures, _textures);
+ glDeleteBuffers(1, &_vertexBuffer);
+ _vertexBuffer = 0;
+#if !TARGET_OS_IPHONE
+ glDeleteVertexArrays(1, &_vertexArray);
+#endif
+ _isInitialized = NO;
+}
+
+#pragma mark - Private
+
+- (void)ensureGLContext {
+ NSAssert(_context, @"context shouldn't be nil");
+#if TARGET_OS_IPHONE
+ if ([EAGLContext currentContext] != _context) {
+ [EAGLContext setCurrentContext:_context];
+ }
+#else
+ if ([NSOpenGLContext currentContext] != _context) {
+ [_context makeCurrentContext];
+ }
+#endif
+}
+
+- (BOOL)setupProgram {
+ NSAssert(!_program, @"program already set up");
+ GLuint vertexShader = CreateShader(GL_VERTEX_SHADER, kVertexShaderSource);
+ NSAssert(vertexShader, @"failed to create vertex shader");
+ GLuint fragmentShader =
+ CreateShader(GL_FRAGMENT_SHADER, kFragmentShaderSource);
+ NSAssert(fragmentShader, @"failed to create fragment shader");
+ _program = CreateProgram(vertexShader, fragmentShader);
+ // Shaders are created only to generate program.
+ if (vertexShader) {
+ glDeleteShader(vertexShader);
+ }
+ if (fragmentShader) {
+ glDeleteShader(fragmentShader);
+ }
+ if (!_program) {
+ return NO;
+ }
+ _position = glGetAttribLocation(_program, "position");
+ _texcoord = glGetAttribLocation(_program, "texcoord");
+ _ySampler = glGetUniformLocation(_program, "s_textureY");
+ _uSampler = glGetUniformLocation(_program, "s_textureU");
+ _vSampler = glGetUniformLocation(_program, "s_textureV");
+ if (_position < 0 || _texcoord < 0 || _ySampler < 0 || _uSampler < 0 ||
+ _vSampler < 0) {
+ return NO;
+ }
+ return YES;
+}
+
+- (BOOL)setupTextures {
+ glGenTextures(kNumTextures, _textures);
+ // Set parameters for each of the textures we created.
+ for (GLsizei i = 0; i < kNumTextures; i++) {
+ glActiveTexture(GL_TEXTURE0 + i);
+ glBindTexture(GL_TEXTURE_2D, _textures[i]);
+ glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
+ glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+ glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+ glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+ }
+ return YES;
+}
+
+- (BOOL)updateTextureSizesForFrame:(RTCVideoFrame *)frame {
+ if (frame.height == _lastDrawnFrame.height &&
+ frame.width == _lastDrawnFrame.width &&
+ frame.chromaWidth == _lastDrawnFrame.chromaWidth &&
+ frame.chromaHeight == _lastDrawnFrame.chromaHeight) {
+ return YES;
+ }
+ GLsizei lumaWidth = frame.width;
+ GLsizei lumaHeight = frame.height;
+ GLsizei chromaWidth = frame.chromaWidth;
+ GLsizei chromaHeight = frame.chromaHeight;
+ for (GLint i = 0; i < kNumTextureSets; i++) {
+ glActiveTexture(GL_TEXTURE0 + i * 3);
+ glTexImage2D(GL_TEXTURE_2D,
+ 0,
+ RTC_PIXEL_FORMAT,
+ lumaWidth,
+ lumaHeight,
+ 0,
+ RTC_PIXEL_FORMAT,
+ GL_UNSIGNED_BYTE,
+ 0);
+ glActiveTexture(GL_TEXTURE0 + i * 3 + 1);
+ glTexImage2D(GL_TEXTURE_2D,
+ 0,
+ RTC_PIXEL_FORMAT,
+ chromaWidth,
+ chromaHeight,
+ 0,
+ RTC_PIXEL_FORMAT,
+ GL_UNSIGNED_BYTE,
+ 0);
+ glActiveTexture(GL_TEXTURE0 + i * 3 + 2);
+ glTexImage2D(GL_TEXTURE_2D,
+ 0,
+ RTC_PIXEL_FORMAT,
+ chromaWidth,
+ chromaHeight,
+ 0,
+ RTC_PIXEL_FORMAT,
+ GL_UNSIGNED_BYTE,
+ 0);
+ }
+ if ((NSUInteger)frame.yPitch != frame.width ||
+ (NSUInteger)frame.uPitch != frame.chromaWidth ||
+ (NSUInteger)frame.vPitch != frame.chromaWidth) {
+ _planeBuffer.reset(new uint8_t[frame.width * frame.height]);
+ } else {
+ _planeBuffer.reset();
+ }
+ return YES;
+}
+
+- (void)uploadPlane:(const uint8_t *)plane
+ sampler:(GLint)sampler
+ offset:(NSUInteger)offset
+ width:(size_t)width
+ height:(size_t)height
+ stride:(int32_t)stride {
+ glActiveTexture(GL_TEXTURE0 + offset);
+ // When setting texture sampler uniforms, the texture index is used not
+ // the texture handle.
+ glUniform1i(sampler, offset);
+#if TARGET_OS_IPHONE
+ BOOL hasUnpackRowLength = _context.API == kEAGLRenderingAPIOpenGLES3;
+#else
+ BOOL hasUnpackRowLength = YES;
+#endif
+ const uint8_t *uploadPlane = plane;
+ if ((size_t)stride != width) {
+ if (hasUnpackRowLength) {
+ // GLES3 allows us to specify stride.
+ glPixelStorei(GL_UNPACK_ROW_LENGTH, stride);
+ glTexImage2D(GL_TEXTURE_2D,
+ 0,
+ RTC_PIXEL_FORMAT,
+ width,
+ height,
+ 0,
+ RTC_PIXEL_FORMAT,
+ GL_UNSIGNED_BYTE,
+ uploadPlane);
+ glPixelStorei(GL_UNPACK_ROW_LENGTH, 0);
+ return;
+ } else {
+ // Make an unpadded copy and upload that instead. Quick profiling showed
+ // that this is faster than uploading row by row using glTexSubImage2D.
+ uint8_t *unpaddedPlane = _planeBuffer.get();
+ for (size_t y = 0; y < height; ++y) {
+ memcpy(unpaddedPlane + y * width, plane + y * stride, width);
+ }
+ uploadPlane = unpaddedPlane;
+ }
+ }
+ glTexImage2D(GL_TEXTURE_2D,
+ 0,
+ RTC_PIXEL_FORMAT,
+ width,
+ height,
+ 0,
+ RTC_PIXEL_FORMAT,
+ GL_UNSIGNED_BYTE,
+ uploadPlane);
+}
+
+- (BOOL)updateTextureDataForFrame:(RTCVideoFrame *)frame {
+ NSUInteger textureOffset = _currentTextureSet * 3;
+ NSAssert(textureOffset + 3 <= kNumTextures, @"invalid offset");
+
+ [self uploadPlane:frame.yPlane
+ sampler:_ySampler
+ offset:textureOffset
+ width:frame.width
+ height:frame.height
+ stride:frame.yPitch];
+
+ [self uploadPlane:frame.uPlane
+ sampler:_uSampler
+ offset:textureOffset + 1
+ width:frame.chromaWidth
+ height:frame.chromaHeight
+ stride:frame.uPitch];
+
+ [self uploadPlane:frame.vPlane
+ sampler:_vSampler
+ offset:textureOffset + 2
+ width:frame.chromaWidth
+ height:frame.chromaHeight
+ stride:frame.vPitch];
+
+ _currentTextureSet = (_currentTextureSet + 1) % kNumTextureSets;
+ return YES;
+}
+
+- (BOOL)setupVertices {
+#if !TARGET_OS_IPHONE
+ NSAssert(!_vertexArray, @"vertex array already set up");
+ glGenVertexArrays(1, &_vertexArray);
+ if (!_vertexArray) {
+ return NO;
+ }
+ glBindVertexArray(_vertexArray);
+#endif
+ NSAssert(!_vertexBuffer, @"vertex buffer already set up");
+ glGenBuffers(1, &_vertexBuffer);
+ if (!_vertexBuffer) {
+#if !TARGET_OS_IPHONE
+ glDeleteVertexArrays(1, &_vertexArray);
+ _vertexArray = 0;
+#endif
+ return NO;
+ }
+ glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer);
+ glBufferData(GL_ARRAY_BUFFER, sizeof(gVertices), gVertices, GL_DYNAMIC_DRAW);
+
+ // Read position attribute from |gVertices| with size of 2 and stride of 4
+ // beginning at the start of the array. The last argument indicates offset
+ // of data within |gVertices| as supplied to the vertex buffer.
+ glVertexAttribPointer(
+ _position, 2, GL_FLOAT, GL_FALSE, 4 * sizeof(GLfloat), (void *)0);
+ glEnableVertexAttribArray(_position);
+
+ // Read texcoord attribute from |gVertices| with size of 2 and stride of 4
+ // beginning at the first texcoord in the array. The last argument indicates
+ // offset of data within |gVertices| as supplied to the vertex buffer.
+ glVertexAttribPointer(_texcoord,
+ 2,
+ GL_FLOAT,
+ GL_FALSE,
+ 4 * sizeof(GLfloat),
+ (void *)(2 * sizeof(GLfloat)));
+ glEnableVertexAttribArray(_texcoord);
+
+ return YES;
+}
+
+@end
diff --git a/webrtc/api/objc/RTCSessionDescription+Private.h b/webrtc/api/objc/RTCSessionDescription+Private.h
new file mode 100644
index 0000000..aa0314d
--- /dev/null
+++ b/webrtc/api/objc/RTCSessionDescription+Private.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCSessionDescription.h"
+
+#include "talk/app/webrtc/jsep.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCSessionDescription ()
+
+/**
+ * The native SessionDescriptionInterface representation of this
+ * RTCSessionDescription object. This is needed to pass to the underlying C++
+ * APIs.
+ */
+@property(nonatomic, readonly)
+ webrtc::SessionDescriptionInterface *nativeDescription;
+
+/**
+ * Initialize an RTCSessionDescription from a native
+ * SessionDescriptionInterface. No ownership is taken of the native session
+ * description.
+ */
+- (instancetype)initWithNativeDescription:
+ (webrtc::SessionDescriptionInterface *)nativeDescription;
+
++ (std::string)stringForType:(RTCSdpType)type;
+
++ (RTCSdpType)typeForString:(const std::string &)string;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCSessionDescription.h b/webrtc/api/objc/RTCSessionDescription.h
new file mode 100644
index 0000000..5f00b1c
--- /dev/null
+++ b/webrtc/api/objc/RTCSessionDescription.h
@@ -0,0 +1,41 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+/**
+ * Represents the session description type. This exposes the same types that are
+ * in C++, which doesn't include the rollback type that is in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCSdpType) {
+ RTCSdpTypeOffer,
+ RTCSdpTypePrAnswer,
+ RTCSdpTypeAnswer,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCSessionDescription : NSObject
+
+/** The type of session description. */
+@property(nonatomic, readonly) RTCSdpType type;
+
+/** The SDP string representation of this session description. */
+@property(nonatomic, readonly) NSString *sdp;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize a session description with a type and SDP string. */
+- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp
+ NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCSessionDescription.mm b/webrtc/api/objc/RTCSessionDescription.mm
new file mode 100644
index 0000000..7ed0760
--- /dev/null
+++ b/webrtc/api/objc/RTCSessionDescription.mm
@@ -0,0 +1,92 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCSessionDescription.h"
+
+#include "webrtc/base/checks.h"
+
+#import "webrtc/api/objc/RTCSessionDescription+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+#import "webrtc/base/objc/RTCLogging.h"
+
+@implementation RTCSessionDescription
+
+@synthesize type = _type;
+@synthesize sdp = _sdp;
+
+- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp {
+ NSParameterAssert(sdp.length);
+ if (self = [super init]) {
+ _type = type;
+ _sdp = [sdp copy];
+ }
+ return self;
+}
+
+- (NSString *)description {
+ return [NSString stringWithFormat:@"RTCSessionDescription:\n%s\n%@",
+ [[self class] stringForType:_type].c_str(),
+ _sdp];
+}
+
+#pragma mark - Private
+
+- (webrtc::SessionDescriptionInterface *)nativeDescription {
+ webrtc::SdpParseError error;
+
+ webrtc::SessionDescriptionInterface *description =
+ webrtc::CreateSessionDescription([[self class] stringForType:_type],
+ _sdp.stdString,
+ &error);
+
+ if (!description) {
+ RTCLogError(@"Failed to create session description: %s\nline: %s",
+ error.description.c_str(),
+ error.line.c_str());
+ }
+
+ return description;
+}
+
+- (instancetype)initWithNativeDescription:
+ (webrtc::SessionDescriptionInterface *)nativeDescription {
+ NSParameterAssert(nativeDescription);
+ std::string sdp;
+ nativeDescription->ToString(&sdp);
+ RTCSdpType type = [[self class] typeForString:nativeDescription->type()];
+
+ return [self initWithType:type
+ sdp:[NSString stringForStdString:sdp]];
+}
+
++ (std::string)stringForType:(RTCSdpType)type {
+ switch (type) {
+ case RTCSdpTypeOffer:
+ return webrtc::SessionDescriptionInterface::kOffer;
+ case RTCSdpTypePrAnswer:
+ return webrtc::SessionDescriptionInterface::kPrAnswer;
+ case RTCSdpTypeAnswer:
+ return webrtc::SessionDescriptionInterface::kAnswer;
+ }
+}
+
++ (RTCSdpType)typeForString:(const std::string &)string {
+ if (string == webrtc::SessionDescriptionInterface::kOffer) {
+ return RTCSdpTypeOffer;
+ } else if (string == webrtc::SessionDescriptionInterface::kPrAnswer) {
+ return RTCSdpTypePrAnswer;
+ } else if (string == webrtc::SessionDescriptionInterface::kAnswer) {
+ return RTCSdpTypeAnswer;
+ } else {
+ RTC_NOTREACHED();
+ }
+}
+
+@end
diff --git a/webrtc/api/objc/RTCStatsReport+Private.h b/webrtc/api/objc/RTCStatsReport+Private.h
new file mode 100644
index 0000000..5b7dc32
--- /dev/null
+++ b/webrtc/api/objc/RTCStatsReport+Private.h
@@ -0,0 +1,24 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCStatsReport.h"
+
+#include "talk/app/webrtc/statstypes.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCStatsReport ()
+
+/** Initialize an RTCStatsReport object from a native StatsReport. */
+- (instancetype)initWithNativeReport:(const webrtc::StatsReport &)nativeReport;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCStatsReport.h b/webrtc/api/objc/RTCStatsReport.h
new file mode 100644
index 0000000..fc66faf
--- /dev/null
+++ b/webrtc/api/objc/RTCStatsReport.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This does not currently conform to the spec. */
+@interface RTCStatsReport : NSObject
+
+/** Time since 1970-01-01T00:00:00Z in milliseconds. */
+@property(nonatomic, readonly) CFTimeInterval timestamp;
+
+/** The type of stats held by this object. */
+@property(nonatomic, readonly) NSString *type;
+
+/** The identifier for this object. */
+@property(nonatomic, readonly) NSString *statsId;
+
+/** A dictionary holding the actual stats. */
+@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *values;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCStatsReport.mm b/webrtc/api/objc/RTCStatsReport.mm
new file mode 100644
index 0000000..35a5229
--- /dev/null
+++ b/webrtc/api/objc/RTCStatsReport.mm
@@ -0,0 +1,62 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCStatsReport.h"
+
+#include "webrtc/base/checks.h"
+
+#import "webrtc/api/objc/RTCStatsReport+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+#import "webrtc/base/objc/RTCLogging.h"
+
+@implementation RTCStatsReport
+
+@synthesize timestamp = _timestamp;
+@synthesize type = _type;
+@synthesize statsId = _statsId;
+@synthesize values = _values;
+
+- (NSString *)description {
+ return [NSString stringWithFormat:@"RTCStatsReport:\n%@\n%@\n%f\n%@",
+ _statsId,
+ _type,
+ _timestamp,
+ _values];
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeReport:(const webrtc::StatsReport &)nativeReport {
+ if (self = [super init]) {
+ _timestamp = nativeReport.timestamp();
+ _type = [NSString stringForStdString:nativeReport.TypeToString()];
+ _statsId = [NSString stringForStdString:
+ nativeReport.id()->ToString()];
+
+ NSUInteger capacity = nativeReport.values().size();
+ NSMutableDictionary *values =
+ [NSMutableDictionary dictionaryWithCapacity:capacity];
+ for (auto const &valuePair : nativeReport.values()) {
+ NSString *key = [NSString stringForStdString:
+ valuePair.second->display_name()];
+ NSString *value = [NSString stringForStdString:
+ valuePair.second->ToString()];
+
+ // Not expecting duplicate keys.
+ RTC_DCHECK(values[key]);
+
+ values[key] = value;
+ }
+ _values = values;
+ }
+ return self;
+}
+
+@end
diff --git a/webrtc/api/objc/RTCVideoFrame+Private.h b/webrtc/api/objc/RTCVideoFrame+Private.h
new file mode 100644
index 0000000..954344a
--- /dev/null
+++ b/webrtc/api/objc/RTCVideoFrame+Private.h
@@ -0,0 +1,24 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoFrame.h"
+
+#include "talk/media/base/videoframe.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCVideoFrame ()
+
+- (instancetype)initWithNativeFrame:(const cricket::VideoFrame *)nativeFrame
+ NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCVideoFrame.h b/webrtc/api/objc/RTCVideoFrame.h
new file mode 100644
index 0000000..8ed23ba
--- /dev/null
+++ b/webrtc/api/objc/RTCVideoFrame.h
@@ -0,0 +1,37 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCVideoFrame : NSObject
+
+/** Width without rotation applied. */
+@property(nonatomic, readonly) size_t width;
+
+/** Height without rotation applied. */
+@property(nonatomic, readonly) size_t height;
+@property(nonatomic, readonly) size_t chromaWidth;
+@property(nonatomic, readonly) size_t chromaHeight;
+@property(nonatomic, readonly) size_t chromaSize;
+// These can return NULL if the object is not backed by a buffer.
+@property(nonatomic, readonly, nullable) const uint8_t *yPlane;
+@property(nonatomic, readonly, nullable) const uint8_t *uPlane;
+@property(nonatomic, readonly, nullable) const uint8_t *vPlane;
+@property(nonatomic, readonly) int32_t yPitch;
+@property(nonatomic, readonly) int32_t uPitch;
+@property(nonatomic, readonly) int32_t vPitch;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/RTCVideoFrame.mm b/webrtc/api/objc/RTCVideoFrame.mm
new file mode 100644
index 0000000..db2d07b
--- /dev/null
+++ b/webrtc/api/objc/RTCVideoFrame.mm
@@ -0,0 +1,79 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoFrame.h"
+
+#include "webrtc/base/scoped_ptr.h"
+
+#import "webrtc/api/objc/RTCVideoFrame+Private.h"
+
+@implementation RTCVideoFrame {
+ rtc::scoped_ptr<cricket::VideoFrame> _videoFrame;
+}
+
+- (size_t)width {
+ return _videoFrame->GetWidth();
+}
+
+- (size_t)height {
+ return _videoFrame->GetHeight();
+}
+
+- (size_t)chromaWidth {
+ return _videoFrame->GetChromaWidth();
+}
+
+- (size_t)chromaHeight {
+ return _videoFrame->GetChromaHeight();
+}
+
+- (size_t)chromaSize {
+ return _videoFrame->GetChromaSize();
+}
+
+- (const uint8_t *)yPlane {
+ const cricket::VideoFrame *const_frame = _videoFrame.get();
+ return const_frame->GetYPlane();
+}
+
+- (const uint8_t *)uPlane {
+ const cricket::VideoFrame *const_frame = _videoFrame.get();
+ return const_frame->GetUPlane();
+}
+
+- (const uint8_t *)vPlane {
+ const cricket::VideoFrame *const_frame = _videoFrame.get();
+ return const_frame->GetVPlane();
+}
+
+- (int32_t)yPitch {
+ return _videoFrame->GetYPitch();
+}
+
+- (int32_t)uPitch {
+ return _videoFrame->GetUPitch();
+}
+
+- (int32_t)vPitch {
+ return _videoFrame->GetVPitch();
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeFrame:(const cricket::VideoFrame *)nativeFrame {
+ if (self = [super init]) {
+ // Keep a shallow copy of the video frame. The underlying frame buffer is
+ // not copied.
+ _videoFrame.reset(nativeFrame->Copy());
+ }
+ return self;
+}
+
+@end
diff --git a/webrtc/api/objc/RTCVideoRenderer.h b/webrtc/api/objc/RTCVideoRenderer.h
new file mode 100644
index 0000000..a974562
--- /dev/null
+++ b/webrtc/api/objc/RTCVideoRenderer.h
@@ -0,0 +1,30 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#if TARGET_OS_IPHONE
+#import <UIKit/UIKit.h>
+#endif
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTCVideoFrame;
+
+@protocol RTCVideoRenderer <NSObject>
+
+/** The size of the frame. */
+- (void)setSize:(CGSize)size;
+
+/** The frame to be displayed. */
+- (void)renderFrame:(RTCVideoFrame *)frame;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/api/objc/WebRTC-Prefix.pch b/webrtc/api/objc/WebRTC-Prefix.pch
new file mode 100644
index 0000000..990b160
--- /dev/null
+++ b/webrtc/api/objc/WebRTC-Prefix.pch
@@ -0,0 +1,13 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if !defined(__has_feature) || !__has_feature(objc_arc)
+#error "This file requires ARC support."
+#endif
diff --git a/webrtc/api/objctests/RTCIceCandidateTest.mm b/webrtc/api/objctests/RTCIceCandidateTest.mm
new file mode 100644
index 0000000..391db44
--- /dev/null
+++ b/webrtc/api/objctests/RTCIceCandidateTest.mm
@@ -0,0 +1,74 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include "webrtc/base/gunit.h"
+
+#import "webrtc/api/objc/RTCIceCandidate.h"
+#import "webrtc/api/objc/RTCIceCandidate+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@interface RTCIceCandidateTest : NSObject
+- (void)testCandidate;
+- (void)testInitFromNativeCandidate;
+@end
+
+@implementation RTCIceCandidateTest
+
+- (void)testCandidate {
+ NSString *sdp = @"candidate:4025901590 1 udp 2122265343 "
+ "fdff:2642:12a6:fe38:c001:beda:fcf9:51aa "
+ "59052 typ host generation 0";
+
+ RTCIceCandidate *candidate = [[RTCIceCandidate alloc] initWithSdp:sdp
+ sdpMLineIndex:0
+ sdpMid:@"audio"];
+
+ rtc::scoped_ptr<webrtc::IceCandidateInterface> nativeCandidate =
+ candidate.nativeCandidate;
+ EXPECT_EQ("audio", nativeCandidate->sdp_mid());
+ EXPECT_EQ(0, nativeCandidate->sdp_mline_index());
+
+ std::string sdpString;
+ nativeCandidate->ToString(&sdpString);
+ EXPECT_EQ(sdp.stdString, sdpString);
+}
+
+- (void)testInitFromNativeCandidate {
+ std::string sdp("candidate:4025901590 1 udp 2122265343 "
+ "fdff:2642:12a6:fe38:c001:beda:fcf9:51aa "
+ "59052 typ host generation 0");
+ webrtc::IceCandidateInterface *nativeCandidate =
+ webrtc::CreateIceCandidate("audio", 0, sdp, nullptr);
+
+ RTCIceCandidate *iceCandidate =
+ [[RTCIceCandidate alloc] initWithNativeCandidate:nativeCandidate];
+ EXPECT_TRUE([@"audio" isEqualToString:iceCandidate.sdpMid]);
+ EXPECT_EQ(0, iceCandidate.sdpMLineIndex);
+
+ EXPECT_EQ(sdp, iceCandidate.sdp.stdString);
+}
+
+@end
+
+TEST(RTCIceCandidateTest, CandidateTest) {
+ @autoreleasepool {
+ RTCIceCandidateTest *test = [[RTCIceCandidateTest alloc] init];
+ [test testCandidate];
+ }
+}
+
+TEST(RTCIceCandidateTest, InitFromCandidateTest) {
+ @autoreleasepool {
+ RTCIceCandidateTest *test = [[RTCIceCandidateTest alloc] init];
+ [test testInitFromNativeCandidate];
+ }
+}
diff --git a/webrtc/api/objctests/RTCIceServerTest.mm b/webrtc/api/objctests/RTCIceServerTest.mm
new file mode 100644
index 0000000..5fa43f8
--- /dev/null
+++ b/webrtc/api/objctests/RTCIceServerTest.mm
@@ -0,0 +1,84 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include <vector>
+
+#include "webrtc/base/gunit.h"
+
+#import "webrtc/api/objc/RTCIceServer.h"
+#import "webrtc/api/objc/RTCIceServer+Private.h"
+
+@interface RTCIceServerTest : NSObject
+- (void)testOneURLServer;
+- (void)testTwoURLServer;
+- (void)testPasswordCredential;
+@end
+
+@implementation RTCIceServerTest
+
+- (void)testOneURLServer {
+ RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:@[
+ @"stun:stun1.example.net" ]];
+
+ webrtc::PeerConnectionInterface::IceServer iceStruct = server.iceServer;
+ EXPECT_EQ((size_t)1, iceStruct.urls.size());
+ EXPECT_EQ("stun:stun1.example.net", iceStruct.urls.front());
+ EXPECT_EQ("", iceStruct.username);
+ EXPECT_EQ("", iceStruct.password);
+}
+
+- (void)testTwoURLServer {
+ RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:@[
+ @"turn1:turn1.example.net", @"turn2:turn2.example.net" ]];
+
+ webrtc::PeerConnectionInterface::IceServer iceStruct = server.iceServer;
+ EXPECT_EQ((size_t)2, iceStruct.urls.size());
+ EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
+ EXPECT_EQ("turn2:turn2.example.net", iceStruct.urls.back());
+ EXPECT_EQ("", iceStruct.username);
+ EXPECT_EQ("", iceStruct.password);
+}
+
+- (void)testPasswordCredential {
+ RTCIceServer *server = [[RTCIceServer alloc]
+ initWithURLStrings:@[ @"turn1:turn1.example.net" ]
+ username:@"username"
+ credential:@"credential"];
+ webrtc::PeerConnectionInterface::IceServer iceStruct = server.iceServer;
+ EXPECT_EQ((size_t)1, iceStruct.urls.size());
+ EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
+ EXPECT_EQ("username", iceStruct.username);
+ EXPECT_EQ("credential", iceStruct.password);
+}
+
+@end
+
+TEST(RTCIceServerTest, OneURLTest) {
+ @autoreleasepool {
+ RTCIceServerTest *test = [[RTCIceServerTest alloc] init];
+ [test testOneURLServer];
+ }
+}
+
+TEST(RTCIceServerTest, TwoURLTest) {
+ @autoreleasepool {
+ RTCIceServerTest *test = [[RTCIceServerTest alloc] init];
+ [test testTwoURLServer];
+ }
+}
+
+TEST(RTCIceServerTest, PasswordCredentialTest) {
+ @autoreleasepool {
+ RTCIceServerTest *test = [[RTCIceServerTest alloc] init];
+ [test testPasswordCredential];
+ }
+}
diff --git a/webrtc/api/objctests/RTCMediaConstraintsTest.mm b/webrtc/api/objctests/RTCMediaConstraintsTest.mm
new file mode 100644
index 0000000..44ffe3d
--- /dev/null
+++ b/webrtc/api/objctests/RTCMediaConstraintsTest.mm
@@ -0,0 +1,66 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include "webrtc/base/gunit.h"
+
+#import "webrtc/api/objc/RTCMediaConstraints.h"
+#import "webrtc/api/objc/RTCMediaConstraints+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@interface RTCMediaConstraintsTest : NSObject
+- (void)testMediaConstraints;
+@end
+
+@implementation RTCMediaConstraintsTest
+
+- (void)testMediaConstraints {
+ NSDictionary *mandatory = @{@"key1": @"value1", @"key2": @"value2"};
+ NSDictionary *optional = @{@"key3": @"value3", @"key4": @"value4"};
+
+ RTCMediaConstraints *constraints = [[RTCMediaConstraints alloc]
+ initWithMandatoryConstraints:mandatory
+ optionalConstraints:optional];
+ rtc::scoped_ptr<webrtc::MediaConstraints> nativeConstraints =
+ [constraints nativeConstraints];
+
+ webrtc::MediaConstraintsInterface::Constraints nativeMandatory =
+ nativeConstraints->GetMandatory();
+ [self expectConstraints:mandatory inNativeConstraints:nativeMandatory];
+
+ webrtc::MediaConstraintsInterface::Constraints nativeOptional =
+ nativeConstraints->GetOptional();
+ [self expectConstraints:optional inNativeConstraints:nativeOptional];
+}
+
+- (void)expectConstraints:(NSDictionary *)constraints
+ inNativeConstraints:
+ (webrtc::MediaConstraintsInterface::Constraints)nativeConstraints {
+ EXPECT_EQ(constraints.count, nativeConstraints.size());
+
+ for (NSString *key in constraints) {
+ NSString *value = constraints[key];
+
+ std::string nativeValue;
+ bool found = nativeConstraints.FindFirst(key.stdString, &nativeValue);
+ EXPECT_TRUE(found);
+ EXPECT_EQ(value.stdString, nativeValue);
+ }
+}
+
+@end
+
+TEST(RTCMediaConstraintsTest, MediaConstraintsTest) {
+ @autoreleasepool {
+ RTCMediaConstraintsTest *test = [[RTCMediaConstraintsTest alloc] init];
+ [test testMediaConstraints];
+ }
+}
diff --git a/webrtc/api/objctests/RTCSessionDescriptionTest.mm b/webrtc/api/objctests/RTCSessionDescriptionTest.mm
new file mode 100644
index 0000000..2404ded
--- /dev/null
+++ b/webrtc/api/objctests/RTCSessionDescriptionTest.mm
@@ -0,0 +1,144 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include "webrtc/base/gunit.h"
+
+#import "webrtc/api/objc/RTCSessionDescription.h"
+#import "webrtc/api/objc/RTCSessionDescription+Private.h"
+#import "webrtc/base/objc/NSString+StdString.h"
+
+@interface RTCSessionDescriptionTest : NSObject
+- (void)testSessionDescriptionConversion;
+- (void)testInitFromNativeSessionDescription;
+@end
+
+@implementation RTCSessionDescriptionTest
+
+/**
+ * Test conversion of an Objective-C RTCSessionDescription to a native
+ * SessionDescriptionInterface (based on the types and SDP strings being equal).
+ */
+- (void)testSessionDescriptionConversion {
+ RTCSessionDescription *description =
+ [[RTCSessionDescription alloc] initWithType:RTCSdpTypeAnswer
+ sdp:[self sdp]];
+
+ webrtc::SessionDescriptionInterface *nativeDescription =
+ description.nativeDescription;
+
+ EXPECT_EQ(RTCSdpTypeAnswer,
+ [RTCSessionDescription typeForString:nativeDescription->type()]);
+
+ std::string sdp;
+ nativeDescription->ToString(&sdp);
+ EXPECT_EQ([self sdp].stdString, sdp);
+}
+
+- (void)testInitFromNativeSessionDescription {
+ webrtc::SessionDescriptionInterface *nativeDescription;
+
+ nativeDescription = webrtc::CreateSessionDescription(
+ webrtc::SessionDescriptionInterface::kAnswer,
+ [self sdp].stdString,
+ nullptr);
+
+ RTCSessionDescription *description =
+ [[RTCSessionDescription alloc] initWithNativeDescription:
+ nativeDescription];
+ EXPECT_EQ(webrtc::SessionDescriptionInterface::kAnswer,
+ [RTCSessionDescription stringForType:description.type]);
+ EXPECT_TRUE([[self sdp] isEqualToString:description.sdp]);
+}
+
+- (NSString *)sdp {
+ return @"v=0\r\n"
+ "o=- 5319989746393411314 2 IN IP4 127.0.0.1\r\n"
+ "s=-\r\n"
+ "t=0 0\r\n"
+ "a=group:BUNDLE audio video\r\n"
+ "a=msid-semantic: WMS ARDAMS\r\n"
+ "m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9 0 8 126\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
+ "a=rtcp:9 IN IP4 0.0.0.0\r\n"
+ "a=ice-ufrag:f3o+0HG7l9nwIWFY\r\n"
+ "a=ice-pwd:VDctmJNCptR2TB7+meDpw7w5\r\n"
+ "a=fingerprint:sha-256 A9:D5:8D:A8:69:22:39:60:92:AD:94:1A:22:2D:5E:"
+ "A5:4A:A9:18:C2:35:5D:46:5E:59:BD:1C:AF:38:9F:E6:E1\r\n"
+ "a=setup:active\r\n"
+ "a=mid:audio\r\n"
+ "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\n"
+ "a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/"
+ "abs-send-time\r\n"
+ "a=sendrecv\r\n"
+ "a=rtcp-mux\r\n"
+ "a=rtpmap:111 opus/48000/2\r\n"
+ "a=fmtp:111 minptime=10; useinbandfec=1\r\n"
+ "a=rtpmap:103 ISAC/16000\r\n"
+ "a=rtpmap:9 G722/8000\r\n"
+ "a=rtpmap:0 PCMU/8000\r\n"
+ "a=rtpmap:8 PCMA/8000\r\n"
+ "a=rtpmap:126 telephone-event/8000\r\n"
+ "a=maxptime:60\r\n"
+ "a=ssrc:1504474588 cname:V+FdIC5AJpxLhdYQ\r\n"
+ "a=ssrc:1504474588 msid:ARDAMS ARDAMSa0\r\n"
+ "a=ssrc:1504474588 mslabel:ARDAMS\r\n"
+ "a=ssrc:1504474588 label:ARDAMSa0\r\n"
+ "m=video 9 UDP/TLS/RTP/SAVPF 100 116 117 96\r\n"
+ "c=IN IP4 0.0.0.0\r\n"
+ "a=rtcp:9 IN IP4 0.0.0.0\r\n"
+ "a=ice-ufrag:f3o+0HG7l9nwIWFY\r\n"
+ "a=ice-pwd:VDctmJNCptR2TB7+meDpw7w5\r\n"
+ "a=fingerprint:sha-256 A9:D5:8D:A8:69:22:39:60:92:AD:94:1A:22:2D:5E:"
+ "A5:4A:A9:18:C2:35:5D:46:5E:59:BD:1C:AF:38:9F:E6:E1\r\n"
+ "a=setup:active\r\n"
+ "a=mid:video\r\n"
+ "a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\n"
+ "a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/"
+ "abs-send-time\r\n"
+ "a=extmap:4 urn:3gpp:video-orientation\r\n"
+ "a=sendrecv\r\n"
+ "a=rtcp-mux\r\n"
+ "a=rtpmap:100 VP8/90000\r\n"
+ "a=rtcp-fb:100 ccm fir\r\n"
+ "a=rtcp-fb:100 nack\r\n"
+ "a=rtcp-fb:100 nack pli\r\n"
+ "a=rtcp-fb:100 goog-remb\r\n"
+ "a=rtpmap:116 red/90000\r\n"
+ "a=rtpmap:117 ulpfec/90000\r\n"
+ "a=rtpmap:96 rtx/90000\r\n"
+ "a=fmtp:96 apt=100\r\n"
+ "a=ssrc-group:FID 498297514 1644357692\r\n"
+ "a=ssrc:498297514 cname:V+FdIC5AJpxLhdYQ\r\n"
+ "a=ssrc:498297514 msid:ARDAMS ARDAMSv0\r\n"
+ "a=ssrc:498297514 mslabel:ARDAMS\r\n"
+ "a=ssrc:498297514 label:ARDAMSv0\r\n"
+ "a=ssrc:1644357692 cname:V+FdIC5AJpxLhdYQ\r\n"
+ "a=ssrc:1644357692 msid:ARDAMS ARDAMSv0\r\n"
+ "a=ssrc:1644357692 mslabel:ARDAMS\r\n"
+ "a=ssrc:1644357692 label:ARDAMSv0\r\n";
+}
+
+@end
+
+TEST(RTCSessionDescriptionTest, SessionDescriptionConversionTest) {
+ @autoreleasepool {
+ RTCSessionDescriptionTest *test = [[RTCSessionDescriptionTest alloc] init];
+ [test testSessionDescriptionConversion];
+ }
+}
+
+TEST(RTCSessionDescriptionTest, InitFromSessionDescriptionTest) {
+ @autoreleasepool {
+ RTCSessionDescriptionTest *test = [[RTCSessionDescriptionTest alloc] init];
+ [test testInitFromNativeSessionDescription];
+ }
+}
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index d5061db..5a9902e 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -14,6 +14,9 @@
"audio_receive_stream.h",
"audio_send_stream.cc",
"audio_send_stream.h",
+ "audio_sink.h",
+ "audio_state.cc",
+ "audio_state.h",
"conversion.h",
"scoped_voe_interface.h",
]
@@ -29,7 +32,7 @@
deps = [
"..:webrtc_common",
- "../voice_engine",
"../system_wrappers",
+ "../voice_engine",
]
}
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 34197c3..64d0083 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -11,20 +11,41 @@
#include "webrtc/audio/audio_receive_stream.h"
#include <string>
+#include <utility>
+#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
+#include "webrtc/call/congestion_controller.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_neteq_stats.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
+namespace {
+
+bool UseSendSideBwe(const webrtc::AudioReceiveStream::Config& config) {
+ if (!config.rtp.transport_cc) {
+ return false;
+ }
+ for (const auto& extension : config.rtp.extensions) {
+ if (extension.name == RtpExtension::kTransportSequenceNumber) {
+ return true;
+ }
+ }
+ return false;
+}
+} // namespace
+
std::string AudioReceiveStream::Config::Rtp::ToString() const {
std::stringstream ss;
ss << "{remote_ssrc: " << remote_ssrc;
@@ -60,120 +81,62 @@
namespace internal {
AudioReceiveStream::AudioReceiveStream(
- RemoteBitrateEstimator* remote_bitrate_estimator,
- const webrtc::AudioReceiveStream::Config& config,
- VoiceEngine* voice_engine)
- : remote_bitrate_estimator_(remote_bitrate_estimator),
- config_(config),
- voice_engine_(voice_engine),
- voe_base_(voice_engine),
+ CongestionController* congestion_controller,
+ const webrtc::AudioReceiveStream::Config& config,
+ const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
+ : config_(config),
+ audio_state_(audio_state),
rtp_header_parser_(RtpHeaderParser::Create()) {
LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
- RTC_DCHECK(config.voe_channel_id != -1);
- RTC_DCHECK(remote_bitrate_estimator_ != nullptr);
- RTC_DCHECK(voice_engine_ != nullptr);
- RTC_DCHECK(rtp_header_parser_ != nullptr);
- for (const auto& ext : config.rtp.extensions) {
- // One-byte-extension local identifiers are in the range 1-14 inclusive.
- RTC_DCHECK_GE(ext.id, 1);
- RTC_DCHECK_LE(ext.id, 14);
- if (ext.name == RtpExtension::kAudioLevel) {
- RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
- kRtpExtensionAudioLevel, ext.id));
- } else if (ext.name == RtpExtension::kAbsSendTime) {
- RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
- kRtpExtensionAbsoluteSendTime, ext.id));
- } else if (ext.name == RtpExtension::kTransportSequenceNumber) {
- RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
- kRtpExtensionTransportSequenceNumber, ext.id));
+ RTC_DCHECK_NE(config_.voe_channel_id, -1);
+ RTC_DCHECK(audio_state_.get());
+ RTC_DCHECK(congestion_controller);
+ RTC_DCHECK(rtp_header_parser_);
+
+ VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
+ channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+ channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
+ for (const auto& extension : config.rtp.extensions) {
+ if (extension.name == RtpExtension::kAudioLevel) {
+ channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
+ bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionAudioLevel, extension.id);
+ RTC_DCHECK(registered);
+ } else if (extension.name == RtpExtension::kAbsSendTime) {
+ channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id);
+ bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionAbsoluteSendTime, extension.id);
+ RTC_DCHECK(registered);
+ } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
+ bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionTransportSequenceNumber, extension.id);
+ RTC_DCHECK(registered);
} else {
RTC_NOTREACHED() << "Unsupported RTP extension.";
}
}
+ // Configure bandwidth estimation.
+ channel_proxy_->SetCongestionControlObjects(
+ nullptr, nullptr, congestion_controller->packet_router());
+ if (config.combined_audio_video_bwe) {
+ if (UseSendSideBwe(config)) {
+ remote_bitrate_estimator_ =
+ congestion_controller->GetRemoteBitrateEstimator(true);
+ } else {
+ remote_bitrate_estimator_ =
+ congestion_controller->GetRemoteBitrateEstimator(false);
+ }
+ RTC_DCHECK(remote_bitrate_estimator_);
+ }
}
AudioReceiveStream::~AudioReceiveStream() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
-}
-
-webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- webrtc::AudioReceiveStream::Stats stats;
- stats.remote_ssrc = config_.rtp.remote_ssrc;
- ScopedVoEInterface<VoECodec> codec(voice_engine_);
- ScopedVoEInterface<VoENetEqStats> neteq(voice_engine_);
- ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_);
- ScopedVoEInterface<VoEVideoSync> sync(voice_engine_);
- ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_);
- unsigned int ssrc = 0;
- webrtc::CallStatistics call_stats = {0};
- webrtc::CodecInst codec_inst = {0};
- // Only collect stats if we have seen some traffic with the SSRC.
- if (rtp->GetRemoteSSRC(config_.voe_channel_id, ssrc) == -1 ||
- rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1 ||
- codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
- return stats;
+ channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
+ if (remote_bitrate_estimator_) {
+ remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
}
-
- stats.bytes_rcvd = call_stats.bytesReceived;
- stats.packets_rcvd = call_stats.packetsReceived;
- stats.packets_lost = call_stats.cumulativeLost;
- stats.fraction_lost = Q8ToFloat(call_stats.fractionLost);
- if (codec_inst.pltype != -1) {
- stats.codec_name = codec_inst.plname;
- }
- stats.ext_seqnum = call_stats.extendedMax;
- if (codec_inst.plfreq / 1000 > 0) {
- stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000);
- }
- {
- int jitter_buffer_delay_ms = 0;
- int playout_buffer_delay_ms = 0;
- sync->GetDelayEstimate(config_.voe_channel_id, &jitter_buffer_delay_ms,
- &playout_buffer_delay_ms);
- stats.delay_estimate_ms =
- jitter_buffer_delay_ms + playout_buffer_delay_ms;
- }
- {
- unsigned int level = 0;
- if (volume->GetSpeechOutputLevelFullRange(config_.voe_channel_id, level)
- != -1) {
- stats.audio_level = static_cast<int32_t>(level);
- }
- }
-
- webrtc::NetworkStatistics ns = {0};
- if (neteq->GetNetworkStatistics(config_.voe_channel_id, ns) != -1) {
- // Get jitter buffer and total delay (alg + jitter + playout) stats.
- stats.jitter_buffer_ms = ns.currentBufferSize;
- stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
- stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
- stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
- stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
- stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
- stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
- }
-
- webrtc::AudioDecodingCallStats ds;
- if (neteq->GetDecodingCallStatistics(config_.voe_channel_id, &ds) != -1) {
- stats.decoding_calls_to_silence_generator =
- ds.calls_to_silence_generator;
- stats.decoding_calls_to_neteq = ds.calls_to_neteq;
- stats.decoding_normal = ds.decoded_normal;
- stats.decoding_plc = ds.decoded_plc;
- stats.decoding_cng = ds.decoded_cng;
- stats.decoding_plc_cng = ds.decoded_plc_cng;
- }
-
- stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
-
- return stats;
-}
-
-const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- return config_;
}
void AudioReceiveStream::Start() {
@@ -204,15 +167,16 @@
// thread. Then this check can be enabled.
// RTC_DCHECK(!thread_checker_.CalledOnValidThread());
RTPHeader header;
-
if (!rtp_header_parser_->Parse(packet, length, &header)) {
return false;
}
- // Only forward if the parsed header has absolute sender time. RTP timestamps
- // may have different rates for audio and video and shouldn't be mixed.
- if (config_.combined_audio_video_bwe &&
- header.extension.hasAbsoluteSendTime) {
+ // Only forward if the parsed header has one of the headers necessary for
+ // bandwidth estimation. RTP timestamps has different rates for audio and
+ // video and shouldn't be mixed.
+ if (remote_bitrate_estimator_ &&
+ (header.extension.hasAbsoluteSendTime ||
+ header.extension.hasTransportSequenceNumber)) {
int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
if (packet_time.timestamp >= 0)
arrival_time_ms = (packet_time.timestamp + 500) / 1000;
@@ -222,5 +186,71 @@
}
return true;
}
+
+webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ webrtc::AudioReceiveStream::Stats stats;
+ stats.remote_ssrc = config_.rtp.remote_ssrc;
+ ScopedVoEInterface<VoECodec> codec(voice_engine());
+
+ webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
+ webrtc::CodecInst codec_inst = {0};
+ if (codec->GetRecCodec(config_.voe_channel_id, codec_inst) == -1) {
+ return stats;
+ }
+
+ stats.bytes_rcvd = call_stats.bytesReceived;
+ stats.packets_rcvd = call_stats.packetsReceived;
+ stats.packets_lost = call_stats.cumulativeLost;
+ stats.fraction_lost = Q8ToFloat(call_stats.fractionLost);
+ stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
+ if (codec_inst.pltype != -1) {
+ stats.codec_name = codec_inst.plname;
+ }
+ stats.ext_seqnum = call_stats.extendedMax;
+ if (codec_inst.plfreq / 1000 > 0) {
+ stats.jitter_ms = call_stats.jitterSamples / (codec_inst.plfreq / 1000);
+ }
+ stats.delay_estimate_ms = channel_proxy_->GetDelayEstimate();
+ stats.audio_level = channel_proxy_->GetSpeechOutputLevelFullRange();
+
+ // Get jitter buffer and total delay (alg + jitter + playout) stats.
+ auto ns = channel_proxy_->GetNetworkStatistics();
+ stats.jitter_buffer_ms = ns.currentBufferSize;
+ stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
+ stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
+ stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
+ stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
+ stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
+ stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
+
+ auto ds = channel_proxy_->GetDecodingCallStatistics();
+ stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
+ stats.decoding_calls_to_neteq = ds.calls_to_neteq;
+ stats.decoding_normal = ds.decoded_normal;
+ stats.decoding_plc = ds.decoded_plc;
+ stats.decoding_cng = ds.decoded_cng;
+ stats.decoding_plc_cng = ds.decoded_plc_cng;
+
+ return stats;
+}
+
+void AudioReceiveStream::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ channel_proxy_->SetSink(std::move(sink));
+}
+
+const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ return config_;
+}
+
+VoiceEngine* AudioReceiveStream::voice_engine() const {
+ internal::AudioState* audio_state =
+ static_cast<internal::AudioState*>(audio_state_.get());
+ VoiceEngine* voice_engine = audio_state->voice_engine();
+ RTC_DCHECK(voice_engine);
+ return voice_engine;
+}
} // namespace internal
} // namespace webrtc
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 5d02b0e..4940c6a 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -12,23 +12,25 @@
#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
#include "webrtc/audio_receive_stream.h"
-#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/audio_state.h"
#include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
namespace webrtc {
-
+class CongestionController;
class RemoteBitrateEstimator;
-class VoiceEngine;
+
+namespace voe {
+class ChannelProxy;
+} // namespace voe
namespace internal {
class AudioReceiveStream final : public webrtc::AudioReceiveStream {
public:
- AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
+ AudioReceiveStream(CongestionController* congestion_controller,
const webrtc::AudioReceiveStream::Config& config,
- VoiceEngine* voice_engine);
+ const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
~AudioReceiveStream() override;
// webrtc::ReceiveStream implementation.
@@ -43,16 +45,19 @@
// webrtc::AudioReceiveStream implementation.
webrtc::AudioReceiveStream::Stats GetStats() const override;
+ void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) override;
+
const webrtc::AudioReceiveStream::Config& config() const;
private:
+ VoiceEngine* voice_engine() const;
+
rtc::ThreadChecker thread_checker_;
- RemoteBitrateEstimator* const remote_bitrate_estimator_;
+ RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
const webrtc::AudioReceiveStream::Config config_;
- VoiceEngine* voice_engine_;
- // We hold one interface pointer to the VoE to make sure it is kept alive.
- ScopedVoEInterface<VoEBase> voe_base_;
+ rtc::scoped_refptr<webrtc::AudioState> audio_state_;
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
+ rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
};
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 4e267f1..eb008b3 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -8,154 +8,320 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <string>
+
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_receive_stream.h"
#include "webrtc/audio/conversion.h"
+#include "webrtc/call/mock/mock_congestion_controller.h"
+#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
+#include "webrtc/modules/pacing/packet_router.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/test/fake_voice_engine.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/test/mock_voe_channel_proxy.h"
+#include "webrtc/test/mock_voice_engine.h"
+#include "webrtc/video/call_stats.h"
+namespace webrtc {
+namespace test {
namespace {
-using webrtc::ByteWriter;
+using testing::_;
+using testing::Return;
-const size_t kAbsoluteSendTimeLength = 4;
-
-void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
- int id,
- uint32_t abs_send_time) {
- const size_t kRtpOneByteHeaderLength = 4;
- const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
- ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId);
-
- const uint32_t kPosLength = 2;
- ByteWriter<uint16_t>::WriteBigEndian(buffer + kPosLength,
- kAbsoluteSendTimeLength / 4);
-
- const uint8_t kLengthOfData = 3;
- buffer[kRtpOneByteHeaderLength] = (id << 4) + (kLengthOfData - 1);
- ByteWriter<uint32_t, kLengthOfData>::WriteBigEndian(
- buffer + kRtpOneByteHeaderLength + 1, abs_send_time);
+AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
+ AudioDecodingCallStats audio_decode_stats;
+ audio_decode_stats.calls_to_silence_generator = 234;
+ audio_decode_stats.calls_to_neteq = 567;
+ audio_decode_stats.decoded_normal = 890;
+ audio_decode_stats.decoded_plc = 123;
+ audio_decode_stats.decoded_cng = 456;
+ audio_decode_stats.decoded_plc_cng = 789;
+ return audio_decode_stats;
}
-size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
- int extension_id,
- uint32_t abs_send_time) {
+const int kChannelId = 2;
+const uint32_t kRemoteSsrc = 1234;
+const uint32_t kLocalSsrc = 5678;
+const size_t kOneByteExtensionHeaderLength = 4;
+const size_t kOneByteExtensionLength = 4;
+const int kAbsSendTimeId = 2;
+const int kAudioLevelId = 3;
+const int kTransportSequenceNumberId = 4;
+const int kJitterBufferDelay = -7;
+const int kPlayoutBufferDelay = 302;
+const unsigned int kSpeechOutputLevel = 99;
+const CallStatistics kCallStats = {
+ 345, 678, 901, 234, -12, 3456, 7890, 567, 890, 123};
+const CodecInst kCodecInst = {
+ 123, "codec_name_recv", 96000, -187, 0, -103};
+const NetworkStatistics kNetworkStats = {
+ 123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0};
+const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
+
+struct ConfigHelper {
+ ConfigHelper()
+ : simulated_clock_(123456),
+ call_stats_(&simulated_clock_),
+ congestion_controller_(&process_thread_,
+ &call_stats_,
+ &bitrate_observer_) {
+ using testing::Invoke;
+
+ EXPECT_CALL(voice_engine_,
+ RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
+ EXPECT_CALL(voice_engine_,
+ DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
+ AudioState::Config config;
+ config.voice_engine = &voice_engine_;
+ audio_state_ = AudioState::Create(config);
+
+ EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
+ .WillOnce(Invoke([this](int channel_id) {
+ EXPECT_FALSE(channel_proxy_);
+ channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
+ EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
+ EXPECT_CALL(*channel_proxy_,
+ SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
+ .Times(1);
+ EXPECT_CALL(*channel_proxy_,
+ SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
+ .Times(1);
+ EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects(
+ nullptr, nullptr, &packet_router_))
+ .Times(1);
+ EXPECT_CALL(congestion_controller_, packet_router())
+ .WillOnce(Return(&packet_router_));
+ EXPECT_CALL(*channel_proxy_,
+ SetCongestionControlObjects(nullptr, nullptr, nullptr))
+ .Times(1);
+ return channel_proxy_;
+ }));
+ stream_config_.voe_channel_id = kChannelId;
+ stream_config_.rtp.local_ssrc = kLocalSsrc;
+ stream_config_.rtp.remote_ssrc = kRemoteSsrc;
+ stream_config_.rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+ stream_config_.rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
+ }
+
+ MockCongestionController* congestion_controller() {
+ return &congestion_controller_;
+ }
+ MockRemoteBitrateEstimator* remote_bitrate_estimator() {
+ return &remote_bitrate_estimator_;
+ }
+ AudioReceiveStream::Config& config() { return stream_config_; }
+ rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
+ MockVoiceEngine& voice_engine() { return voice_engine_; }
+
+ void SetupMockForBweFeedback(bool send_side_bwe) {
+ EXPECT_CALL(congestion_controller_,
+ GetRemoteBitrateEstimator(send_side_bwe))
+ .WillOnce(Return(&remote_bitrate_estimator_));
+ EXPECT_CALL(remote_bitrate_estimator_,
+ RemoveStream(stream_config_.rtp.remote_ssrc));
+ }
+
+ void SetupMockForGetStats() {
+ using testing::DoAll;
+ using testing::SetArgReferee;
+
+ ASSERT_TRUE(channel_proxy_);
+ EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
+ .WillOnce(Return(kCallStats));
+ EXPECT_CALL(*channel_proxy_, GetDelayEstimate())
+ .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
+ EXPECT_CALL(*channel_proxy_, GetSpeechOutputLevelFullRange())
+ .WillOnce(Return(kSpeechOutputLevel));
+ EXPECT_CALL(*channel_proxy_, GetNetworkStatistics())
+ .WillOnce(Return(kNetworkStats));
+ EXPECT_CALL(*channel_proxy_, GetDecodingCallStatistics())
+ .WillOnce(Return(kAudioDecodeStats));
+
+ EXPECT_CALL(voice_engine_, GetRecCodec(kChannelId, _))
+ .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
+ }
+
+ private:
+ SimulatedClock simulated_clock_;
+ CallStats call_stats_;
+ PacketRouter packet_router_;
+ testing::NiceMock<MockBitrateObserver> bitrate_observer_;
+ testing::NiceMock<MockProcessThread> process_thread_;
+ MockCongestionController congestion_controller_;
+ MockRemoteBitrateEstimator remote_bitrate_estimator_;
+ testing::StrictMock<MockVoiceEngine> voice_engine_;
+ rtc::scoped_refptr<AudioState> audio_state_;
+ AudioReceiveStream::Config stream_config_;
+ testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
+};
+
+void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
+ int id,
+ uint32_t extension_value,
+ size_t value_length) {
+ const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
+ ByteWriter<uint16_t>::WriteBigEndian(&(*it), kRtpOneByteHeaderExtensionId);
+ it += 2;
+
+ ByteWriter<uint16_t>::WriteBigEndian(&(*it), kOneByteExtensionLength / 4);
+ it += 2;
+ const size_t kExtensionDataLength = kOneByteExtensionLength - 1;
+ uint32_t shifted_value = extension_value
+ << (8 * (kExtensionDataLength - value_length));
+ *it = (id << 4) + (value_length - 1);
+ ++it;
+ ByteWriter<uint32_t, kExtensionDataLength>::WriteBigEndian(&(*it),
+ shifted_value);
+}
+
+std::vector<uint8_t> CreateRtpHeaderWithOneByteExtension(
+ int extension_id,
+ uint32_t extension_value,
+ size_t value_length) {
+ std::vector<uint8_t> header;
+ header.resize(webrtc::kRtpHeaderSize + kOneByteExtensionHeaderLength +
+ kOneByteExtensionLength);
header[0] = 0x80; // Version 2.
header[0] |= 0x10; // Set extension bit.
header[1] = 100; // Payload type.
header[1] |= 0x80; // Marker bit is set.
- ByteWriter<uint16_t>::WriteBigEndian(header + 2, 0x1234); // Sequence number.
- ByteWriter<uint32_t>::WriteBigEndian(header + 4, 0x5678); // Timestamp.
- ByteWriter<uint32_t>::WriteBigEndian(header + 8, 0x4321); // SSRC.
- int32_t rtp_header_length = webrtc::kRtpHeaderSize;
+ ByteWriter<uint16_t>::WriteBigEndian(&header[2], 0x1234); // Sequence number.
+ ByteWriter<uint32_t>::WriteBigEndian(&header[4], 0x5678); // Timestamp.
+ ByteWriter<uint32_t>::WriteBigEndian(&header[8], 0x4321); // SSRC.
- BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
- abs_send_time);
- rtp_header_length += kAbsoluteSendTimeLength;
- return rtp_header_length;
+ BuildOneByteExtension(header.begin() + webrtc::kRtpHeaderSize, extension_id,
+ extension_value, value_length);
+ return header;
}
} // namespace
-namespace webrtc {
-namespace test {
-
TEST(AudioReceiveStreamTest, ConfigToString) {
- const int kAbsSendTimeId = 3;
AudioReceiveStream::Config config;
- config.rtp.remote_ssrc = 1234;
- config.rtp.local_ssrc = 5678;
+ config.rtp.remote_ssrc = kRemoteSsrc;
+ config.rtp.local_ssrc = kLocalSsrc;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
- config.voe_channel_id = 1;
+ config.voe_channel_id = kChannelId;
config.combined_audio_video_bwe = true;
- EXPECT_EQ("{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
- "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, "
+ EXPECT_EQ(
+ "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
+ "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}]}, "
"receive_transport: nullptr, rtcp_send_transport: nullptr, "
- "voe_channel_id: 1, combined_audio_video_bwe: true}", config.ToString());
+ "voe_channel_id: 2, combined_audio_video_bwe: true}",
+ config.ToString());
}
TEST(AudioReceiveStreamTest, ConstructDestruct) {
- MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
- AudioReceiveStream::Config config;
- config.voe_channel_id = 1;
- internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
- &voice_engine);
+ ConfigHelper helper;
+ internal::AudioReceiveStream recv_stream(
+ helper.congestion_controller(), helper.config(), helper.audio_state());
+}
+
+MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
+ return arg.extension.hasAbsoluteSendTime ==
+ expected_extension.hasAbsoluteSendTime &&
+ arg.extension.absoluteSendTime ==
+ expected_extension.absoluteSendTime &&
+ arg.extension.hasTransportSequenceNumber ==
+ expected_extension.hasTransportSequenceNumber &&
+ arg.extension.transportSequenceNumber ==
+ expected_extension.transportSequenceNumber;
}
TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
- MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
- AudioReceiveStream::Config config;
- config.combined_audio_video_bwe = true;
- config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
- const int kAbsSendTimeId = 3;
- config.rtp.extensions.push_back(
- RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
- internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
- &voice_engine);
- uint8_t rtp_packet[30];
+ ConfigHelper helper;
+ helper.config().combined_audio_video_bwe = true;
+ helper.SetupMockForBweFeedback(false);
+ internal::AudioReceiveStream recv_stream(
+ helper.congestion_controller(), helper.config(), helper.audio_state());
const int kAbsSendTimeValue = 1234;
- CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
+ std::vector<uint8_t> rtp_packet =
+ CreateRtpHeaderWithOneByteExtension(kAbsSendTimeId, kAbsSendTimeValue, 3);
PacketTime packet_time(5678000, 0);
const size_t kExpectedHeaderLength = 20;
- EXPECT_CALL(remote_bitrate_estimator,
- IncomingPacket(packet_time.timestamp / 1000,
- sizeof(rtp_packet) - kExpectedHeaderLength, testing::_, false))
+ RTPHeaderExtension expected_extension;
+ expected_extension.hasAbsoluteSendTime = true;
+ expected_extension.absoluteSendTime = kAbsSendTimeValue;
+ EXPECT_CALL(*helper.remote_bitrate_estimator(),
+ IncomingPacket(packet_time.timestamp / 1000,
+ rtp_packet.size() - kExpectedHeaderLength,
+ VerifyHeaderExtension(expected_extension), false))
.Times(1);
EXPECT_TRUE(
- recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
+ recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
+}
+
+TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
+ ConfigHelper helper;
+ helper.config().combined_audio_video_bwe = true;
+ helper.config().rtp.transport_cc = true;
+ helper.config().rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
+ helper.SetupMockForBweFeedback(true);
+ internal::AudioReceiveStream recv_stream(
+ helper.congestion_controller(), helper.config(), helper.audio_state());
+ const int kTransportSequenceNumberValue = 1234;
+ std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
+ kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
+ PacketTime packet_time(5678000, 0);
+ const size_t kExpectedHeaderLength = 20;
+ RTPHeaderExtension expected_extension;
+ expected_extension.hasTransportSequenceNumber = true;
+ expected_extension.transportSequenceNumber = kTransportSequenceNumberValue;
+ EXPECT_CALL(*helper.remote_bitrate_estimator(),
+ IncomingPacket(packet_time.timestamp / 1000,
+ rtp_packet.size() - kExpectedHeaderLength,
+ VerifyHeaderExtension(expected_extension), false))
+ .Times(1);
+ EXPECT_TRUE(
+ recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
}
TEST(AudioReceiveStreamTest, GetStats) {
- MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
- AudioReceiveStream::Config config;
- config.rtp.remote_ssrc = FakeVoiceEngine::kRecvSsrc;
- config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
- internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
- &voice_engine);
-
+ ConfigHelper helper;
+ internal::AudioReceiveStream recv_stream(
+ helper.congestion_controller(), helper.config(), helper.audio_state());
+ helper.SetupMockForGetStats();
AudioReceiveStream::Stats stats = recv_stream.GetStats();
- const CallStatistics& call_stats = FakeVoiceEngine::kRecvCallStats;
- const CodecInst& codec_inst = FakeVoiceEngine::kRecvCodecInst;
- const NetworkStatistics& net_stats = FakeVoiceEngine::kRecvNetworkStats;
- const AudioDecodingCallStats& decode_stats =
- FakeVoiceEngine::kRecvAudioDecodingCallStats;
- EXPECT_EQ(FakeVoiceEngine::kRecvSsrc, stats.remote_ssrc);
- EXPECT_EQ(static_cast<int64_t>(call_stats.bytesReceived), stats.bytes_rcvd);
- EXPECT_EQ(static_cast<uint32_t>(call_stats.packetsReceived),
+ EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
+ EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
+ EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
stats.packets_rcvd);
- EXPECT_EQ(call_stats.cumulativeLost, stats.packets_lost);
- EXPECT_EQ(Q8ToFloat(call_stats.fractionLost), stats.fraction_lost);
- EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name);
- EXPECT_EQ(call_stats.extendedMax, stats.ext_seqnum);
- EXPECT_EQ(call_stats.jitterSamples / (codec_inst.plfreq / 1000),
+ EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
+ EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost);
+ EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
+ EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum);
+ EXPECT_EQ(kCallStats.jitterSamples / (kCodecInst.plfreq / 1000),
stats.jitter_ms);
- EXPECT_EQ(net_stats.currentBufferSize, stats.jitter_buffer_ms);
- EXPECT_EQ(net_stats.preferredBufferSize, stats.jitter_buffer_preferred_ms);
- EXPECT_EQ(static_cast<uint32_t>(FakeVoiceEngine::kRecvJitterBufferDelay +
- FakeVoiceEngine::kRecvPlayoutBufferDelay), stats.delay_estimate_ms);
- EXPECT_EQ(static_cast<int32_t>(FakeVoiceEngine::kRecvSpeechOutputLevel),
- stats.audio_level);
- EXPECT_EQ(Q14ToFloat(net_stats.currentExpandRate), stats.expand_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentSpeechExpandRate),
+ EXPECT_EQ(kNetworkStats.currentBufferSize, stats.jitter_buffer_ms);
+ EXPECT_EQ(kNetworkStats.preferredBufferSize,
+ stats.jitter_buffer_preferred_ms);
+ EXPECT_EQ(static_cast<uint32_t>(kJitterBufferDelay + kPlayoutBufferDelay),
+ stats.delay_estimate_ms);
+ EXPECT_EQ(static_cast<int32_t>(kSpeechOutputLevel), stats.audio_level);
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate);
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate),
stats.speech_expand_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentSecondaryDecodedRate),
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDecodedRate),
stats.secondary_decoded_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentAccelerateRate), stats.accelerate_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentPreemptiveRate),
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentAccelerateRate),
+ stats.accelerate_rate);
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate),
stats.preemptive_expand_rate);
- EXPECT_EQ(decode_stats.calls_to_silence_generator,
+ EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator,
stats.decoding_calls_to_silence_generator);
- EXPECT_EQ(decode_stats.calls_to_neteq, stats.decoding_calls_to_neteq);
- EXPECT_EQ(decode_stats.decoded_normal, stats.decoding_normal);
- EXPECT_EQ(decode_stats.decoded_plc, stats.decoding_plc);
- EXPECT_EQ(decode_stats.decoded_cng, stats.decoding_cng);
- EXPECT_EQ(decode_stats.decoded_plc_cng, stats.decoding_plc_cng);
- EXPECT_EQ(call_stats.capture_start_ntp_time_ms_,
+ EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
+ EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
+ EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
+ EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
+ EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
+ EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
stats.capture_start_ntp_time_ms);
}
} // namespace test
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index ccfdca5..35a6552 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -12,13 +12,20 @@
#include <string>
+#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
+#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_codec.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
namespace webrtc {
std::string AudioSendStream::Config::Rtp::ToString() const {
@@ -32,6 +39,7 @@
}
}
ss << ']';
+ ss << ", c_name: " << c_name;
ss << '}';
return ss.str();
}
@@ -48,115 +56,43 @@
}
namespace internal {
-AudioSendStream::AudioSendStream(const webrtc::AudioSendStream::Config& config,
- VoiceEngine* voice_engine)
- : config_(config),
- voice_engine_(voice_engine),
- voe_base_(voice_engine) {
+AudioSendStream::AudioSendStream(
+ const webrtc::AudioSendStream::Config& config,
+ const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+ CongestionController* congestion_controller)
+ : config_(config), audio_state_(audio_state) {
LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
- RTC_DCHECK_NE(config.voe_channel_id, -1);
- RTC_DCHECK(voice_engine_);
+ RTC_DCHECK_NE(config_.voe_channel_id, -1);
+ RTC_DCHECK(audio_state_.get());
+ RTC_DCHECK(congestion_controller);
+
+ VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
+ channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+ channel_proxy_->SetCongestionControlObjects(
+ congestion_controller->pacer(),
+ congestion_controller->GetTransportFeedbackObserver(),
+ congestion_controller->packet_router());
+ channel_proxy_->SetRTCPStatus(true);
+ channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
+ channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
+
+ for (const auto& extension : config.rtp.extensions) {
+ if (extension.name == RtpExtension::kAbsSendTime) {
+ channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
+ } else if (extension.name == RtpExtension::kAudioLevel) {
+ channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
+ } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
+ channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
+ } else {
+ RTC_NOTREACHED() << "Registering unsupported RTP extension.";
+ }
+ }
}
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
-}
-
-webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- webrtc::AudioSendStream::Stats stats;
- stats.local_ssrc = config_.rtp.ssrc;
- ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine_);
- ScopedVoEInterface<VoECodec> codec(voice_engine_);
- ScopedVoEInterface<VoERTP_RTCP> rtp(voice_engine_);
- ScopedVoEInterface<VoEVolumeControl> volume(voice_engine_);
- unsigned int ssrc = 0;
- webrtc::CallStatistics call_stats = {0};
- if (rtp->GetLocalSSRC(config_.voe_channel_id, ssrc) == -1 ||
- rtp->GetRTCPStatistics(config_.voe_channel_id, call_stats) == -1) {
- return stats;
- }
-
- stats.bytes_sent = call_stats.bytesSent;
- stats.packets_sent = call_stats.packetsSent;
-
- webrtc::CodecInst codec_inst = {0};
- if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
- RTC_DCHECK_NE(codec_inst.pltype, -1);
- stats.codec_name = codec_inst.plname;
-
- // Get data from the last remote RTCP report.
- std::vector<webrtc::ReportBlock> blocks;
- if (rtp->GetRemoteRTCPReportBlocks(config_.voe_channel_id, &blocks) != -1) {
- for (const webrtc::ReportBlock& block : blocks) {
- // Lookup report for send ssrc only.
- if (block.source_SSRC == stats.local_ssrc) {
- stats.packets_lost = block.cumulative_num_packets_lost;
- stats.fraction_lost = Q8ToFloat(block.fraction_lost);
- stats.ext_seqnum = block.extended_highest_sequence_number;
- // Convert samples to milliseconds.
- if (codec_inst.plfreq / 1000 > 0) {
- stats.jitter_ms =
- block.interarrival_jitter / (codec_inst.plfreq / 1000);
- }
- break;
- }
- }
- }
- }
-
- // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
- // returns 0 to indicate an error value.
- if (call_stats.rttMs > 0) {
- stats.rtt_ms = call_stats.rttMs;
- }
-
- // Local speech level.
- {
- unsigned int level = 0;
- if (volume->GetSpeechInputLevelFullRange(level) != -1) {
- stats.audio_level = static_cast<int32_t>(level);
- }
- }
-
- // TODO(ajm): Re-enable this metric once we have a reliable implementation.
- stats.aec_quality_min = -1;
-
- bool echo_metrics_on = false;
- if (processing->GetEcMetricsStatus(echo_metrics_on) != -1 &&
- echo_metrics_on) {
- // These can also be negative, but in practice -1 is only used to signal
- // insufficient data, since the resolution is limited to multiples of 4 ms.
- int median = -1;
- int std = -1;
- float dummy = 0.0f;
- if (processing->GetEcDelayMetrics(median, std, dummy) != -1) {
- stats.echo_delay_median_ms = median;
- stats.echo_delay_std_ms = std;
- }
-
- // These can take on valid negative values, so use the lowest possible level
- // as default rather than -1.
- int erl = -100;
- int erle = -100;
- int dummy1 = 0;
- int dummy2 = 0;
- if (processing->GetEchoMetrics(erl, erle, dummy1, dummy2) != -1) {
- stats.echo_return_loss = erl;
- stats.echo_return_loss_enhancement = erle;
- }
- }
-
- // TODO(solenberg): Collect typing noise warnings here too!
- // bool typing_noise_detected = typing_noise_detected_;
-
- return stats;
-}
-
-const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
- RTC_DCHECK(thread_checker_.CalledOnValidThread());
- return config_;
+ channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
}
void AudioSendStream::Start() {
@@ -178,5 +114,108 @@
// RTC_DCHECK(!thread_checker_.CalledOnValidThread());
return false;
}
+
+bool AudioSendStream::SendTelephoneEvent(int payload_type, uint8_t event,
+ uint32_t duration_ms) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type) &&
+ channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
+}
+
+webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ webrtc::AudioSendStream::Stats stats;
+ stats.local_ssrc = config_.rtp.ssrc;
+ ScopedVoEInterface<VoEAudioProcessing> processing(voice_engine());
+ ScopedVoEInterface<VoECodec> codec(voice_engine());
+ ScopedVoEInterface<VoEVolumeControl> volume(voice_engine());
+
+ webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
+ stats.bytes_sent = call_stats.bytesSent;
+ stats.packets_sent = call_stats.packetsSent;
+ // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
+ // returns 0 to indicate an error value.
+ if (call_stats.rttMs > 0) {
+ stats.rtt_ms = call_stats.rttMs;
+ }
+ // TODO(solenberg): [was ajm]: Re-enable this metric once we have a reliable
+ // implementation.
+ stats.aec_quality_min = -1;
+
+ webrtc::CodecInst codec_inst = {0};
+ if (codec->GetSendCodec(config_.voe_channel_id, codec_inst) != -1) {
+ RTC_DCHECK_NE(codec_inst.pltype, -1);
+ stats.codec_name = codec_inst.plname;
+
+ // Get data from the last remote RTCP report.
+ for (const auto& block : channel_proxy_->GetRemoteRTCPReportBlocks()) {
+ // Lookup report for send ssrc only.
+ if (block.source_SSRC == stats.local_ssrc) {
+ stats.packets_lost = block.cumulative_num_packets_lost;
+ stats.fraction_lost = Q8ToFloat(block.fraction_lost);
+ stats.ext_seqnum = block.extended_highest_sequence_number;
+ // Convert samples to milliseconds.
+ if (codec_inst.plfreq / 1000 > 0) {
+ stats.jitter_ms =
+ block.interarrival_jitter / (codec_inst.plfreq / 1000);
+ }
+ break;
+ }
+ }
+ }
+
+ // Local speech level.
+ {
+ unsigned int level = 0;
+ int error = volume->GetSpeechInputLevelFullRange(level);
+ RTC_DCHECK_EQ(0, error);
+ stats.audio_level = static_cast<int32_t>(level);
+ }
+
+ bool echo_metrics_on = false;
+ int error = processing->GetEcMetricsStatus(echo_metrics_on);
+ RTC_DCHECK_EQ(0, error);
+ if (echo_metrics_on) {
+ // These can also be negative, but in practice -1 is only used to signal
+ // insufficient data, since the resolution is limited to multiples of 4 ms.
+ int median = -1;
+ int std = -1;
+ float dummy = 0.0f;
+ error = processing->GetEcDelayMetrics(median, std, dummy);
+ RTC_DCHECK_EQ(0, error);
+ stats.echo_delay_median_ms = median;
+ stats.echo_delay_std_ms = std;
+
+ // These can take on valid negative values, so use the lowest possible level
+ // as default rather than -1.
+ int erl = -100;
+ int erle = -100;
+ int dummy1 = 0;
+ int dummy2 = 0;
+ error = processing->GetEchoMetrics(erl, erle, dummy1, dummy2);
+ RTC_DCHECK_EQ(0, error);
+ stats.echo_return_loss = erl;
+ stats.echo_return_loss_enhancement = erle;
+ }
+
+ internal::AudioState* audio_state =
+ static_cast<internal::AudioState*>(audio_state_.get());
+ stats.typing_noise_detected = audio_state->typing_noise_detected();
+
+ return stats;
+}
+
+const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ return config_;
+}
+
+VoiceEngine* AudioSendStream::voice_engine() const {
+ internal::AudioState* audio_state =
+ static_cast<internal::AudioState*>(audio_state_.get());
+ VoiceEngine* voice_engine = audio_state->voice_engine();
+ RTC_DCHECK(voice_engine);
+ return voice_engine;
+}
} // namespace internal
} // namespace webrtc
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index ae81dfc..8b96350 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -12,20 +12,24 @@
#define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
#include "webrtc/audio_send_stream.h"
-#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/audio_state.h"
#include "webrtc/base/thread_checker.h"
-#include "webrtc/voice_engine/include/voe_base.h"
+#include "webrtc/base/scoped_ptr.h"
namespace webrtc {
-
+class CongestionController;
class VoiceEngine;
-namespace internal {
+namespace voe {
+class ChannelProxy;
+} // namespace voe
+namespace internal {
class AudioSendStream final : public webrtc::AudioSendStream {
public:
AudioSendStream(const webrtc::AudioSendStream::Config& config,
- VoiceEngine* voice_engine);
+ const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+ CongestionController* congestion_controller);
~AudioSendStream() override;
// webrtc::SendStream implementation.
@@ -35,16 +39,19 @@
bool DeliverRtcp(const uint8_t* packet, size_t length) override;
// webrtc::AudioSendStream implementation.
+ bool SendTelephoneEvent(int payload_type, uint8_t event,
+ uint32_t duration_ms) override;
webrtc::AudioSendStream::Stats GetStats() const override;
const webrtc::AudioSendStream::Config& config() const;
private:
+ VoiceEngine* voice_engine() const;
+
rtc::ThreadChecker thread_checker_;
const webrtc::AudioSendStream::Config config_;
- VoiceEngine* voice_engine_;
- // We hold one interface pointer to the VoE to make sure it is kept alive.
- ScopedVoEInterface<VoEBase> voe_base_;
+ rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+ rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
};
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 227ec83..466c157 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -8,69 +8,238 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <string>
+#include <vector>
+
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_send_stream.h"
+#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
-#include "webrtc/test/fake_voice_engine.h"
+#include "webrtc/call/congestion_controller.h"
+#include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/test/mock_voe_channel_proxy.h"
+#include "webrtc/test/mock_voice_engine.h"
+#include "webrtc/video/call_stats.h"
namespace webrtc {
namespace test {
+namespace {
+
+using testing::_;
+using testing::Return;
+
+const int kChannelId = 1;
+const uint32_t kSsrc = 1234;
+const char* kCName = "foo_name";
+const int kAudioLevelId = 2;
+const int kAbsSendTimeId = 3;
+const int kTransportSequenceNumberId = 4;
+const int kEchoDelayMedian = 254;
+const int kEchoDelayStdDev = -3;
+const int kEchoReturnLoss = -65;
+const int kEchoReturnLossEnhancement = 101;
+const unsigned int kSpeechInputLevel = 96;
+const CallStatistics kCallStats = {
+ 1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123};
+const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671};
+const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
+const int kTelephoneEventPayloadType = 123;
+const uint8_t kTelephoneEventCode = 45;
+const uint32_t kTelephoneEventDuration = 6789;
+
+struct ConfigHelper {
+ ConfigHelper()
+ : stream_config_(nullptr),
+ call_stats_(Clock::GetRealTimeClock()),
+ process_thread_(ProcessThread::Create("AudioTestThread")),
+ congestion_controller_(process_thread_.get(),
+ &call_stats_,
+ &bitrate_observer_) {
+ using testing::Invoke;
+ using testing::StrEq;
+
+ EXPECT_CALL(voice_engine_,
+ RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
+ EXPECT_CALL(voice_engine_,
+ DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
+ AudioState::Config config;
+ config.voice_engine = &voice_engine_;
+ audio_state_ = AudioState::Create(config);
+
+ EXPECT_CALL(voice_engine_, ChannelProxyFactory(kChannelId))
+ .WillOnce(Invoke([this](int channel_id) {
+ EXPECT_FALSE(channel_proxy_);
+ channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
+ EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
+ EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
+ EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
+ EXPECT_CALL(*channel_proxy_,
+ SetSendAbsoluteSenderTimeStatus(true, kAbsSendTimeId)).Times(1);
+ EXPECT_CALL(*channel_proxy_,
+ SetSendAudioLevelIndicationStatus(true, kAudioLevelId)).Times(1);
+ EXPECT_CALL(*channel_proxy_, EnableSendTransportSequenceNumber(
+ kTransportSequenceNumberId))
+ .Times(1);
+ EXPECT_CALL(*channel_proxy_,
+ SetCongestionControlObjects(
+ congestion_controller_.pacer(),
+ congestion_controller_.GetTransportFeedbackObserver(),
+ congestion_controller_.packet_router()))
+ .Times(1);
+ EXPECT_CALL(*channel_proxy_,
+ SetCongestionControlObjects(nullptr, nullptr, nullptr))
+ .Times(1);
+ return channel_proxy_;
+ }));
+ stream_config_.voe_channel_id = kChannelId;
+ stream_config_.rtp.ssrc = kSsrc;
+ stream_config_.rtp.c_name = kCName;
+ stream_config_.rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
+ stream_config_.rtp.extensions.push_back(
+ RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
+ stream_config_.rtp.extensions.push_back(RtpExtension(
+ RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
+ }
+
+ AudioSendStream::Config& config() { return stream_config_; }
+ rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
+ CongestionController* congestion_controller() {
+ return &congestion_controller_;
+ }
+
+ void SetupMockForSendTelephoneEvent() {
+ EXPECT_TRUE(channel_proxy_);
+ EXPECT_CALL(*channel_proxy_,
+ SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType))
+ .WillOnce(Return(true));
+ EXPECT_CALL(*channel_proxy_,
+ SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
+ .WillOnce(Return(true));
+ }
+
+ void SetupMockForGetStats() {
+ using testing::DoAll;
+ using testing::SetArgReferee;
+
+ std::vector<ReportBlock> report_blocks;
+ webrtc::ReportBlock block = kReportBlock;
+ report_blocks.push_back(block); // Has wrong SSRC.
+ block.source_SSRC = kSsrc;
+ report_blocks.push_back(block); // Correct block.
+ block.fraction_lost = 0;
+ report_blocks.push_back(block); // Duplicate SSRC, bad fraction_lost.
+
+ EXPECT_TRUE(channel_proxy_);
+ EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
+ .WillRepeatedly(Return(kCallStats));
+ EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
+ .WillRepeatedly(Return(report_blocks));
+
+ EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
+ .WillRepeatedly(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
+ EXPECT_CALL(voice_engine_, GetSpeechInputLevelFullRange(_))
+ .WillRepeatedly(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0)));
+ EXPECT_CALL(voice_engine_, GetEcMetricsStatus(_))
+ .WillRepeatedly(DoAll(SetArgReferee<0>(true), Return(0)));
+ EXPECT_CALL(voice_engine_, GetEchoMetrics(_, _, _, _))
+ .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoReturnLoss),
+ SetArgReferee<1>(kEchoReturnLossEnhancement),
+ Return(0)));
+ EXPECT_CALL(voice_engine_, GetEcDelayMetrics(_, _, _))
+ .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoDelayMedian),
+ SetArgReferee<1>(kEchoDelayStdDev), Return(0)));
+ }
+
+ private:
+ testing::StrictMock<MockVoiceEngine> voice_engine_;
+ rtc::scoped_refptr<AudioState> audio_state_;
+ AudioSendStream::Config stream_config_;
+ testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
+ CallStats call_stats_;
+ testing::NiceMock<MockBitrateObserver> bitrate_observer_;
+ rtc::scoped_ptr<ProcessThread> process_thread_;
+ CongestionController congestion_controller_;
+};
+} // namespace
TEST(AudioSendStreamTest, ConfigToString) {
- const int kAbsSendTimeId = 3;
AudioSendStream::Config config(nullptr);
- config.rtp.ssrc = 1234;
+ config.rtp.ssrc = kSsrc;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
- config.voe_channel_id = 1;
+ config.rtp.c_name = kCName;
+ config.voe_channel_id = kChannelId;
config.cng_payload_type = 42;
config.red_payload_type = 17;
- EXPECT_EQ("{rtp: {ssrc: 1234, extensions: [{name: "
- "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, "
- "voe_channel_id: 1, cng_payload_type: 42, red_payload_type: 17}",
+ EXPECT_EQ(
+ "{rtp: {ssrc: 1234, extensions: [{name: "
+ "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
+ "c_name: foo_name}, voe_channel_id: 1, cng_payload_type: 42, "
+ "red_payload_type: 17}",
config.ToString());
}
TEST(AudioSendStreamTest, ConstructDestruct) {
- FakeVoiceEngine voice_engine;
- AudioSendStream::Config config(nullptr);
- config.voe_channel_id = 1;
- internal::AudioSendStream send_stream(config, &voice_engine);
+ ConfigHelper helper;
+ internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+ helper.congestion_controller());
+}
+
+TEST(AudioSendStreamTest, SendTelephoneEvent) {
+ ConfigHelper helper;
+ internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+ helper.congestion_controller());
+ helper.SetupMockForSendTelephoneEvent();
+ EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
+ kTelephoneEventCode, kTelephoneEventDuration));
}
TEST(AudioSendStreamTest, GetStats) {
- FakeVoiceEngine voice_engine;
- AudioSendStream::Config config(nullptr);
- config.rtp.ssrc = FakeVoiceEngine::kSendSsrc;
- config.voe_channel_id = FakeVoiceEngine::kSendChannelId;
- internal::AudioSendStream send_stream(config, &voice_engine);
-
+ ConfigHelper helper;
+ internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+ helper.congestion_controller());
+ helper.SetupMockForGetStats();
AudioSendStream::Stats stats = send_stream.GetStats();
- const CallStatistics& call_stats = FakeVoiceEngine::kSendCallStats;
- const CodecInst& codec_inst = FakeVoiceEngine::kSendCodecInst;
- const ReportBlock& report_block = FakeVoiceEngine::kSendReportBlock;
- EXPECT_EQ(FakeVoiceEngine::kSendSsrc, stats.local_ssrc);
- EXPECT_EQ(static_cast<int64_t>(call_stats.bytesSent), stats.bytes_sent);
- EXPECT_EQ(call_stats.packetsSent, stats.packets_sent);
- EXPECT_EQ(static_cast<int32_t>(report_block.cumulative_num_packets_lost),
+ EXPECT_EQ(kSsrc, stats.local_ssrc);
+ EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
+ EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
+ EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost),
stats.packets_lost);
- EXPECT_EQ(Q8ToFloat(report_block.fraction_lost), stats.fraction_lost);
- EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name);
- EXPECT_EQ(static_cast<int32_t>(report_block.extended_highest_sequence_number),
+ EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
+ EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
+ EXPECT_EQ(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number),
stats.ext_seqnum);
- EXPECT_EQ(static_cast<int32_t>(report_block.interarrival_jitter /
- (codec_inst.plfreq / 1000)), stats.jitter_ms);
- EXPECT_EQ(call_stats.rttMs, stats.rtt_ms);
- EXPECT_EQ(static_cast<int32_t>(FakeVoiceEngine::kSendSpeechInputLevel),
- stats.audio_level);
+ EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
+ (kCodecInst.plfreq / 1000)),
+ stats.jitter_ms);
+ EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
+ EXPECT_EQ(static_cast<int32_t>(kSpeechInputLevel), stats.audio_level);
EXPECT_EQ(-1, stats.aec_quality_min);
- EXPECT_EQ(FakeVoiceEngine::kSendEchoDelayMedian, stats.echo_delay_median_ms);
- EXPECT_EQ(FakeVoiceEngine::kSendEchoDelayStdDev, stats.echo_delay_std_ms);
- EXPECT_EQ(FakeVoiceEngine::kSendEchoReturnLoss, stats.echo_return_loss);
- EXPECT_EQ(FakeVoiceEngine::kSendEchoReturnLossEnhancement,
- stats.echo_return_loss_enhancement);
+ EXPECT_EQ(kEchoDelayMedian, stats.echo_delay_median_ms);
+ EXPECT_EQ(kEchoDelayStdDev, stats.echo_delay_std_ms);
+ EXPECT_EQ(kEchoReturnLoss, stats.echo_return_loss);
+ EXPECT_EQ(kEchoReturnLossEnhancement, stats.echo_return_loss_enhancement);
EXPECT_FALSE(stats.typing_noise_detected);
}
+
+TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
+ ConfigHelper helper;
+ internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
+ helper.congestion_controller());
+ helper.SetupMockForGetStats();
+ EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
+
+ internal::AudioState* internal_audio_state =
+ static_cast<internal::AudioState*>(helper.audio_state().get());
+ VoiceEngineObserver* voe_observer =
+ static_cast<VoiceEngineObserver*>(internal_audio_state);
+ voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
+ EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
+ voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
+ EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
+}
} // namespace test
} // namespace webrtc
diff --git a/webrtc/audio/audio_sink.h b/webrtc/audio/audio_sink.h
new file mode 100644
index 0000000..999644f
--- /dev/null
+++ b/webrtc/audio/audio_sink.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_AUDIO_SINK_H_
+#define WEBRTC_AUDIO_AUDIO_SINK_H_
+
+#if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
+// Avoid conflict with format_macros.h.
+#define __STDC_FORMAT_MACROS
+#endif
+
+#include <inttypes.h>
+#include <stddef.h>
+
+namespace webrtc {
+
+// Represents a simple push audio sink.
+class AudioSinkInterface {
+ public:
+ virtual ~AudioSinkInterface() {}
+
+ struct Data {
+ Data(int16_t* data,
+ size_t samples_per_channel,
+ int sample_rate,
+ size_t channels,
+ uint32_t timestamp)
+ : data(data),
+ samples_per_channel(samples_per_channel),
+ sample_rate(sample_rate),
+ channels(channels),
+ timestamp(timestamp) {}
+
+ int16_t* data; // The actual 16bit audio data.
+ size_t samples_per_channel; // Number of frames in the buffer.
+ int sample_rate; // Sample rate in Hz.
+ size_t channels; // Number of channels in the audio data.
+ uint32_t timestamp; // The RTP timestamp of the first sample.
+ };
+
+ virtual void OnData(const Data& audio) = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_AUDIO_SINK_H_
diff --git a/webrtc/audio/audio_state.cc b/webrtc/audio/audio_state.cc
new file mode 100644
index 0000000..e63f97a
--- /dev/null
+++ b/webrtc/audio/audio_state.cc
@@ -0,0 +1,79 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/audio/audio_state.h"
+
+#include "webrtc/base/atomicops.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/voice_engine/include/voe_errors.h"
+
+namespace webrtc {
+namespace internal {
+
+AudioState::AudioState(const AudioState::Config& config)
+ : config_(config), voe_base_(config.voice_engine) {
+ process_thread_checker_.DetachFromThread();
+ // Only one AudioState should be created per VoiceEngine.
+ RTC_CHECK(voe_base_->RegisterVoiceEngineObserver(*this) != -1);
+}
+
+AudioState::~AudioState() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ voe_base_->DeRegisterVoiceEngineObserver();
+}
+
+VoiceEngine* AudioState::voice_engine() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ return config_.voice_engine;
+}
+
+bool AudioState::typing_noise_detected() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ rtc::CritScope lock(&crit_sect_);
+ return typing_noise_detected_;
+}
+
+// Reference count; implementation copied from rtc::RefCountedObject.
+int AudioState::AddRef() const {
+ return rtc::AtomicOps::Increment(&ref_count_);
+}
+
+// Reference count; implementation copied from rtc::RefCountedObject.
+int AudioState::Release() const {
+ int count = rtc::AtomicOps::Decrement(&ref_count_);
+ if (!count) {
+ delete this;
+ }
+ return count;
+}
+
+void AudioState::CallbackOnError(int channel_id, int err_code) {
+ RTC_DCHECK(process_thread_checker_.CalledOnValidThread());
+
+ // All call sites in VoE, as of this writing, specify -1 as channel_id.
+ RTC_DCHECK(channel_id == -1);
+ LOG(LS_INFO) << "VoiceEngine error " << err_code << " reported on channel "
+ << channel_id << ".";
+ if (err_code == VE_TYPING_NOISE_WARNING) {
+ rtc::CritScope lock(&crit_sect_);
+ typing_noise_detected_ = true;
+ } else if (err_code == VE_TYPING_NOISE_OFF_WARNING) {
+ rtc::CritScope lock(&crit_sect_);
+ typing_noise_detected_ = false;
+ }
+}
+} // namespace internal
+
+rtc::scoped_refptr<AudioState> AudioState::Create(
+ const AudioState::Config& config) {
+ return rtc::scoped_refptr<AudioState>(new internal::AudioState(config));
+}
+} // namespace webrtc
diff --git a/webrtc/audio/audio_state.h b/webrtc/audio/audio_state.h
new file mode 100644
index 0000000..2cb83e4
--- /dev/null
+++ b/webrtc/audio/audio_state.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_AUDIO_STATE_H_
+#define WEBRTC_AUDIO_AUDIO_STATE_H_
+
+#include "webrtc/audio_state.h"
+#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+
+namespace webrtc {
+namespace internal {
+
+class AudioState final : public webrtc::AudioState,
+ public webrtc::VoiceEngineObserver {
+ public:
+ explicit AudioState(const AudioState::Config& config);
+ ~AudioState() override;
+
+ VoiceEngine* voice_engine();
+ bool typing_noise_detected() const;
+
+ private:
+ // rtc::RefCountInterface implementation.
+ int AddRef() const override;
+ int Release() const override;
+
+ // webrtc::VoiceEngineObserver implementation.
+ void CallbackOnError(int channel_id, int err_code) override;
+
+ rtc::ThreadChecker thread_checker_;
+ rtc::ThreadChecker process_thread_checker_;
+ const webrtc::AudioState::Config config_;
+
+ // We hold one interface pointer to the VoE to make sure it is kept alive.
+ ScopedVoEInterface<VoEBase> voe_base_;
+
+ // The critical section isn't strictly needed in this case, but xSAN bots may
+ // trigger on unprotected cross-thread access.
+ mutable rtc::CriticalSection crit_sect_;
+ bool typing_noise_detected_ GUARDED_BY(crit_sect_) = false;
+
+ // Reference count; implementation copied from rtc::RefCountedObject.
+ mutable volatile int ref_count_ = 0;
+
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
+};
+} // namespace internal
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_AUDIO_STATE_H_
diff --git a/webrtc/audio/audio_state_unittest.cc b/webrtc/audio/audio_state_unittest.cc
new file mode 100644
index 0000000..11fbdb4
--- /dev/null
+++ b/webrtc/audio/audio_state_unittest.cc
@@ -0,0 +1,80 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/audio/audio_state.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/test/mock_voice_engine.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+struct ConfigHelper {
+ ConfigHelper() {
+ EXPECT_CALL(voice_engine_,
+ RegisterVoiceEngineObserver(testing::_)).WillOnce(testing::Return(0));
+ EXPECT_CALL(voice_engine_,
+ DeRegisterVoiceEngineObserver()).WillOnce(testing::Return(0));
+ config_.voice_engine = &voice_engine_;
+ }
+ AudioState::Config& config() { return config_; }
+ MockVoiceEngine& voice_engine() { return voice_engine_; }
+
+ private:
+ testing::StrictMock<MockVoiceEngine> voice_engine_;
+ AudioState::Config config_;
+};
+} // namespace
+
+TEST(AudioStateTest, Create) {
+ ConfigHelper helper;
+ rtc::scoped_refptr<AudioState> audio_state =
+ AudioState::Create(helper.config());
+ EXPECT_TRUE(audio_state.get());
+}
+
+TEST(AudioStateTest, ConstructDestruct) {
+ ConfigHelper helper;
+ rtc::scoped_ptr<internal::AudioState> audio_state(
+ new internal::AudioState(helper.config()));
+}
+
+TEST(AudioStateTest, GetVoiceEngine) {
+ ConfigHelper helper;
+ rtc::scoped_ptr<internal::AudioState> audio_state(
+ new internal::AudioState(helper.config()));
+ EXPECT_EQ(audio_state->voice_engine(), &helper.voice_engine());
+}
+
+TEST(AudioStateTest, TypingNoiseDetected) {
+ ConfigHelper helper;
+ rtc::scoped_ptr<internal::AudioState> audio_state(
+ new internal::AudioState(helper.config()));
+ VoiceEngineObserver* voe_observer =
+ static_cast<VoiceEngineObserver*>(audio_state.get());
+ EXPECT_FALSE(audio_state->typing_noise_detected());
+
+ voe_observer->CallbackOnError(-1, VE_NOT_INITED);
+ EXPECT_FALSE(audio_state->typing_noise_detected());
+
+ voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
+ EXPECT_TRUE(audio_state->typing_noise_detected());
+ voe_observer->CallbackOnError(-1, VE_NOT_INITED);
+ EXPECT_TRUE(audio_state->typing_noise_detected());
+
+ voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
+ EXPECT_FALSE(audio_state->typing_noise_detected());
+ voe_observer->CallbackOnError(-1, VE_NOT_INITED);
+ EXPECT_FALSE(audio_state->typing_noise_detected());
+}
+} // namespace test
+} // namespace webrtc
diff --git a/webrtc/audio/webrtc_audio.gypi b/webrtc/audio/webrtc_audio.gypi
index b9d45db..53b7d16 100644
--- a/webrtc/audio/webrtc_audio.gypi
+++ b/webrtc/audio/webrtc_audio.gypi
@@ -18,6 +18,9 @@
'audio/audio_receive_stream.h',
'audio/audio_send_stream.cc',
'audio/audio_send_stream.h',
+ 'audio/audio_sink.h',
+ 'audio/audio_state.cc',
+ 'audio/audio_state.h',
'audio/conversion.h',
'audio/scoped_voe_interface.h',
],
diff --git a/webrtc/audio_receive_stream.h b/webrtc/audio_receive_stream.h
index 3e5a518..8cab094 100644
--- a/webrtc/audio_receive_stream.h
+++ b/webrtc/audio_receive_stream.h
@@ -15,6 +15,7 @@
#include <string>
#include <vector>
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/config.h"
#include "webrtc/stream.h"
#include "webrtc/transport.h"
@@ -23,6 +24,12 @@
namespace webrtc {
class AudioDecoder;
+class AudioSinkInterface;
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
class AudioReceiveStream : public ReceiveStream {
public:
@@ -66,6 +73,12 @@
// Sender SSRC used for sending RTCP (such as receiver reports).
uint32_t local_ssrc = 0;
+ // Enable feedback for send side bandwidth estimation.
+ // See
+ // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
+ // for details.
+ bool transport_cc = false;
+
// RTP header extensions used for the received stream.
std::vector<RtpExtension> extensions;
} rtp;
@@ -95,6 +108,16 @@
};
virtual Stats GetStats() const = 0;
+
+ // Sets an audio sink that receives unmixed audio from the receive stream.
+ // Ownership of the sink is passed to the stream and can be used by the
+ // caller to do lifetime management (i.e. when the sink's dtor is called).
+ // Only one sink can be set and passing a null sink, clears an existing one.
+ // NOTE: Audio must still somehow be pulled through AudioTransport for audio
+ // to stream through this sink. In practice, this happens if mixed audio
+ // is being pulled+rendered and/or if audio is being pulled for the purposes
+ // of feeding to the AEC.
+ virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
};
} // namespace webrtc
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h
index 89b73e6..d1af9e0 100644
--- a/webrtc/audio_send_stream.h
+++ b/webrtc/audio_send_stream.h
@@ -23,6 +23,11 @@
namespace webrtc {
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
+
class AudioSendStream : public SendStream {
public:
struct Stats {
@@ -59,8 +64,11 @@
// Sender SSRC.
uint32_t ssrc = 0;
- // RTP header extensions used for the received stream.
+ // RTP header extensions used for the sent stream.
std::vector<RtpExtension> extensions;
+
+ // RTCP CNAME, see RFC 3550.
+ std::string c_name;
} rtp;
// Transport for outgoing packets. The transport is expected to exist for
@@ -81,6 +89,9 @@
int red_payload_type = -1; // pt, or -1 to disable REDundant coding.
};
+ // TODO(solenberg): Make payload_type a config property instead.
+ virtual bool SendTelephoneEvent(int payload_type, uint8_t event,
+ uint32_t duration_ms) = 0;
virtual Stats GetStats() const = 0;
};
} // namespace webrtc
diff --git a/webrtc/audio_state.h b/webrtc/audio_state.h
new file mode 100644
index 0000000..fa5784c
--- /dev/null
+++ b/webrtc/audio_state.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_AUDIO_STATE_H_
+#define WEBRTC_AUDIO_STATE_H_
+
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+
+namespace webrtc {
+
+class AudioDeviceModule;
+class VoiceEngine;
+
+// WORK IN PROGRESS
+// This class is under development and is not yet intended for for use outside
+// of WebRtc/Libjingle. Please use the VoiceEngine API instead.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
+
+// AudioState holds the state which must be shared between multiple instances of
+// webrtc::Call for audio processing purposes.
+class AudioState : public rtc::RefCountInterface {
+ public:
+ struct Config {
+ // VoiceEngine used for audio streams and audio/video synchronization.
+ // AudioState will tickle the VoE refcount to keep it alive for as long as
+ // the AudioState itself.
+ VoiceEngine* voice_engine = nullptr;
+
+ // The AudioDeviceModule associated with the Calls.
+ AudioDeviceModule* audio_device_module = nullptr;
+ };
+
+ // TODO(solenberg): Replace scoped_refptr with shared_ptr once we can use it.
+ static rtc::scoped_refptr<AudioState> Create(
+ const AudioState::Config& config);
+
+ virtual ~AudioState() {}
+};
+} // namespace webrtc
+
+#endif // WEBRTC_AUDIO_STATE_H_
diff --git a/webrtc/base/Android.mk b/webrtc/base/Android.mk
index 6a786df..f2aafad 100644
--- a/webrtc/base/Android.mk
+++ b/webrtc/base/Android.mk
@@ -21,6 +21,7 @@
checks.cc \
criticalsection.cc \
event.cc \
+ event_tracer.cc \
logging.cc \
platform_file.cc \
platform_thread.cc \
diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn
index 11a2664..0f7a3f2 100644
--- a/webrtc/base/BUILD.gn
+++ b/webrtc/base/BUILD.gn
@@ -10,6 +10,13 @@
import("//build/config/ui.gni")
import("../build/webrtc.gni")
+# Enable OpenSSL (BoringSSL) for iOS. This is covered in webrtc/supplement.gypi
+# for the GYP build.
+import("//build_overrides/webrtc.gni")
+if (is_ios && !build_with_chromium) {
+ use_openssl = true
+}
+
config("rtc_base_config") {
include_dirs = [
"//third_party/jsoncpp/overrides/include",
@@ -41,6 +48,7 @@
config("ios_config") {
libs = [
+ "AVFoundation.framework",
"CFNetwork.framework",
#"Foundation.framework", # Already included in //build/config:default_libs.
@@ -90,6 +98,7 @@
# The subset of rtc_base approved for use outside of libjingle.
static_library("rtc_base_approved") {
+ deps = []
configs += [ "..:common_config" ]
public_configs = [ "..:common_inherited_config" ]
@@ -107,23 +116,28 @@
"byteorder.h",
"checks.cc",
"checks.h",
+ "constructormagic.h",
"criticalsection.cc",
"criticalsection.h",
+ "deprecation.h",
"event.cc",
"event.h",
"event_tracer.cc",
"event_tracer.h",
"exp_filter.cc",
"exp_filter.h",
- "maybe.h",
"md5.cc",
"md5.h",
"md5digest.cc",
"md5digest.h",
+ "optional.h",
"platform_file.cc",
"platform_file.h",
"platform_thread.cc",
"platform_thread.h",
+ "platform_thread_types.h",
+ "random.cc",
+ "random.h",
"safe_conversions.h",
"safe_conversions_impl.h",
"scoped_ptr.h",
@@ -143,10 +157,16 @@
"trace_event.h",
]
- if (!build_with_chromium) {
+ if (build_with_chromium) {
+ # Dependency on chromium's logging (in //base).
+ deps += [ "//base:base" ]
sources += [
- "basictypes.h",
- "constructormagic.h",
+ "../../webrtc_overrides/webrtc/base/logging.cc",
+ "../../webrtc_overrides/webrtc/base/logging.h",
+ ]
+ include_dirs = [ "../../webrtc_overrides" ]
+ } else {
+ sources += [
"logging.cc",
"logging.h",
]
@@ -157,7 +177,8 @@
cflags = []
cflags_cc = []
libs = []
- deps = [
+ deps = []
+ public_deps = [
":rtc_base_approved",
]
@@ -194,7 +215,6 @@
"autodetectproxy.h",
"base64.cc",
"base64.h",
- "basicdefs.h",
"common.cc",
"common.h",
"crc32.cc",
@@ -312,19 +332,20 @@
if (is_posix) {
sources += [
+ "ifaddrs-android.h",
+ "ifaddrs_converter.cc",
"unixfilesystem.cc",
"unixfilesystem.h",
]
}
if (build_with_chromium) {
- sources += [
- "../../webrtc_overrides/webrtc/base/logging.cc",
- "../../webrtc_overrides/webrtc/base/logging.h",
- ]
-
deps += [ "..:webrtc_common" ]
+ if (is_mac) {
+ sources += [ "macifaddrs_converter.cc" ]
+ }
+
if (is_win) {
sources += [ "../../webrtc_overrides/webrtc/base/win32socketinit.cc" ]
}
@@ -592,5 +613,28 @@
if (is_nacl) {
deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
defines += [ "timezone=_timezone" ]
+ sources -= [ "ifaddrs_converter.cc" ]
+ }
+}
+
+if (is_ios) {
+ source_set("rtc_base_objc") {
+ deps = [
+ ":rtc_base",
+ ]
+ cflags = [ "-fobjc-arc" ]
+ configs += [ "..:common_config" ]
+ public_configs = [ "..:common_inherited_config" ]
+
+ sources = [
+ "objc/NSString+StdString.h",
+ "objc/NSString+StdString.mm",
+ "objc/RTCCameraPreviewView.h",
+ "objc/RTCCameraPreviewView.m",
+ "objc/RTCDispatcher.h",
+ "objc/RTCDispatcher.m",
+ "objc/RTCLogging.h",
+ "objc/RTCLogging.mm",
+ ]
}
}
diff --git a/webrtc/base/OWNERS b/webrtc/base/OWNERS
index 9a527df..2f40090 100644
--- a/webrtc/base/OWNERS
+++ b/webrtc/base/OWNERS
@@ -9,4 +9,10 @@
sergeyu@chromium.org
tommi@webrtc.org
+# These are for the common case of adding or renaming files. If you're doing
+# structural changes, please get a review from a reviewer in this file.
+per-file *.gyp=*
+per-file *.gypi=*
+
per-file BUILD.gn=kjellander@webrtc.org
+
diff --git a/webrtc/base/array_view.h b/webrtc/base/array_view.h
index 019bd8b..a7ca66c 100644
--- a/webrtc/base/array_view.h
+++ b/webrtc/base/array_view.h
@@ -11,18 +11,63 @@
#ifndef WEBRTC_BASE_ARRAY_VIEW_H_
#define WEBRTC_BASE_ARRAY_VIEW_H_
-#include <vector>
-
#include "webrtc/base/checks.h"
namespace rtc {
-// Keeps track of an array (a pointer and a size) that it doesn't own.
-// ArrayView objects are immutable except for assignment, and small enough to
-// be cheaply passed by value.
+// Many functions read from or write to arrays. The obvious way to do this is
+// to use two arguments, a pointer to the first element and an element count:
//
-// Note that ArrayView<T> and ArrayView<const T> are distinct types; this is
-// how you would represent mutable and unmutable views of an array.
+// bool Contains17(const int* arr, size_t size) {
+// for (size_t i = 0; i < size; ++i) {
+// if (arr[i] == 17)
+// return true;
+// }
+// return false;
+// }
+//
+// This is flexible, since it doesn't matter how the array is stored (C array,
+// std::vector, rtc::Buffer, ...), but it's error-prone because the caller has
+// to correctly specify the array length:
+//
+// Contains17(arr, arraysize(arr)); // C array
+// Contains17(&arr[0], arr.size()); // std::vector
+// Contains17(arr, size); // pointer + size
+// ...
+//
+// It's also kind of messy to have two separate arguments for what is
+// conceptually a single thing.
+//
+// Enter rtc::ArrayView<T>. It contains a T pointer (to an array it doesn't
+// own) and a count, and supports the basic things you'd expect, such as
+// indexing and iteration. It allows us to write our function like this:
+//
+// bool Contains17(rtc::ArrayView<const int> arr) {
+// for (auto e : arr) {
+// if (e == 17)
+// return true;
+// }
+// return false;
+// }
+//
+// And even better, because a bunch of things will implicitly convert to
+// ArrayView, we can call it like this:
+//
+// Contains17(arr); // C array
+// Contains17(arr); // std::vector
+// Contains17(rtc::ArrayView<int>(arr, size)); // pointer + size
+// ...
+//
+// One important point is that ArrayView<T> and ArrayView<const T> are
+// different types, which allow and don't allow mutation of the array elements,
+// respectively. The implicit conversions work just like you'd hope, so that
+// e.g. vector<int> will convert to either ArrayView<int> or ArrayView<const
+// int>, but const vector<int> will convert only to ArrayView<const int>.
+// (ArrayView itself can be the source type in such conversions, so
+// ArrayView<int> will convert to ArrayView<const int>.)
+//
+// Note: ArrayView is tiny (just a pointer and a count) and trivially copyable,
+// so it's probably cheaper to pass it by value than by const reference.
template <typename T>
class ArrayView final {
public:
@@ -50,17 +95,12 @@
// std::vector).
template <typename U>
ArrayView(U& u) : ArrayView(u.data(), u.size()) {}
- // TODO(kwiberg): Remove the special case for std::vector (and the include of
- // <vector>); it is handled by the general case in C++11, since std::vector
- // has a data() method there.
- template <typename U>
- ArrayView(std::vector<U>& u)
- : ArrayView(u.empty() ? nullptr : &u[0], u.size()) {}
// Indexing, size, and iteration. These allow mutation even if the ArrayView
// is const, because the ArrayView doesn't own the array. (To prevent
// mutation, use ArrayView<const T>.)
size_t size() const { return size_; }
+ bool empty() const { return size_ == 0; }
T* data() const { return data_; }
T& operator[](size_t idx) const {
RTC_DCHECK_LT(idx, size_);
@@ -72,6 +112,15 @@
const T* cbegin() const { return data_; }
const T* cend() const { return data_ + size_; }
+ // Comparing two ArrayViews compares their (pointer,size) pairs; it does
+ // *not* dereference the pointers.
+ friend bool operator==(const ArrayView& a, const ArrayView& b) {
+ return a.data_ == b.data_ && a.size_ == b.size_;
+ }
+ friend bool operator!=(const ArrayView& a, const ArrayView& b) {
+ return !(a == b);
+ }
+
private:
// Invariant: !data_ iff size_ == 0.
void CheckInvariant() const { RTC_DCHECK_EQ(!data_, size_ == 0); }
diff --git a/webrtc/base/array_view_unittest.cc b/webrtc/base/array_view_unittest.cc
index 0d1bff0..8bb1bcc 100644
--- a/webrtc/base/array_view_unittest.cc
+++ b/webrtc/base/array_view_unittest.cc
@@ -214,4 +214,20 @@
}
}
+TEST(ArrayViewTest, TestEmpty) {
+ EXPECT_TRUE(ArrayView<int>().empty());
+ const int a[] = {1, 2, 3};
+ EXPECT_FALSE(ArrayView<const int>(a).empty());
+}
+
+TEST(ArrayViewTest, TestCompare) {
+ int a[] = {1, 2, 3};
+ int b[] = {1, 2, 3};
+ EXPECT_EQ(ArrayView<int>(a), ArrayView<int>(a));
+ EXPECT_EQ(ArrayView<int>(), ArrayView<int>());
+ EXPECT_NE(ArrayView<int>(a), ArrayView<int>(b));
+ EXPECT_NE(ArrayView<int>(a), ArrayView<int>());
+ EXPECT_NE(ArrayView<int>(a), ArrayView<int>(a, 2));
+}
+
} // namespace rtc
diff --git a/webrtc/base/atomicops.h b/webrtc/base/atomicops.h
index a863566..a286bf0 100644
--- a/webrtc/base/atomicops.h
+++ b/webrtc/base/atomicops.h
@@ -42,6 +42,16 @@
new_value,
old_value);
}
+ // Pointer variants.
+ template <typename T>
+ static T* AcquireLoadPtr(T* volatile* ptr) {
+ return *ptr;
+ }
+ template <typename T>
+ static T* CompareAndSwapPtr(T* volatile* ptr, T* old_value, T* new_value) {
+ return static_cast<T*>(::InterlockedCompareExchangePointer(
+ reinterpret_cast<PVOID volatile*>(ptr), new_value, old_value));
+ }
#else
static int Increment(volatile int* i) {
return __sync_add_and_fetch(i, 1);
@@ -58,6 +68,15 @@
static int CompareAndSwap(volatile int* i, int old_value, int new_value) {
return __sync_val_compare_and_swap(i, old_value, new_value);
}
+ // Pointer variants.
+ template <typename T>
+ static T* AcquireLoadPtr(T* volatile* ptr) {
+ return __atomic_load_n(ptr, __ATOMIC_ACQUIRE);
+ }
+ template <typename T>
+ static T* CompareAndSwapPtr(T* volatile* ptr, T* old_value, T* new_value) {
+ return __sync_val_compare_and_swap(ptr, old_value, new_value);
+ }
#endif
};
diff --git a/webrtc/base/autodetectproxy_unittest.cc b/webrtc/base/autodetectproxy_unittest.cc
index bc57304..2ae7a6a 100644
--- a/webrtc/base/autodetectproxy_unittest.cc
+++ b/webrtc/base/autodetectproxy_unittest.cc
@@ -12,7 +12,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/httpcommon.h"
#include "webrtc/base/httpcommon-inl.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/base.gyp b/webrtc/base/base.gyp
index 4b6ad85..c918263 100644
--- a/webrtc/base/base.gyp
+++ b/webrtc/base/base.gyp
@@ -22,6 +22,46 @@
}],
],
}],
+ # TODO(tkchin): Mac support. There are a bunch of problems right now because
+ # of some settings pulled down from Chromium.
+ ['OS=="ios"', {
+ 'targets': [
+ {
+ 'target_name': 'rtc_base_objc',
+ 'type': 'static_library',
+ 'dependencies': [
+ 'rtc_base',
+ ],
+ 'sources': [
+ 'objc/NSString+StdString.h',
+ 'objc/NSString+StdString.mm',
+ 'objc/RTCDispatcher.h',
+ 'objc/RTCDispatcher.m',
+ 'objc/RTCLogging.h',
+ 'objc/RTCLogging.mm',
+ ],
+ 'conditions': [
+ ['OS=="ios"', {
+ 'sources': [
+ 'objc/RTCCameraPreviewView.h',
+ 'objc/RTCCameraPreviewView.m',
+ ],
+ 'all_dependent_settings': {
+ 'xcode_settings': {
+ 'OTHER_LDFLAGS': [
+ '-framework AVFoundation',
+ ],
+ },
+ },
+ }],
+ ],
+ 'xcode_settings': {
+ 'CLANG_ENABLE_OBJC_ARC': 'YES',
+ 'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'YES',
+ },
+ }
+ ],
+ }], # OS=="ios"
],
'targets': [
{
@@ -31,7 +71,6 @@
'sources': [
'array_view.h',
'atomicops.h',
- 'basictypes.h',
'bitbuffer.cc',
'bitbuffer.h',
'buffer.cc',
@@ -46,6 +85,7 @@
'constructormagic.h',
'criticalsection.cc',
'criticalsection.h',
+ 'deprecation.h',
'event.cc',
'event.h',
'event_tracer.cc',
@@ -54,15 +94,18 @@
'exp_filter.h',
'logging.cc',
'logging.h',
- 'maybe.h',
'md5.cc',
'md5.h',
'md5digest.cc',
'md5digest.h',
+ 'optional.h',
'platform_file.cc',
'platform_file.h',
'platform_thread.cc',
'platform_thread.h',
+ 'platform_thread_types.h',
+ 'random.cc',
+ 'random.h',
'ratetracker.cc',
'ratetracker.h',
'safe_conversions.h',
@@ -85,9 +128,16 @@
],
'conditions': [
['build_with_chromium==1', {
+ 'dependencies': [
+ '<(DEPTH)/base/base.gyp:base',
+ ],
'include_dirs': [
'../../webrtc_overrides',
],
+ 'sources': [
+ '../../webrtc_overrides/webrtc/base/logging.cc',
+ '../../webrtc_overrides/webrtc/base/logging.h',
+ ],
'sources!': [
'logging.cc',
'logging.h',
@@ -102,6 +152,9 @@
'<(webrtc_root)/common.gyp:webrtc_common',
'rtc_base_approved',
],
+ 'export_dependent_settings': [
+ 'rtc_base_approved',
+ ],
'defines': [
'FEATURE_ENABLE_SSL',
'SSL_USE_OPENSSL',
@@ -131,7 +184,6 @@
'bandwidthsmoother.h',
'base64.cc',
'base64.h',
- 'basicdefs.h',
'bind.h',
'callback.h',
'common.cc',
@@ -172,6 +224,9 @@
'httpserver.h',
'ifaddrs-android.cc',
'ifaddrs-android.h',
+ 'ifaddrs_converter.cc',
+ 'ifaddrs_converter.h',
+ 'macifaddrs_converter.cc',
'iosfilesystem.mm',
'ipaddress.cc',
'ipaddress.h',
@@ -380,8 +435,6 @@
'../../boringssl/src/include',
],
'sources': [
- '../../webrtc_overrides/webrtc/base/logging.cc',
- '../../webrtc_overrides/webrtc/base/logging.h',
'../../webrtc_overrides/webrtc/base/win32socketinit.cc',
],
'sources!': [
@@ -390,7 +443,6 @@
'bandwidthsmoother.h',
'bind.h',
'callback.h',
- 'constructormagic.h',
'dbus.cc',
'dbus.h',
'diskcache_win32.cc',
@@ -494,6 +546,17 @@
'WEBRTC_EXTERNAL_JSON',
],
}],
+ ['OS=="win" and clang==1', {
+ 'msvs_settings': {
+ 'VCCLCompilerTool': {
+ 'AdditionalOptions': [
+ # Disable warnings failing when compiling with Clang on Windows.
+ # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+ '-Wno-missing-braces',
+ ],
+ },
+ },
+ }],
],
}],
['OS == "android"', {
@@ -602,6 +665,9 @@
],
}],
['OS=="win"', {
+ 'sources!': [
+ 'ifaddrs_converter.cc',
+ ],
'link_settings': {
'libraries': [
'-lcrypt32.lib',
@@ -653,6 +719,7 @@
}],
['OS!="ios" and OS!="mac"', {
'sources!': [
+ 'macifaddrs_converter.cc',
'scoped_autorelease_pool.mm',
],
}],
diff --git a/webrtc/base/base_tests.gyp b/webrtc/base/base_tests.gyp
index ee371f4..5d73d50 100644
--- a/webrtc/base/base_tests.gyp
+++ b/webrtc/base/base_tests.gyp
@@ -71,22 +71,25 @@
'httpserver_unittest.cc',
'ipaddress_unittest.cc',
'logging_unittest.cc',
- 'maybe_unittest.cc',
'md5digest_unittest.cc',
'messagedigest_unittest.cc',
'messagequeue_unittest.cc',
'multipart_unittest.cc',
'nat_unittest.cc',
'network_unittest.cc',
+ 'optional_unittest.cc',
'optionsfile_unittest.cc',
'pathutils_unittest.cc',
+ 'platform_thread_unittest.cc',
'profiler_unittest.cc',
'proxy_unittest.cc',
'proxydetect_unittest.cc',
+ 'random_unittest.cc',
'ratelimiter_unittest.cc',
'ratetracker_unittest.cc',
'referencecountedsingletonfactory_unittest.cc',
'rollingaccumulator_unittest.cc',
+ 'rtccertificate_unittests.cc',
'scopedptrcollection_unittest.cc',
'sha1digest_unittest.cc',
'sharedexclusivelock_unittest.cc',
@@ -131,6 +134,18 @@
'win32windowpicker_unittest.cc',
],
}],
+ ['OS=="win" and clang==1', {
+ 'msvs_settings': {
+ 'VCCLCompilerTool': {
+ 'AdditionalOptions': [
+ # Disable warnings failing when compiling with Clang on Windows.
+ # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
+ '-Wno-missing-braces',
+ '-Wno-unused-const-variable',
+ ],
+ },
+ },
+ }],
['OS=="mac"', {
'sources': [
'macutils_unittest.cc',
diff --git a/webrtc/base/basicdefs.h b/webrtc/base/basicdefs.h
deleted file mode 100644
index 1dee2ae..0000000
--- a/webrtc/base/basicdefs.h
+++ /dev/null
@@ -1,20 +0,0 @@
-/*
- * Copyright 2004 The WebRTC Project Authors. All rights reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_BASE_BASICDEFS_H_
-#define WEBRTC_BASE_BASICDEFS_H_
-
-#if HAVE_CONFIG_H
-#include "config.h" // NOLINT
-#endif
-
-#define ARRAY_SIZE(x) (static_cast<int>(sizeof(x) / sizeof(x[0])))
-
-#endif // WEBRTC_BASE_BASICDEFS_H_
diff --git a/webrtc/base/bitbuffer_unittest.cc b/webrtc/base/bitbuffer_unittest.cc
index 99701f7..ce42257 100644
--- a/webrtc/base/bitbuffer_unittest.cc
+++ b/webrtc/base/bitbuffer_unittest.cc
@@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/bitbuffer.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/common.h"
@@ -301,11 +302,11 @@
char test_string[] = "my precious";
uint8_t bytes[64] = {0};
BitBufferWriter buffer(bytes, 64);
- for (size_t i = 0; i < ARRAY_SIZE(test_string); ++i) {
+ for (size_t i = 0; i < arraysize(test_string); ++i) {
EXPECT_TRUE(buffer.WriteExponentialGolomb(test_string[i]));
}
buffer.Seek(0, 0);
- for (size_t i = 0; i < ARRAY_SIZE(test_string); ++i) {
+ for (size_t i = 0; i < arraysize(test_string); ++i) {
uint32_t val;
EXPECT_TRUE(buffer.ReadExponentialGolomb(&val));
EXPECT_LE(val, std::numeric_limits<uint8_t>::max());
diff --git a/webrtc/base/buffer.cc b/webrtc/base/buffer.cc
index 90e687b..62855f1 100644
--- a/webrtc/base/buffer.cc
+++ b/webrtc/base/buffer.cc
@@ -11,6 +11,7 @@
#include "webrtc/base/buffer.h"
#include <cassert>
+#include <utility>
namespace rtc {
@@ -22,7 +23,9 @@
}
Buffer::Buffer(Buffer&& buf)
- : size_(buf.size()), capacity_(buf.capacity()), data_(buf.data_.Pass()) {
+ : size_(buf.size()),
+ capacity_(buf.capacity()),
+ data_(std::move(buf.data_)) {
assert(IsConsistent());
buf.OnMovedFrom();
}
diff --git a/webrtc/base/buffer.h b/webrtc/base/buffer.h
index 7b9402b..ff9bb73 100644
--- a/webrtc/base/buffer.h
+++ b/webrtc/base/buffer.h
@@ -15,6 +15,8 @@
#include <cassert>
#include <cstring>
#include <utility> // std::swap (C++11 and later)
+
+#include "webrtc/base/deprecation.h"
#include "webrtc/base/scoped_ptr.h"
namespace rtc {
@@ -104,7 +106,7 @@
assert(buf.IsConsistent());
size_ = buf.size_;
capacity_ = buf.capacity_;
- data_ = buf.data_.Pass();
+ data_ = std::move(buf.data_);
buf.OnMovedFrom();
return *this;
}
@@ -164,15 +166,17 @@
return;
scoped_ptr<uint8_t[]> new_data(new uint8_t[capacity]);
std::memcpy(new_data.get(), data_.get(), size_);
- data_ = new_data.Pass();
+ data_ = std::move(new_data);
capacity_ = capacity;
assert(IsConsistent());
}
- // We can't call std::move(b), so call b.Pass() instead to do the same job.
- Buffer&& Pass() {
+ // b.Pass() does the same thing as std::move(b).
+ // Deprecated; remove in March 2016 (bug 5373).
+ RTC_DEPRECATED Buffer&& Pass() { return DEPRECATED_Pass(); }
+ Buffer&& DEPRECATED_Pass() {
assert(IsConsistent());
- return static_cast<Buffer&&>(*this);
+ return std::move(*this);
}
// Resets the buffer to zero size and capacity. Works even if the buffer has
diff --git a/webrtc/base/buffer_unittest.cc b/webrtc/base/buffer_unittest.cc
index f1ae6b8..0b93b9b 100644
--- a/webrtc/base/buffer_unittest.cc
+++ b/webrtc/base/buffer_unittest.cc
@@ -138,7 +138,7 @@
TEST(BufferTest, TestMoveConstruct) {
Buffer buf1(kTestData, 3, 40);
const uint8_t* data = buf1.data();
- Buffer buf2(buf1.Pass());
+ Buffer buf2(buf1.DEPRECATED_Pass());
EXPECT_EQ(buf2.size(), 3u);
EXPECT_EQ(buf2.capacity(), 40u);
EXPECT_EQ(buf2.data(), data);
@@ -152,7 +152,7 @@
Buffer buf1(kTestData, 3, 40);
const uint8_t* data = buf1.data();
Buffer buf2(kTestData);
- buf2 = buf1.Pass();
+ buf2 = buf1.DEPRECATED_Pass();
EXPECT_EQ(buf2.size(), 3u);
EXPECT_EQ(buf2.capacity(), 40u);
EXPECT_EQ(buf2.data(), data);
diff --git a/webrtc/base/bufferqueue.cc b/webrtc/base/bufferqueue.cc
index 955af51..1ac57ab 100644
--- a/webrtc/base/bufferqueue.cc
+++ b/webrtc/base/bufferqueue.cc
@@ -38,19 +38,19 @@
return false;
}
+ bool was_writable = queue_.size() < capacity_;
Buffer* packet = queue_.front();
queue_.pop_front();
- size_t next_packet_size = packet->size();
- if (bytes > next_packet_size) {
- bytes = next_packet_size;
- }
-
+ bytes = std::min(bytes, packet->size());
memcpy(buffer, packet->data(), bytes);
if (bytes_read) {
*bytes_read = bytes;
}
free_list_.push_back(packet);
+ if (!was_writable) {
+ NotifyWritableForTest();
+ }
return true;
}
@@ -61,6 +61,7 @@
return false;
}
+ bool was_readable = !queue_.empty();
Buffer* packet;
if (!free_list_.empty()) {
packet = free_list_.back();
@@ -74,6 +75,9 @@
*bytes_written = bytes;
}
queue_.push_back(packet);
+ if (!was_readable) {
+ NotifyReadableForTest();
+ }
return true;
}
diff --git a/webrtc/base/bufferqueue.h b/webrtc/base/bufferqueue.h
index 4941fcc..458f018 100644
--- a/webrtc/base/bufferqueue.h
+++ b/webrtc/base/bufferqueue.h
@@ -21,26 +21,33 @@
class BufferQueue {
public:
- // Creates a buffer queue queue with a given capacity and default buffer size.
+ // Creates a buffer queue with a given capacity and default buffer size.
BufferQueue(size_t capacity, size_t default_size);
- ~BufferQueue();
+ virtual ~BufferQueue();
// Return number of queued buffers.
size_t size() const;
// ReadFront will only read one buffer at a time and will truncate buffers
// that don't fit in the passed memory.
+ // Returns true unless no data could be returned.
bool ReadFront(void* data, size_t bytes, size_t* bytes_read);
// WriteBack always writes either the complete memory or nothing.
+ // Returns true unless no data could be written.
bool WriteBack(const void* data, size_t bytes, size_t* bytes_written);
+ protected:
+ // These methods are called when the state of the queue changes.
+ virtual void NotifyReadableForTest() {}
+ virtual void NotifyWritableForTest() {}
+
private:
size_t capacity_;
size_t default_size_;
- std::deque<Buffer*> queue_;
- std::vector<Buffer*> free_list_;
- mutable CriticalSection crit_; // object lock
+ mutable CriticalSection crit_;
+ std::deque<Buffer*> queue_ GUARDED_BY(crit_);
+ std::vector<Buffer*> free_list_ GUARDED_BY(crit_);
RTC_DISALLOW_COPY_AND_ASSIGN(BufferQueue);
};
diff --git a/webrtc/base/bytebuffer_unittest.cc b/webrtc/base/bytebuffer_unittest.cc
index 56b0e05..0287d85 100644
--- a/webrtc/base/bytebuffer_unittest.cc
+++ b/webrtc/base/bytebuffer_unittest.cc
@@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/bytebuffer.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/common.h"
@@ -114,7 +115,7 @@
TEST(ByteBufferTest, TestReadWriteBuffer) {
ByteBuffer::ByteOrder orders[2] = { ByteBuffer::ORDER_HOST,
ByteBuffer::ORDER_NETWORK };
- for (size_t i = 0; i < ARRAY_SIZE(orders); i++) {
+ for (size_t i = 0; i < arraysize(orders); i++) {
ByteBuffer buffer(orders[i]);
EXPECT_EQ(orders[i], buffer.Order());
uint8_t ru8;
diff --git a/webrtc/base/callback_unittest.cc b/webrtc/base/callback_unittest.cc
index 66c9391..db294cd 100644
--- a/webrtc/base/callback_unittest.cc
+++ b/webrtc/base/callback_unittest.cc
@@ -11,6 +11,8 @@
#include "webrtc/base/bind.h"
#include "webrtc/base/callback.h"
#include "webrtc/base/gunit.h"
+#include "webrtc/base/keep_ref_until_done.h"
+#include "webrtc/base/refcount.h"
namespace rtc {
@@ -26,6 +28,21 @@
int b(int x) const { return x * x; }
};
+class RefCountedBindTester : public RefCountInterface {
+ public:
+ RefCountedBindTester() : count_(0) {}
+ int AddRef() const override {
+ return ++count_;
+ }
+ int Release() const {
+ return --count_;
+ }
+ int RefCount() const { return count_; }
+
+ private:
+ mutable int count_;
+};
+
} // namespace
TEST(CallbackTest, VoidReturn) {
@@ -78,4 +95,46 @@
EXPECT_EQ(25, cb1());
}
+TEST(KeepRefUntilDoneTest, simple) {
+ RefCountedBindTester t;
+ EXPECT_EQ(0, t.RefCount());
+ {
+ Callback0<void> cb = KeepRefUntilDone(&t);
+ EXPECT_EQ(1, t.RefCount());
+ cb();
+ EXPECT_EQ(1, t.RefCount());
+ cb();
+ EXPECT_EQ(1, t.RefCount());
+ }
+ EXPECT_EQ(0, t.RefCount());
+}
+
+TEST(KeepRefUntilDoneTest, copy) {
+ RefCountedBindTester t;
+ EXPECT_EQ(0, t.RefCount());
+ Callback0<void> cb2;
+ {
+ Callback0<void> cb = KeepRefUntilDone(&t);
+ EXPECT_EQ(1, t.RefCount());
+ cb2 = cb;
+ }
+ EXPECT_EQ(1, t.RefCount());
+ cb2 = Callback0<void>();
+ EXPECT_EQ(0, t.RefCount());
+}
+
+TEST(KeepRefUntilDoneTest, scopedref) {
+ RefCountedBindTester t;
+ EXPECT_EQ(0, t.RefCount());
+ {
+ scoped_refptr<RefCountedBindTester> t_scoped_ref(&t);
+ Callback0<void> cb = KeepRefUntilDone(t_scoped_ref);
+ t_scoped_ref = nullptr;
+ EXPECT_EQ(1, t.RefCount());
+ cb();
+ EXPECT_EQ(1, t.RefCount());
+ }
+ EXPECT_EQ(0, t.RefCount());
+}
+
} // namespace rtc
diff --git a/webrtc/base/common.h b/webrtc/base/common.h
index e615c76..1b1dac6 100644
--- a/webrtc/base/common.h
+++ b/webrtc/base/common.h
@@ -54,14 +54,16 @@
#endif // !defined(WEBRTC_WIN)
-#define ARRAY_SIZE(x) (static_cast<int>(sizeof(x) / sizeof(x[0])))
-
/////////////////////////////////////////////////////////////////////////////
// Assertions
/////////////////////////////////////////////////////////////////////////////
#ifndef ENABLE_DEBUG
-#define ENABLE_DEBUG _DEBUG
+#if !defined(NDEBUG)
+#define ENABLE_DEBUG 1
+#else
+#define ENABLE_DEBUG 0
+#endif
#endif // !defined(ENABLE_DEBUG)
// Even for release builds, allow for the override of LogAssert. Though no
@@ -176,7 +178,7 @@
// Forces compiler to inline, even against its better judgement. Use wisely.
#if defined(__GNUC__)
-#define FORCE_INLINE __attribute__((always_inline))
+#define FORCE_INLINE __attribute__ ((__always_inline__))
#elif defined(WEBRTC_WIN)
#define FORCE_INLINE __forceinline
#else
@@ -191,7 +193,7 @@
// libjingle are merged.
#if !defined(WARN_UNUSED_RESULT)
#if defined(__GNUC__) || defined(__clang__)
-#define WARN_UNUSED_RESULT __attribute__((warn_unused_result))
+#define WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
#else
#define WARN_UNUSED_RESULT
#endif
diff --git a/webrtc/base/cpumonitor_unittest.cc b/webrtc/base/cpumonitor_unittest.cc
deleted file mode 100644
index 379f62f..0000000
--- a/webrtc/base/cpumonitor_unittest.cc
+++ /dev/null
@@ -1,389 +0,0 @@
-/*
- * Copyright 2010 The WebRTC Project Authors. All rights reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <iomanip>
-#include <iostream>
-#include <vector>
-
-#if defined(WEBRTC_WIN)
-#include "webrtc/base/win32.h"
-#endif
-
-#include "webrtc/base/cpumonitor.h"
-#include "webrtc/base/flags.h"
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread.h"
-#include "webrtc/base/timeutils.h"
-#include "webrtc/base/timing.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
-
-namespace rtc {
-
-static const int kMaxCpus = 1024;
-static const int kSettleTime = 100; // Amount of time to between tests.
-static const int kIdleTime = 500; // Amount of time to be idle in ms.
-static const int kBusyTime = 1000; // Amount of time to be busy in ms.
-static const int kLongInterval = 2000; // Interval longer than busy times
-
-class BusyThread : public rtc::Thread {
- public:
- BusyThread(double load, double duration, double interval) :
- load_(load), duration_(duration), interval_(interval) {
- }
- virtual ~BusyThread() {
- Stop();
- }
- void Run() {
- Timing time;
- double busy_time = interval_ * load_ / 100.0;
- for (;;) {
- time.BusyWait(busy_time);
- time.IdleWait(interval_ - busy_time);
- if (duration_) {
- duration_ -= interval_;
- if (duration_ <= 0) {
- break;
- }
- }
- }
- }
- private:
- double load_;
- double duration_;
- double interval_;
-};
-
-class CpuLoadListener : public sigslot::has_slots<> {
- public:
- CpuLoadListener()
- : current_cpus_(0),
- cpus_(0),
- process_load_(.0f),
- system_load_(.0f),
- count_(0) {
- }
-
- void OnCpuLoad(int current_cpus, int cpus, float proc_load, float sys_load) {
- current_cpus_ = current_cpus;
- cpus_ = cpus;
- process_load_ = proc_load;
- system_load_ = sys_load;
- ++count_;
- }
-
- int current_cpus() const { return current_cpus_; }
- int cpus() const { return cpus_; }
- float process_load() const { return process_load_; }
- float system_load() const { return system_load_; }
- int count() const { return count_; }
-
- private:
- int current_cpus_;
- int cpus_;
- float process_load_;
- float system_load_;
- int count_;
-};
-
-// Set affinity (which cpu to run on), but respecting FLAG_affinity:
-// -1 means no affinity - run on whatever cpu is available.
-// 0 .. N means run on specific cpu. The tool will create N threads and call
-// SetThreadAffinity on 0 to N - 1 as cpu. FLAG_affinity sets the first cpu
-// so the range becomes affinity to affinity + N - 1
-// Note that this function affects Windows scheduling, effectively giving
-// the thread with affinity for a specified CPU more priority on that CPU.
-bool SetThreadAffinity(BusyThread* t, int cpu, int affinity) {
-#if defined(WEBRTC_WIN)
- if (affinity >= 0) {
- return ::SetThreadAffinityMask(t->GetHandle(),
- 1 << (cpu + affinity)) != FALSE;
- }
-#endif
- return true;
-}
-
-bool SetThreadPriority(BusyThread* t, int prio) {
- if (!prio) {
- return true;
- }
- bool ok = t->SetPriority(static_cast<rtc::ThreadPriority>(prio));
- if (!ok) {
- std::cout << "Error setting thread priority." << std::endl;
- }
- return ok;
-}
-
-int CpuLoad(double cpuload, double duration, int numthreads,
- int priority, double interval, int affinity) {
- int ret = 0;
- std::vector<BusyThread*> threads;
- for (int i = 0; i < numthreads; ++i) {
- threads.push_back(new BusyThread(cpuload, duration, interval));
- // NOTE(fbarchard): Priority must be done before Start.
- if (!SetThreadPriority(threads[i], priority) ||
- !threads[i]->Start() ||
- !SetThreadAffinity(threads[i], i, affinity)) {
- ret = 1;
- break;
- }
- }
- // Wait on each thread
- if (ret == 0) {
- for (int i = 0; i < numthreads; ++i) {
- threads[i]->Stop();
- }
- }
-
- for (int i = 0; i < numthreads; ++i) {
- delete threads[i];
- }
- return ret;
-}
-
-// Make 2 CPUs busy
-static void CpuTwoBusyLoop(int busytime) {
- CpuLoad(100.0, busytime / 1000.0, 2, 1, 0.050, -1);
-}
-
-// Make 1 CPUs busy
-static void CpuBusyLoop(int busytime) {
- CpuLoad(100.0, busytime / 1000.0, 1, 1, 0.050, -1);
-}
-
-// Make 1 use half CPU time.
-static void CpuHalfBusyLoop(int busytime) {
- CpuLoad(50.0, busytime / 1000.0, 1, 1, 0.050, -1);
-}
-
-void TestCpuSampler(bool test_proc, bool test_sys, bool force_fallback) {
- CpuSampler sampler;
- sampler.set_force_fallback(force_fallback);
- EXPECT_TRUE(sampler.Init());
- sampler.set_load_interval(100);
- int cpus = sampler.GetMaxCpus();
-
- // Test1: CpuSampler under idle situation.
- Thread::SleepMs(kSettleTime);
- sampler.GetProcessLoad();
- sampler.GetSystemLoad();
-
- Thread::SleepMs(kIdleTime);
-
- float proc_idle = 0.f, sys_idle = 0.f;
- if (test_proc) {
- proc_idle = sampler.GetProcessLoad();
- }
- if (test_sys) {
- sys_idle = sampler.GetSystemLoad();
- }
- if (test_proc) {
- LOG(LS_INFO) << "ProcessLoad Idle: "
- << std::setiosflags(std::ios_base::fixed)
- << std::setprecision(2) << std::setw(6) << proc_idle;
- EXPECT_GE(proc_idle, 0.f);
- EXPECT_LE(proc_idle, static_cast<float>(cpus));
- }
- if (test_sys) {
- LOG(LS_INFO) << "SystemLoad Idle: "
- << std::setiosflags(std::ios_base::fixed)
- << std::setprecision(2) << std::setw(6) << sys_idle;
- EXPECT_GE(sys_idle, 0.f);
- EXPECT_LE(sys_idle, static_cast<float>(cpus));
- }
-
- // Test2: CpuSampler with main process at 50% busy.
- Thread::SleepMs(kSettleTime);
- sampler.GetProcessLoad();
- sampler.GetSystemLoad();
-
- CpuHalfBusyLoop(kBusyTime);
-
- float proc_halfbusy = 0.f, sys_halfbusy = 0.f;
- if (test_proc) {
- proc_halfbusy = sampler.GetProcessLoad();
- }
- if (test_sys) {
- sys_halfbusy = sampler.GetSystemLoad();
- }
- if (test_proc) {
- LOG(LS_INFO) << "ProcessLoad Halfbusy: "
- << std::setiosflags(std::ios_base::fixed)
- << std::setprecision(2) << std::setw(6) << proc_halfbusy;
- EXPECT_GE(proc_halfbusy, 0.f);
- EXPECT_LE(proc_halfbusy, static_cast<float>(cpus));
- }
- if (test_sys) {
- LOG(LS_INFO) << "SystemLoad Halfbusy: "
- << std::setiosflags(std::ios_base::fixed)
- << std::setprecision(2) << std::setw(6) << sys_halfbusy;
- EXPECT_GE(sys_halfbusy, 0.f);
- EXPECT_LE(sys_halfbusy, static_cast<float>(cpus));
- }
-
- // Test3: CpuSampler with main process busy.
- Thread::SleepMs(kSettleTime);
- sampler.GetProcessLoad();
- sampler.GetSystemLoad();
-
- CpuBusyLoop(kBusyTime);
-
- float proc_busy = 0.f, sys_busy = 0.f;
- if (test_proc) {
- proc_busy = sampler.GetProcessLoad();
- }
- if (test_sys) {
- sys_busy = sampler.GetSystemLoad();
- }
- if (test_proc) {
- LOG(LS_INFO) << "ProcessLoad Busy: "
- << std::setiosflags(std::ios_base::fixed)
- << std::setprecision(2) << std::setw(6) << proc_busy;
- EXPECT_GE(proc_busy, 0.f);
- EXPECT_LE(proc_busy, static_cast<float>(cpus));
- }
- if (test_sys) {
- LOG(LS_INFO) << "SystemLoad Busy: "
- << std::setiosflags(std::ios_base::fixed)
- << std::setprecision(2) << std::setw(6) << sys_busy;
- EXPECT_GE(sys_busy, 0.f);
- EXPECT_LE(sys_busy, static_cast<float>(cpus));
- }
-
- // Test4: CpuSampler with 2 cpus process busy.
- if (cpus >= 2) {
- Thread::SleepMs(kSettleTime);
- sampler.GetProcessLoad();
- sampler.GetSystemLoad();
-
- CpuTwoBusyLoop(kBusyTime);
-
- float proc_twobusy = 0.f, sys_twobusy = 0.f;
- if (test_proc) {
- proc_twobusy = sampler.GetProcessLoad();
- }
- if (test_sys) {
- sys_twobusy = sampler.GetSystemLoad();
- }
- if (test_proc) {
- LOG(LS_INFO) << "ProcessLoad 2 CPU Busy:"
- << std::setiosflags(std::ios_base::fixed)
- << std::setprecision(2) << std::setw(6) << proc_twobusy;
- EXPECT_GE(proc_twobusy, 0.f);
- EXPECT_LE(proc_twobusy, static_cast<float>(cpus));
- }
- if (test_sys) {
- LOG(LS_INFO) << "SystemLoad 2 CPU Busy: "
- << std::setiosflags(std::ios_base::fixed)
- << std::setprecision(2) << std::setw(6) << sys_twobusy;
- EXPECT_GE(sys_twobusy, 0.f);
- EXPECT_LE(sys_twobusy, static_cast<float>(cpus));
- }
- }
-
- // Test5: CpuSampler with idle process after being busy.
- Thread::SleepMs(kSettleTime);
- sampler.GetProcessLoad();
- sampler.GetSystemLoad();
-
- Thread::SleepMs(kIdleTime);
-
- if (test_proc) {
- proc_idle = sampler.GetProcessLoad();
- }
- if (test_sys) {
- sys_idle = sampler.GetSystemLoad();
- }
- if (test_proc) {
- LOG(LS_INFO) << "ProcessLoad Idle: "
- << std::setiosflags(std::ios_base::fixed)
- << std::setprecision(2) << std::setw(6) << proc_idle;
- EXPECT_GE(proc_idle, 0.f);
- EXPECT_LE(proc_idle, proc_busy);
- }
- if (test_sys) {
- LOG(LS_INFO) << "SystemLoad Idle: "
- << std::setiosflags(std::ios_base::fixed)
- << std::setprecision(2) << std::setw(6) << sys_idle;
- EXPECT_GE(sys_idle, 0.f);
- EXPECT_LE(sys_idle, static_cast<float>(cpus));
- }
-}
-
-TEST(CpuMonitorTest, TestCpus) {
- CpuSampler sampler;
- EXPECT_TRUE(sampler.Init());
- int current_cpus = sampler.GetCurrentCpus();
- int cpus = sampler.GetMaxCpus();
- LOG(LS_INFO) << "Current Cpus: " << std::setw(9) << current_cpus;
- LOG(LS_INFO) << "Maximum Cpus: " << std::setw(9) << cpus;
- EXPECT_GT(cpus, 0);
- EXPECT_LE(cpus, kMaxCpus);
- EXPECT_GT(current_cpus, 0);
- EXPECT_LE(current_cpus, cpus);
-}
-
-#if defined(WEBRTC_WIN)
-// Tests overall system CpuSampler using legacy OS fallback code if applicable.
-TEST(CpuMonitorTest, TestGetSystemLoadForceFallback) {
- TestCpuSampler(false, true, true);
-}
-#endif
-
-// Tests both process and system functions in use at same time.
-TEST(CpuMonitorTest, TestGetBothLoad) {
- TestCpuSampler(true, true, false);
-}
-
-// Tests a query less than the interval produces the same value.
-TEST(CpuMonitorTest, TestInterval) {
- CpuSampler sampler;
- EXPECT_TRUE(sampler.Init());
-
- // Test1: Set interval to large value so sampler will not update.
- sampler.set_load_interval(kLongInterval);
-
- sampler.GetProcessLoad();
- sampler.GetSystemLoad();
-
- float proc_orig = sampler.GetProcessLoad();
- float sys_orig = sampler.GetSystemLoad();
-
- Thread::SleepMs(kIdleTime);
-
- float proc_halftime = sampler.GetProcessLoad();
- float sys_halftime = sampler.GetSystemLoad();
-
- EXPECT_EQ(proc_orig, proc_halftime);
- EXPECT_EQ(sys_orig, sys_halftime);
-}
-
-TEST(CpuMonitorTest, TestCpuMonitor) {
- CpuMonitor monitor(Thread::Current());
- CpuLoadListener listener;
- monitor.SignalUpdate.connect(&listener, &CpuLoadListener::OnCpuLoad);
- EXPECT_TRUE(monitor.Start(10));
- // We have checked cpu load more than twice.
- EXPECT_TRUE_WAIT(listener.count() > 2, 1000);
- EXPECT_GT(listener.current_cpus(), 0);
- EXPECT_GT(listener.cpus(), 0);
- EXPECT_GE(listener.process_load(), .0f);
- EXPECT_GE(listener.system_load(), .0f);
-
- monitor.Stop();
- // Wait 20 ms to ake sure all signals are delivered.
- Thread::Current()->ProcessMessages(20);
- int old_count = listener.count();
- Thread::Current()->ProcessMessages(20);
- // Verfy no more siganls.
- EXPECT_EQ(old_count, listener.count());
-}
-
-} // namespace rtc
diff --git a/webrtc/base/crc32.cc b/webrtc/base/crc32.cc
index eae338a..97b8214 100644
--- a/webrtc/base/crc32.cc
+++ b/webrtc/base/crc32.cc
@@ -10,7 +10,7 @@
#include "webrtc/base/crc32.h"
-#include "webrtc/base/basicdefs.h"
+#include "webrtc/base/arraysize.h"
namespace rtc {
@@ -22,9 +22,9 @@
static uint32_t kCrc32Table[256] = {0};
static void EnsureCrc32TableInited() {
- if (kCrc32Table[ARRAY_SIZE(kCrc32Table) - 1])
+ if (kCrc32Table[arraysize(kCrc32Table) - 1])
return; // already inited
- for (uint32_t i = 0; i < ARRAY_SIZE(kCrc32Table); ++i) {
+ for (uint32_t i = 0; i < arraysize(kCrc32Table); ++i) {
uint32_t c = i;
for (size_t j = 0; j < 8; ++j) {
if (c & 1) {
diff --git a/webrtc/base/criticalsection.h b/webrtc/base/criticalsection.h
index ddbf857..5b3eaf5 100644
--- a/webrtc/base/criticalsection.h
+++ b/webrtc/base/criticalsection.h
@@ -89,7 +89,7 @@
#if defined(WEBRTC_WIN)
_Check_return_ bool locked() const;
#else
- bool locked() const __attribute__((warn_unused_result));
+ bool locked() const __attribute__ ((__warn_unused_result__));
#endif
private:
CriticalSection* const cs_;
diff --git a/webrtc/base/criticalsection_unittest.cc b/webrtc/base/criticalsection_unittest.cc
index ff4fdef..d6990c0 100644
--- a/webrtc/base/criticalsection_unittest.cc
+++ b/webrtc/base/criticalsection_unittest.cc
@@ -14,9 +14,9 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/event.h"
#include "webrtc/base/gunit.h"
+#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scopedptrcollection.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
@@ -220,6 +220,28 @@
EXPECT_EQ(0, value);
}
+TEST(AtomicOpsTest, SimplePtr) {
+ class Foo {};
+ Foo* volatile foo = nullptr;
+ scoped_ptr<Foo> a(new Foo());
+ scoped_ptr<Foo> b(new Foo());
+ // Reading the initial value should work as expected.
+ EXPECT_TRUE(rtc::AtomicOps::AcquireLoadPtr(&foo) == nullptr);
+ // Setting using compare and swap should work.
+ EXPECT_TRUE(rtc::AtomicOps::CompareAndSwapPtr(
+ &foo, static_cast<Foo*>(nullptr), a.get()) == nullptr);
+ EXPECT_TRUE(rtc::AtomicOps::AcquireLoadPtr(&foo) == a.get());
+ // Setting another value but with the wrong previous pointer should fail
+ // (remain a).
+ EXPECT_TRUE(rtc::AtomicOps::CompareAndSwapPtr(
+ &foo, static_cast<Foo*>(nullptr), b.get()) == a.get());
+ EXPECT_TRUE(rtc::AtomicOps::AcquireLoadPtr(&foo) == a.get());
+ // Replacing a with b should work.
+ EXPECT_TRUE(rtc::AtomicOps::CompareAndSwapPtr(&foo, a.get(), b.get()) ==
+ a.get());
+ EXPECT_TRUE(rtc::AtomicOps::AcquireLoadPtr(&foo) == b.get());
+}
+
TEST(AtomicOpsTest, Increment) {
// Create and start lots of threads.
AtomicOpRunner<IncrementOp, UniqueValueVerifier> runner(0);
diff --git a/webrtc/base/deprecation.h b/webrtc/base/deprecation.h
new file mode 100644
index 0000000..ce950f9
--- /dev/null
+++ b/webrtc/base/deprecation.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_DEPRECATION_H_
+#define WEBRTC_BASE_DEPRECATION_H_
+
+// Annotate the declarations of deprecated functions with this to cause a
+// compiler warning when they're used. Like so:
+//
+// RTC_DEPRECATED std::pony PonyPlz(const std::pony_spec& ps);
+//
+// NOTE 1: The annotation goes on the declaration in the .h file, not the
+// definition in the .cc file!
+//
+// NOTE 2: In order to keep unit testing the deprecated function without
+// getting warnings, do something like this:
+//
+// std::pony DEPRECATED_PonyPlz(const std::pony_spec& ps);
+// RTC_DEPRECATED inline std::pony PonyPlz(const std::pony_spec& ps) {
+// return DEPRECATED_PonyPlz(ps);
+// }
+//
+// In other words, rename the existing function, and provide an inline wrapper
+// using the original name that calls it. That way, callers who are willing to
+// call it using the DEPRECATED_-prefixed name don't get the warning.
+//
+// TODO(kwiberg): Remove this when we can use [[deprecated]] from C++14.
+#if defined(_MSC_VER)
+// Note: Deprecation warnings seem to fail to trigger on Windows
+// (https://bugs.chromium.org/p/webrtc/issues/detail?id=5368).
+#define RTC_DEPRECATED __declspec(deprecated)
+#elif defined(__GNUC__)
+#define RTC_DEPRECATED __attribute__ ((__deprecated__))
+#else
+#define RTC_DEPRECATED
+#endif
+
+#endif // WEBRTC_BASE_DEPRECATION_H_
diff --git a/webrtc/base/diskcache.cc b/webrtc/base/diskcache.cc
index 6bbc53e..a1fba6a 100644
--- a/webrtc/base/diskcache.cc
+++ b/webrtc/base/diskcache.cc
@@ -15,6 +15,7 @@
#endif
#include <algorithm>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/common.h"
#include "webrtc/base/diskcache.h"
#include "webrtc/base/fileutils.h"
@@ -23,11 +24,11 @@
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
-#ifdef _DEBUG
+#if !defined(NDEBUG)
#define TRANSPARENT_CACHE_NAMES 1
-#else // !_DEBUG
+#else
#define TRANSPARENT_CACHE_NAMES 0
-#endif // !_DEBUG
+#endif
namespace rtc {
@@ -211,14 +212,14 @@
}
bool DiskCache::CheckLimit() {
-#ifdef _DEBUG
+#if !defined(NDEBUG)
// Temporary check to make sure everything is working correctly.
size_t cache_size = 0;
for (EntryMap::iterator it = map_.begin(); it != map_.end(); ++it) {
cache_size += it->second.size;
}
ASSERT(cache_size == total_size_);
-#endif // _DEBUG
+#endif
// TODO: Replace this with a non-brain-dead algorithm for clearing out the
// oldest resources... something that isn't O(n^2)
@@ -263,7 +264,7 @@
#endif // !TRANSPARENT_CACHE_NAMES
char extension[32];
- sprintfn(extension, ARRAY_SIZE(extension), ".%u", index);
+ sprintfn(extension, arraysize(extension), ".%u", index);
Pathname pathname;
pathname.SetFolder(folder_);
diff --git a/webrtc/base/event_tracer.cc b/webrtc/base/event_tracer.cc
index 5c6d39f..4174589 100644
--- a/webrtc/base/event_tracer.cc
+++ b/webrtc/base/event_tracer.cc
@@ -7,15 +7,26 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-
#include "webrtc/base/event_tracer.h"
+#include <inttypes.h>
+
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/base/trace_event.h"
+
namespace webrtc {
namespace {
-GetCategoryEnabledPtr g_get_category_enabled_ptr = 0;
-AddTraceEventPtr g_add_trace_event_ptr = 0;
+GetCategoryEnabledPtr g_get_category_enabled_ptr = nullptr;
+AddTraceEventPtr g_add_trace_event_ptr = nullptr;
} // namespace
@@ -25,7 +36,6 @@
g_add_trace_event_ptr = add_trace_event_ptr;
}
-// static
const unsigned char* EventTracer::GetCategoryEnabled(const char* name) {
if (g_get_category_enabled_ptr)
return g_get_category_enabled_ptr(name);
@@ -34,7 +44,8 @@
return reinterpret_cast<const unsigned char*>("\0");
}
-// static
+// Arguments to this function (phase, etc.) are as defined in
+// webrtc/base/trace_event.h.
void EventTracer::AddTraceEvent(char phase,
const unsigned char* category_enabled,
const char* name,
@@ -58,3 +69,202 @@
}
} // namespace webrtc
+
+namespace rtc {
+namespace tracing {
+namespace {
+
+static bool EventTracingThreadFunc(void* params);
+
+// Atomic-int fast path for avoiding logging when disabled.
+static volatile int g_event_logging_active = 0;
+
+// TODO(pbos): Log metadata for all threads, etc.
+class EventLogger final {
+ public:
+ EventLogger()
+ : logging_thread_(EventTracingThreadFunc, this, "EventTracingThread"),
+ shutdown_event_(false, false) {}
+ ~EventLogger() { RTC_DCHECK(thread_checker_.CalledOnValidThread()); }
+
+ void AddTraceEvent(const char* name,
+ const unsigned char* category_enabled,
+ char phase,
+ uint64_t timestamp,
+ int pid,
+ rtc::PlatformThreadId thread_id) {
+ rtc::CritScope lock(&crit_);
+ trace_events_.push_back(
+ {name, category_enabled, phase, timestamp, 1, thread_id});
+ }
+
+// The TraceEvent format is documented here:
+// https://docs.google.com/document/d/1CvAClvFfyA5R-PhYUmn5OOQtYMH4h6I0nSsKchNAySU/preview
+ void Log() {
+ RTC_DCHECK(output_file_);
+ static const int kLoggingIntervalMs = 100;
+ fprintf(output_file_, "{ \"traceEvents\": [\n");
+ bool has_logged_event = false;
+ while (true) {
+ bool shutting_down = shutdown_event_.Wait(kLoggingIntervalMs);
+ std::vector<TraceEvent> events;
+ {
+ rtc::CritScope lock(&crit_);
+ trace_events_.swap(events);
+ }
+ for (const TraceEvent& e : events) {
+ fprintf(output_file_,
+ "%s{ \"name\": \"%s\""
+ ", \"cat\": \"%s\""
+ ", \"ph\": \"%c\""
+ ", \"ts\": %" PRIu64
+ ", \"pid\": %d"
+#if defined(WEBRTC_WIN)
+ ", \"tid\": %lu"
+#else
+ ", \"tid\": %d"
+#endif // defined(WEBRTC_WIN)
+ "}\n",
+ has_logged_event ? "," : " ", e.name, e.category_enabled,
+ e.phase, e.timestamp, e.pid, e.tid);
+ has_logged_event = true;
+ }
+ if (shutting_down)
+ break;
+ }
+ fprintf(output_file_, "]}\n");
+ if (output_file_owned_)
+ fclose(output_file_);
+ output_file_ = nullptr;
+ }
+
+ void Start(FILE* file, bool owned) {
+ RTC_DCHECK(file);
+ RTC_DCHECK(!output_file_);
+ output_file_ = file;
+ output_file_owned_ = owned;
+ {
+ rtc::CritScope lock(&crit_);
+ // Since the atomic fast-path for adding events to the queue can be
+ // bypassed while the logging thread is shutting down there may be some
+ // stale events in the queue, hence the vector needs to be cleared to not
+ // log events from a previous logging session (which may be days old).
+ trace_events_.clear();
+ }
+ // Enable event logging (fast-path). This should be disabled since starting
+ // shouldn't be done twice.
+ RTC_CHECK_EQ(0,
+ rtc::AtomicOps::CompareAndSwap(&g_event_logging_active, 0, 1));
+
+ // Finally start, everything should be set up now.
+ logging_thread_.Start();
+ }
+
+ void Stop() {
+ // Try to stop. Abort if we're not currently logging.
+ if (rtc::AtomicOps::CompareAndSwap(&g_event_logging_active, 1, 0) == 0)
+ return;
+
+ // Wake up logging thread to finish writing.
+ shutdown_event_.Set();
+ // Join the logging thread.
+ logging_thread_.Stop();
+ }
+
+ private:
+ struct TraceEvent {
+ const char* name;
+ const unsigned char* category_enabled;
+ char phase;
+ uint64_t timestamp;
+ int pid;
+ rtc::PlatformThreadId tid;
+ };
+
+ rtc::CriticalSection crit_;
+ std::vector<TraceEvent> trace_events_ GUARDED_BY(crit_);
+ rtc::PlatformThread logging_thread_;
+ rtc::Event shutdown_event_;
+ rtc::ThreadChecker thread_checker_;
+ FILE* output_file_ = nullptr;
+ bool output_file_owned_ = false;
+};
+
+static bool EventTracingThreadFunc(void* params) {
+ static_cast<EventLogger*>(params)->Log();
+ return true;
+}
+
+static EventLogger* volatile g_event_logger = nullptr;
+static const char* const kDisabledTracePrefix = TRACE_DISABLED_BY_DEFAULT("");
+const unsigned char* InternalGetCategoryEnabled(const char* name) {
+ const char* prefix_ptr = &kDisabledTracePrefix[0];
+ const char* name_ptr = name;
+ // Check whether name contains the default-disabled prefix.
+ while (*prefix_ptr == *name_ptr && *prefix_ptr != '\0') {
+ ++prefix_ptr;
+ ++name_ptr;
+ }
+ return reinterpret_cast<const unsigned char*>(*prefix_ptr == '\0' ? ""
+ : name);
+}
+
+void InternalAddTraceEvent(char phase,
+ const unsigned char* category_enabled,
+ const char* name,
+ unsigned long long id,
+ int num_args,
+ const char** arg_names,
+ const unsigned char* arg_types,
+ const unsigned long long* arg_values,
+ unsigned char flags) {
+ // Fast path for when event tracing is inactive.
+ if (rtc::AtomicOps::AcquireLoad(&g_event_logging_active) == 0)
+ return;
+
+ g_event_logger->AddTraceEvent(name, category_enabled, phase,
+ rtc::TimeMicros(), 1, rtc::CurrentThreadId());
+}
+
+} // namespace
+
+void SetupInternalTracer() {
+ RTC_CHECK(rtc::AtomicOps::CompareAndSwapPtr(
+ &g_event_logger, static_cast<EventLogger*>(nullptr),
+ new EventLogger()) == nullptr);
+ g_event_logger = new EventLogger();
+ webrtc::SetupEventTracer(InternalGetCategoryEnabled, InternalAddTraceEvent);
+}
+
+void StartInternalCaptureToFile(FILE* file) {
+ g_event_logger->Start(file, false);
+}
+
+bool StartInternalCapture(const char* filename) {
+ FILE* file = fopen(filename, "w");
+ if (!file) {
+ LOG(LS_ERROR) << "Failed to open trace file '" << filename
+ << "' for writing.";
+ return false;
+ }
+ g_event_logger->Start(file, true);
+ return true;
+}
+
+void StopInternalCapture() {
+ g_event_logger->Stop();
+}
+
+void ShutdownInternalTracer() {
+ StopInternalCapture();
+ EventLogger* old_logger = rtc::AtomicOps::AcquireLoadPtr(&g_event_logger);
+ RTC_DCHECK(old_logger);
+ RTC_CHECK(rtc::AtomicOps::CompareAndSwapPtr(
+ &g_event_logger, old_logger,
+ static_cast<EventLogger*>(nullptr)) == old_logger);
+ delete old_logger;
+ webrtc::SetupEventTracer(nullptr, nullptr);
+}
+
+} // namespace tracing
+} // namespace rtc
diff --git a/webrtc/base/event_tracer.h b/webrtc/base/event_tracer.h
index cfc6e9e..51c8cfd 100644
--- a/webrtc/base/event_tracer.h
+++ b/webrtc/base/event_tracer.h
@@ -26,6 +26,8 @@
#ifndef WEBRTC_BASE_EVENT_TRACER_H_
#define WEBRTC_BASE_EVENT_TRACER_H_
+#include <stdio.h>
+
namespace webrtc {
typedef const unsigned char* (*GetCategoryEnabledPtr)(const char* name);
@@ -68,4 +70,16 @@
} // namespace webrtc
+namespace rtc {
+namespace tracing {
+// Set up internal event tracer.
+void SetupInternalTracer();
+bool StartInternalCapture(const char* filename);
+void StartInternalCaptureToFile(FILE* file);
+void StopInternalCapture();
+// Make sure we run this, this will tear down the internal tracing.
+void ShutdownInternalTracer();
+} // namespace tracing
+} // namespace rtc
+
#endif // WEBRTC_BASE_EVENT_TRACER_H_
diff --git a/webrtc/base/fakenetwork.h b/webrtc/base/fakenetwork.h
index fb99d59..e3996e6 100644
--- a/webrtc/base/fakenetwork.h
+++ b/webrtc/base/fakenetwork.h
@@ -12,6 +12,7 @@
#define WEBRTC_BASE_FAKENETWORK_H_
#include <string>
+#include <utility>
#include <vector>
#include "webrtc/base/network.h"
@@ -31,7 +32,7 @@
public:
FakeNetworkManager() : thread_(Thread::Current()) {}
- typedef std::vector<SocketAddress> IfaceList;
+ typedef std::vector<std::pair<SocketAddress, AdapterType>> IfaceList;
void AddInterface(const SocketAddress& iface) {
// Ensure a unique name for the interface if its name is not given.
@@ -39,16 +40,22 @@
}
void AddInterface(const SocketAddress& iface, const std::string& if_name) {
+ AddInterface(iface, if_name, ADAPTER_TYPE_UNKNOWN);
+ }
+
+ void AddInterface(const SocketAddress& iface,
+ const std::string& if_name,
+ AdapterType type) {
SocketAddress address(if_name, 0);
address.SetResolvedIP(iface.ipaddr());
- ifaces_.push_back(address);
+ ifaces_.push_back(std::make_pair(address, type));
DoUpdateNetworks();
}
void RemoveInterface(const SocketAddress& iface) {
for (IfaceList::iterator it = ifaces_.begin();
it != ifaces_.end(); ++it) {
- if (it->EqualIPs(iface)) {
+ if (it->first.EqualIPs(iface)) {
ifaces_.erase(it);
break;
}
@@ -76,6 +83,7 @@
}
using NetworkManagerBase::set_enumeration_permission;
+ using NetworkManagerBase::set_default_local_addresses;
private:
void DoUpdateNetworks() {
@@ -85,17 +93,17 @@
for (IfaceList::iterator it = ifaces_.begin();
it != ifaces_.end(); ++it) {
int prefix_length = 0;
- if (it->ipaddr().family() == AF_INET) {
+ if (it->first.ipaddr().family() == AF_INET) {
prefix_length = kFakeIPv4NetworkPrefixLength;
- } else if (it->ipaddr().family() == AF_INET6) {
+ } else if (it->first.ipaddr().family() == AF_INET6) {
prefix_length = kFakeIPv6NetworkPrefixLength;
}
- IPAddress prefix = TruncateIP(it->ipaddr(), prefix_length);
- scoped_ptr<Network> net(new Network(it->hostname(),
- it->hostname(),
- prefix,
- prefix_length));
- net->AddIP(it->ipaddr());
+ IPAddress prefix = TruncateIP(it->first.ipaddr(), prefix_length);
+ scoped_ptr<Network> net(new Network(it->first.hostname(),
+ it->first.hostname(), prefix,
+ prefix_length, it->second));
+ net->set_default_local_address_provider(this);
+ net->AddIP(it->first.ipaddr());
networks.push_back(net.release());
}
bool changed;
@@ -111,6 +119,9 @@
int next_index_ = 0;
int start_count_ = 0;
bool sent_first_update_ = false;
+
+ IPAddress default_local_ipv4_address_;
+ IPAddress default_local_ipv6_address_;
};
} // namespace rtc
diff --git a/webrtc/base/fakesslidentity.h b/webrtc/base/fakesslidentity.h
index 7926580..ec603a5 100644
--- a/webrtc/base/fakesslidentity.h
+++ b/webrtc/base/fakesslidentity.h
@@ -25,9 +25,11 @@
// SHA-1 is the default digest algorithm because it is available in all build
// configurations used for unit testing.
explicit FakeSSLCertificate(const std::string& data)
- : data_(data), digest_algorithm_(DIGEST_SHA_1) {}
+ : data_(data), digest_algorithm_(DIGEST_SHA_1), expiration_time_(-1) {}
explicit FakeSSLCertificate(const std::vector<std::string>& certs)
- : data_(certs.front()), digest_algorithm_(DIGEST_SHA_1) {
+ : data_(certs.front()),
+ digest_algorithm_(DIGEST_SHA_1),
+ expiration_time_(-1) {
std::vector<std::string>::const_iterator it;
// Skip certs[0].
for (it = certs.begin() + 1; it != certs.end(); ++it) {
@@ -45,6 +47,12 @@
VERIFY(SSLIdentity::PemToDer(kPemTypeCertificate, data_, &der_string));
der_buffer->SetData(der_string.c_str(), der_string.size());
}
+ int64_t CertificateExpirationTime() const override {
+ return expiration_time_;
+ }
+ void SetCertificateExpirationTime(int64_t expiration_time) {
+ expiration_time_ = expiration_time;
+ }
void set_digest_algorithm(const std::string& algorithm) {
digest_algorithm_ = algorithm;
}
@@ -78,6 +86,8 @@
std::string data_;
std::vector<FakeSSLCertificate> certs_;
std::string digest_algorithm_;
+ // Expiration time in seconds relative to epoch, 1970-01-01T00:00:00Z (UTC).
+ int64_t expiration_time_;
};
class FakeSSLIdentity : public rtc::SSLIdentity {
diff --git a/webrtc/base/filerotatingstream.cc b/webrtc/base/filerotatingstream.cc
index 65dfd63..0809994 100644
--- a/webrtc/base/filerotatingstream.cc
+++ b/webrtc/base/filerotatingstream.cc
@@ -281,7 +281,7 @@
// Rotates the files by deleting the file at |rotation_index_|, which is the
// oldest file and then renaming the newer files to have an incremented index.
// See header file comments for example.
- RTC_DCHECK_LE(rotation_index_, file_names_.size());
+ RTC_DCHECK_LT(rotation_index_, file_names_.size());
std::string file_to_delete = file_names_[rotation_index_];
if (Filesystem::IsFile(file_to_delete)) {
if (!Filesystem::DeleteFile(file_to_delete)) {
diff --git a/webrtc/base/fileutils.cc b/webrtc/base/fileutils.cc
index 6f385d7..cb23153 100644
--- a/webrtc/base/fileutils.cc
+++ b/webrtc/base/fileutils.cc
@@ -10,6 +10,7 @@
#include <assert.h>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/pathutils.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/stringutils.h"
@@ -273,8 +274,8 @@
}
version += 1;
char version_base[MAX_PATH];
- sprintfn(version_base, ARRAY_SIZE(version_base), "%s-%u",
- basename.c_str(), version);
+ sprintfn(version_base, arraysize(version_base), "%s-%u", basename.c_str(),
+ version);
path.SetBasename(version_base);
}
return true;
diff --git a/webrtc/base/format_macros.h b/webrtc/base/format_macros.h
index 5d7dcc3..90f86a6 100644
--- a/webrtc/base/format_macros.h
+++ b/webrtc/base/format_macros.h
@@ -73,6 +73,8 @@
#else // WEBRTC_WIN
+#include <inttypes.h>
+
#if !defined(PRId64)
#define PRId64 "I64d"
#endif
diff --git a/webrtc/base/gunit.h b/webrtc/base/gunit.h
index c2bc844..1a6c363 100644
--- a/webrtc/base/gunit.h
+++ b/webrtc/base/gunit.h
@@ -13,7 +13,7 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
-#if defined(WEBRTC_ANDROID) || defined(GTEST_RELATIVE_PATH)
+#if defined(GTEST_RELATIVE_PATH)
#include "testing/gtest/include/gtest/gtest.h"
#else
#include "testing/base/public/gunit.h"
@@ -35,7 +35,7 @@
rtc::Thread::Current()->ProcessMessages(1); \
res = (ex); \
} \
- } while (0);
+ } while (0)
// The typical EXPECT_XXXX and ASSERT_XXXXs, but done until true or a timeout.
#define EXPECT_TRUE_WAIT(ex, timeout) \
@@ -43,28 +43,28 @@
bool res; \
WAIT_(ex, timeout, res); \
if (!res) EXPECT_TRUE(ex); \
- } while (0);
+ } while (0)
#define EXPECT_EQ_WAIT(v1, v2, timeout) \
do { \
bool res; \
WAIT_(v1 == v2, timeout, res); \
if (!res) EXPECT_EQ(v1, v2); \
- } while (0);
+ } while (0)
#define ASSERT_TRUE_WAIT(ex, timeout) \
do { \
bool res; \
WAIT_(ex, timeout, res); \
if (!res) ASSERT_TRUE(ex); \
- } while (0);
+ } while (0)
#define ASSERT_EQ_WAIT(v1, v2, timeout) \
do { \
bool res; \
WAIT_(v1 == v2, timeout, res); \
if (!res) ASSERT_EQ(v1, v2); \
- } while (0);
+ } while (0)
// Version with a "soft" timeout and a margin. This logs if the timeout is
// exceeded, but it only fails if the expression still isn't true after the
@@ -82,6 +82,6 @@
if (!res) { \
EXPECT_TRUE(ex); \
} \
- } while (0);
+ } while (0)
#endif // WEBRTC_BASE_GUNIT_H_
diff --git a/webrtc/base/helpers.cc b/webrtc/base/helpers.cc
index 8e59b64..1ad5d0e 100644
--- a/webrtc/base/helpers.cc
+++ b/webrtc/base/helpers.cc
@@ -164,17 +164,21 @@
int seed_;
};
-// TODO: Use Base64::Base64Table instead.
-static const char BASE64[64] = {
- 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M',
- 'N', 'O', 'P', 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', 'Y', 'Z',
- 'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm',
- 'n', 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z',
- '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '+', '/'
-};
-
namespace {
+// TODO: Use Base64::Base64Table instead.
+static const char kBase64[64] = {
+ 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M',
+ 'N', 'O', 'P', 'Q', 'R', 'S', 'T', 'U', 'V', 'W', 'X', 'Y', 'Z',
+ 'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm',
+ 'n', 'o', 'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z',
+ '0', '1', '2', '3', '4', '5', '6', '7', '8', '9', '+', '/'};
+
+static const char kHex[16] = {'0', '1', '2', '3', '4', '5', '6', '7',
+ '8', '9', 'a', 'b', 'c', 'd', 'e', 'f'};
+
+static const char kUuidDigit17[4] = {'8', '9', 'a', 'b'};
+
// This round about way of creating a global RNG is to safe-guard against
// indeterminant static initialization order.
scoped_ptr<RandomGenerator>& GetGlobalRng() {
@@ -232,7 +236,7 @@
}
bool CreateRandomString(size_t len, std::string* str) {
- return CreateRandomString(len, BASE64, 64, str);
+ return CreateRandomString(len, kBase64, 64, str);
}
bool CreateRandomString(size_t len, const std::string& table,
@@ -241,6 +245,41 @@
static_cast<int>(table.size()), str);
}
+// Version 4 UUID is of the form:
+// xxxxxxxx-xxxx-4xxx-yxxx-xxxxxxxxxxxx
+// Where 'x' is a hex digit, and 'y' is 8, 9, a or b.
+std::string CreateRandomUuid() {
+ std::string str;
+ scoped_ptr<uint8_t[]> bytes(new uint8_t[31]);
+ if (!Rng().Generate(bytes.get(), 31)) {
+ LOG(LS_ERROR) << "Failed to generate random string!";
+ return str;
+ }
+ str.reserve(36);
+ for (size_t i = 0; i < 8; ++i) {
+ str.push_back(kHex[bytes[i] % 16]);
+ }
+ str.push_back('-');
+ for (size_t i = 8; i < 12; ++i) {
+ str.push_back(kHex[bytes[i] % 16]);
+ }
+ str.push_back('-');
+ str.push_back('4');
+ for (size_t i = 12; i < 15; ++i) {
+ str.push_back(kHex[bytes[i] % 16]);
+ }
+ str.push_back('-');
+ str.push_back(kUuidDigit17[bytes[15] % 4]);
+ for (size_t i = 16; i < 19; ++i) {
+ str.push_back(kHex[bytes[i] % 16]);
+ }
+ str.push_back('-');
+ for (size_t i = 19; i < 31; ++i) {
+ str.push_back(kHex[bytes[i] % 16]);
+ }
+ return str;
+}
+
uint32_t CreateRandomId() {
uint32_t id;
if (!Rng().Generate(&id, sizeof(id))) {
diff --git a/webrtc/base/helpers.h b/webrtc/base/helpers.h
index 102c08b..0e79373 100644
--- a/webrtc/base/helpers.h
+++ b/webrtc/base/helpers.h
@@ -39,6 +39,9 @@
bool CreateRandomString(size_t length, const std::string& table,
std::string* str);
+// Generates a (cryptographically) random UUID version 4 string.
+std::string CreateRandomUuid();
+
// Generates a random id.
uint32_t CreateRandomId();
diff --git a/webrtc/base/helpers_unittest.cc b/webrtc/base/helpers_unittest.cc
index 6ea0167..83cc685 100644
--- a/webrtc/base/helpers_unittest.cc
+++ b/webrtc/base/helpers_unittest.cc
@@ -43,16 +43,23 @@
EXPECT_EQ(256U, random2.size());
}
+TEST_F(RandomTest, TestCreateRandomUuid) {
+ std::string random = CreateRandomUuid();
+ EXPECT_EQ(36U, random.size());
+}
+
TEST_F(RandomTest, TestCreateRandomForTest) {
// Make sure we get the output we expect.
SetRandomTestMode(true);
EXPECT_EQ(2154761789U, CreateRandomId());
EXPECT_EQ("h0ISP4S5SJKH/9EY", CreateRandomString(16));
+ EXPECT_EQ("41706e92-cdd3-46d9-a22d-8ff1737ffb11", CreateRandomUuid());
// Reset and make sure we get the same output.
SetRandomTestMode(true);
EXPECT_EQ(2154761789U, CreateRandomId());
EXPECT_EQ("h0ISP4S5SJKH/9EY", CreateRandomString(16));
+ EXPECT_EQ("41706e92-cdd3-46d9-a22d-8ff1737ffb11", CreateRandomUuid());
// Test different character sets.
SetRandomTestMode(true);
diff --git a/webrtc/base/httpcommon-inl.h b/webrtc/base/httpcommon-inl.h
index d1c0bf0..188d9e6 100644
--- a/webrtc/base/httpcommon-inl.h
+++ b/webrtc/base/httpcommon-inl.h
@@ -11,6 +11,7 @@
#ifndef WEBRTC_BASE_HTTPCOMMON_INL_H__
#define WEBRTC_BASE_HTTPCOMMON_INL_H__
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/common.h"
#include "webrtc/base/httpcommon.h"
@@ -80,7 +81,7 @@
template<class CTYPE>
void Url<CTYPE>::do_get_url(string* val) const {
CTYPE protocol[9];
- asccpyn(protocol, ARRAY_SIZE(protocol), secure_ ? "https://" : "http://");
+ asccpyn(protocol, arraysize(protocol), secure_ ? "https://" : "http://");
val->append(protocol);
do_get_address(val);
do_get_full_path(val);
@@ -91,8 +92,8 @@
val->append(host_);
if (port_ != HttpDefaultPort(secure_)) {
CTYPE format[5], port[32];
- asccpyn(format, ARRAY_SIZE(format), ":%hu");
- sprintfn(port, ARRAY_SIZE(port), format, port_);
+ asccpyn(format, arraysize(format), ":%hu");
+ sprintfn(port, arraysize(port), format, port_);
val->append(port);
}
}
diff --git a/webrtc/base/httpcommon.cc b/webrtc/base/httpcommon.cc
index 0c3547e..c90bea5 100644
--- a/webrtc/base/httpcommon.cc
+++ b/webrtc/base/httpcommon.cc
@@ -21,6 +21,7 @@
#include <algorithm>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/common.h"
#include "webrtc/base/cryptstring.h"
@@ -377,7 +378,7 @@
gmt = non_gmt + ((zone[0] == '+') ? offset : -offset);
} else {
size_t zindex;
- if (!find_string(zindex, zone, kTimeZones, ARRAY_SIZE(kTimeZones))) {
+ if (!find_string(zindex, zone, kTimeZones, arraysize(kTimeZones))) {
return false;
}
gmt = non_gmt + kTimeZoneOffsets[zindex] * 60 * 60;
diff --git a/webrtc/base/ifaddrs_converter.cc b/webrtc/base/ifaddrs_converter.cc
new file mode 100644
index 0000000..7dd3555
--- /dev/null
+++ b/webrtc/base/ifaddrs_converter.cc
@@ -0,0 +1,60 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/ifaddrs_converter.h"
+
+namespace rtc {
+
+IfAddrsConverter::IfAddrsConverter() {}
+
+IfAddrsConverter::~IfAddrsConverter() {}
+
+bool IfAddrsConverter::ConvertIfAddrsToIPAddress(
+ const struct ifaddrs* interface,
+ InterfaceAddress* ip,
+ IPAddress* mask) {
+ switch (interface->ifa_addr->sa_family) {
+ case AF_INET: {
+ *ip = IPAddress(
+ reinterpret_cast<sockaddr_in*>(interface->ifa_addr)->sin_addr);
+ *mask = IPAddress(
+ reinterpret_cast<sockaddr_in*>(interface->ifa_netmask)->sin_addr);
+ return true;
+ }
+ case AF_INET6: {
+ int ip_attributes = IPV6_ADDRESS_FLAG_NONE;
+ if (!ConvertNativeAttributesToIPAttributes(interface, &ip_attributes)) {
+ return false;
+ }
+ *ip = InterfaceAddress(
+ reinterpret_cast<sockaddr_in6*>(interface->ifa_addr)->sin6_addr,
+ ip_attributes);
+ *mask = IPAddress(
+ reinterpret_cast<sockaddr_in6*>(interface->ifa_netmask)->sin6_addr);
+ return true;
+ }
+ default: { return false; }
+ }
+}
+
+bool IfAddrsConverter::ConvertNativeAttributesToIPAttributes(
+ const struct ifaddrs* interface,
+ int* ip_attributes) {
+ *ip_attributes = IPV6_ADDRESS_FLAG_NONE;
+ return true;
+}
+
+#if !defined(WEBRTC_MAC)
+// For MAC and IOS, it's defined in macifaddrs_converter.cc
+IfAddrsConverter* CreateIfAddrsConverter() {
+ return new IfAddrsConverter();
+}
+#endif
+} // namespace rtc
diff --git a/webrtc/base/ifaddrs_converter.h b/webrtc/base/ifaddrs_converter.h
new file mode 100644
index 0000000..0a1cdb9
--- /dev/null
+++ b/webrtc/base/ifaddrs_converter.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_IFADDRS_CONVERTER_H_
+#define WEBRTC_BASE_IFADDRS_CONVERTER_H_
+
+#if defined(WEBRTC_ANDROID)
+#include "webrtc/base/ifaddrs-android.h"
+#else
+#include <ifaddrs.h>
+#endif // WEBRTC_ANDROID
+
+#include "webrtc/base/ipaddress.h"
+
+namespace rtc {
+
+// This class converts native interface addresses to our internal IPAddress
+// class. Subclasses should override ConvertNativeToIPAttributes to implement
+// the different ways of retrieving IPv6 attributes for various POSIX platforms.
+class IfAddrsConverter {
+ public:
+ IfAddrsConverter();
+ virtual ~IfAddrsConverter();
+ virtual bool ConvertIfAddrsToIPAddress(const struct ifaddrs* interface,
+ InterfaceAddress* ipaddress,
+ IPAddress* mask);
+
+ protected:
+ virtual bool ConvertNativeAttributesToIPAttributes(
+ const struct ifaddrs* interface,
+ int* ip_attributes);
+};
+
+IfAddrsConverter* CreateIfAddrsConverter();
+
+} // namespace rtc
+
+#endif // WEBRTC_BASE_IFADDRS_CONVERTER_H_
diff --git a/webrtc/base/ipaddress.cc b/webrtc/base/ipaddress.cc
index 316207f..c92f33c 100644
--- a/webrtc/base/ipaddress.cc
+++ b/webrtc/base/ipaddress.cc
@@ -27,8 +27,10 @@
#include "webrtc/base/ipaddress.h"
#include "webrtc/base/byteorder.h"
-#include "webrtc/base/nethelpers.h"
+#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
+#include "webrtc/base/nethelpers.h"
+#include "webrtc/base/stringutils.h"
#include "webrtc/base/win32.h"
namespace rtc {
@@ -41,8 +43,6 @@
static const in6_addr kV4CompatibilityPrefix = {{{0}}};
static const in6_addr k6BonePrefix = {{{0x3f, 0xfe, 0}}};
-bool IPAddress::strip_sensitive_ = false;
-
static bool IsPrivateV4(uint32_t ip);
static in_addr ExtractMappedAddress(const in6_addr& addr);
@@ -144,9 +144,10 @@
}
std::string IPAddress::ToSensitiveString() const {
- if (!strip_sensitive_)
- return ToString();
-
+#if !defined(NDEBUG)
+ // Return non-stripped in debug.
+ return ToString();
+#else
switch (family_) {
case AF_INET: {
std::string address = ToString();
@@ -158,12 +159,20 @@
return address;
}
case AF_INET6: {
- // TODO(grunell): Return a string of format 1:2:3:x:x:x:x:x or such
- // instead of zeroing out.
- return TruncateIP(*this, 128 - 80).ToString();
+ std::string result;
+ result.resize(INET6_ADDRSTRLEN);
+ in6_addr addr = ipv6_address();
+ size_t len =
+ rtc::sprintfn(&(result[0]), result.size(), "%x:%x:%x:x:x:x:x:x",
+ (addr.s6_addr[0] << 8) + addr.s6_addr[1],
+ (addr.s6_addr[2] << 8) + addr.s6_addr[3],
+ (addr.s6_addr[4] << 8) + addr.s6_addr[5]);
+ result.resize(len);
+ return result;
}
}
return std::string();
+#endif
}
IPAddress IPAddress::Normalized() const {
@@ -186,10 +195,6 @@
return IPAddress(v6addr);
}
-void IPAddress::set_strip_sensitive(bool enable) {
- strip_sensitive_ = enable;
-}
-
bool InterfaceAddress::operator==(const InterfaceAddress &other) const {
return ipv6_flags_ == other.ipv6_flags() &&
static_cast<const IPAddress&>(*this) == other;
@@ -506,4 +511,15 @@
}
return rtc::IPAddress();
}
+
+IPAddress GetAnyIP(int family) {
+ if (family == AF_INET) {
+ return rtc::IPAddress(INADDR_ANY);
+ }
+ if (family == AF_INET6) {
+ return rtc::IPAddress(in6addr_any);
+ }
+ return rtc::IPAddress();
+}
+
} // Namespace rtc
diff --git a/webrtc/base/ipaddress.h b/webrtc/base/ipaddress.h
index fe2d6e2..ef1e3d8 100644
--- a/webrtc/base/ipaddress.h
+++ b/webrtc/base/ipaddress.h
@@ -112,16 +112,12 @@
// Whether this is an unspecified IP address.
bool IsNil() const;
- static void set_strip_sensitive(bool enable);
-
private:
int family_;
union {
in_addr ip4;
in6_addr ip6;
} u_;
-
- static bool strip_sensitive_;
};
// IP class which could represent IPv6 address flags which is only
@@ -180,6 +176,7 @@
IPAddress TruncateIP(const IPAddress& ip, int length);
IPAddress GetLoopbackIP(int family);
+IPAddress GetAnyIP(int family);
// Returns the number of contiguously set bits, counting from the MSB in network
// byte order, in this IPAddress. Bits after the first 0 encountered are not
diff --git a/webrtc/base/ipaddress_unittest.cc b/webrtc/base/ipaddress_unittest.cc
index d5cb6f7..62773c1 100644
--- a/webrtc/base/ipaddress_unittest.cc
+++ b/webrtc/base/ipaddress_unittest.cc
@@ -25,6 +25,10 @@
0x00, 0x04, 0x10, 0x00,
0xbe, 0x30, 0x5b, 0xff,
0xfe, 0xe5, 0x00, 0xc3}}};
+static const in6_addr kIPv6PublicAddr2 = {{{0x24, 0x01, 0x00, 0x00,
+ 0x00, 0x00, 0x10, 0x00,
+ 0xbe, 0x30, 0x5b, 0xff,
+ 0xfe, 0xe5, 0x00, 0xc3}}};
static const in6_addr kIPv4MappedAnyAddr = {{{0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0xff, 0xff,
@@ -52,7 +56,12 @@
"2620:0:1008:1201:2089:6dda:385e:80c0";
static const std::string kIPv6PublicAddrString =
"2401:fa00:4:1000:be30:5bff:fee5:c3";
-static const std::string kIPv6PublicAddrAnonymizedString = "2401:fa00:4::";
+static const std::string kIPv6PublicAddr2String =
+ "2401::1000:be30:5bff:fee5:c3";
+static const std::string kIPv6PublicAddrAnonymizedString =
+ "2401:fa00:4:x:x:x:x:x";
+static const std::string kIPv6PublicAddr2AnonymizedString =
+ "2401:0:0:x:x:x:x:x";
static const std::string kIPv4MappedAnyAddrString = "::ffff:0:0";
static const std::string kIPv4MappedRFC1918AddrString = "::ffff:c0a8:701";
static const std::string kIPv4MappedLoopbackAddrString = "::ffff:7f00:1";
@@ -888,20 +897,20 @@
TEST(IPAddressTest, TestToSensitiveString) {
IPAddress addr_v4 = IPAddress(kIPv4PublicAddr);
- EXPECT_EQ(kIPv4PublicAddrString, addr_v4.ToString());
- EXPECT_EQ(kIPv4PublicAddrString, addr_v4.ToSensitiveString());
- IPAddress::set_strip_sensitive(true);
- EXPECT_EQ(kIPv4PublicAddrString, addr_v4.ToString());
- EXPECT_EQ(kIPv4PublicAddrAnonymizedString, addr_v4.ToSensitiveString());
- IPAddress::set_strip_sensitive(false);
-
IPAddress addr_v6 = IPAddress(kIPv6PublicAddr);
+ IPAddress addr_v6_2 = IPAddress(kIPv6PublicAddr2);
+ EXPECT_EQ(kIPv4PublicAddrString, addr_v4.ToString());
EXPECT_EQ(kIPv6PublicAddrString, addr_v6.ToString());
- EXPECT_EQ(kIPv6PublicAddrString, addr_v6.ToSensitiveString());
- IPAddress::set_strip_sensitive(true);
- EXPECT_EQ(kIPv6PublicAddrString, addr_v6.ToString());
+ EXPECT_EQ(kIPv6PublicAddr2String, addr_v6_2.ToString());
+#if defined(NDEBUG)
+ EXPECT_EQ(kIPv4PublicAddrAnonymizedString, addr_v4.ToSensitiveString());
EXPECT_EQ(kIPv6PublicAddrAnonymizedString, addr_v6.ToSensitiveString());
- IPAddress::set_strip_sensitive(false);
+ EXPECT_EQ(kIPv6PublicAddr2AnonymizedString, addr_v6_2.ToSensitiveString());
+#else
+ EXPECT_EQ(kIPv4PublicAddrString, addr_v4.ToSensitiveString());
+ EXPECT_EQ(kIPv6PublicAddrString, addr_v6.ToSensitiveString());
+ EXPECT_EQ(kIPv6PublicAddr2String, addr_v6_2.ToSensitiveString());
+#endif // defined(NDEBUG)
}
TEST(IPAddressTest, TestInterfaceAddress) {
diff --git a/webrtc/base/keep_ref_until_done.h b/webrtc/base/keep_ref_until_done.h
new file mode 100644
index 0000000..269e1c8
--- /dev/null
+++ b/webrtc/base/keep_ref_until_done.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_
+#define WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_
+
+#include "webrtc/base/bind.h"
+#include "webrtc/base/callback.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+
+namespace rtc {
+
+namespace impl {
+template <class T>
+static inline void DoNothing(const scoped_refptr<T>& object) {}
+} // namespace impl
+
+// KeepRefUntilDone keeps a reference to |object| until the returned
+// callback goes out of scope. If the returned callback is copied, the
+// reference will be released when the last callback goes out of scope.
+template <class ObjectT>
+static inline Callback0<void> KeepRefUntilDone(ObjectT* object) {
+ return rtc::Bind(&impl::DoNothing<ObjectT>, scoped_refptr<ObjectT>(object));
+}
+
+template <class ObjectT>
+static inline Callback0<void> KeepRefUntilDone(
+ const scoped_refptr<ObjectT>& object) {
+ return rtc::Bind(&impl::DoNothing<ObjectT>, object);
+}
+
+} // namespace rtc
+
+
+#endif // WEBRTC_BASE_KEEP_REF_UNTIL_DONE_H_
diff --git a/webrtc/base/latebindingsymboltable_unittest.cc b/webrtc/base/latebindingsymboltable_unittest.cc
index 30ebd17..0079f20 100644
--- a/webrtc/base/latebindingsymboltable_unittest.cc
+++ b/webrtc/base/latebindingsymboltable_unittest.cc
@@ -21,9 +21,10 @@
#define LIBM_SYMBOLS_CLASS_NAME LibmTestSymbolTable
#define LIBM_SYMBOLS_LIST \
- X(acos) \
- X(sin) \
- X(tan)
+ X(acosf) \
+ X(sinf) \
+ X(tanf)
+
#define LATE_BINDING_SYMBOL_TABLE_CLASS_NAME LIBM_SYMBOLS_CLASS_NAME
#define LATE_BINDING_SYMBOL_TABLE_SYMBOLS_LIST LIBM_SYMBOLS_LIST
@@ -39,9 +40,9 @@
EXPECT_FALSE(table.IsLoaded());
ASSERT_TRUE(table.Load());
EXPECT_TRUE(table.IsLoaded());
- EXPECT_EQ(table.acos()(0.5), acos(0.5));
- EXPECT_EQ(table.sin()(0.5), sin(0.5));
- EXPECT_EQ(table.tan()(0.5), tan(0.5));
+ EXPECT_EQ(table.acosf()(0.5f), acosf(0.5f));
+ EXPECT_EQ(table.sinf()(0.5f), sinf(0.5f));
+ EXPECT_EQ(table.tanf()(0.5f), tanf(0.5f));
// It would be nice to check that the addresses are the same, but the nature
// of dynamic linking and relocation makes them actually be different.
table.Unload();
diff --git a/webrtc/base/linux.cc b/webrtc/base/linux.cc
index 1586b27..0894d39 100644
--- a/webrtc/base/linux.cc
+++ b/webrtc/base/linux.cc
@@ -233,33 +233,6 @@
return true;
}
-#if !defined(WEBRTC_CHROMIUM_BUILD)
-static bool ExpectLineFromStream(FileStream* stream,
- std::string* out) {
- StreamResult res = stream->ReadLine(out);
- if (res != SR_SUCCESS) {
- if (res != SR_EOS) {
- LOG(LS_ERROR) << "Error when reading from stream";
- } else {
- LOG(LS_ERROR) << "Incorrect number of lines in stream";
- }
- return false;
- }
- return true;
-}
-
-static void ExpectEofFromStream(FileStream* stream) {
- std::string unused;
- StreamResult res = stream->ReadLine(&unused);
- if (res == SR_SUCCESS) {
- LOG(LS_WARNING) << "Ignoring unexpected extra lines from stream";
- } else if (res != SR_EOS) {
- LOG(LS_WARNING) << "Error when checking for extra lines from stream";
- }
-}
-
-#endif
-
std::string ReadLinuxUname() {
struct utsname buf;
if (uname(&buf) < 0) {
diff --git a/webrtc/base/logging.cc b/webrtc/base/logging.cc
index b02be27..686b9b2 100644
--- a/webrtc/base/logging.cc
+++ b/webrtc/base/logging.cc
@@ -92,13 +92,13 @@
/////////////////////////////////////////////////////////////////////////////
// By default, release builds don't log, debug builds at info level
-#if _DEBUG
+#if !defined(NDEBUG)
LoggingSeverity LogMessage::min_sev_ = LS_INFO;
LoggingSeverity LogMessage::dbg_sev_ = LS_INFO;
-#else // !_DEBUG
+#else
LoggingSeverity LogMessage::min_sev_ = LS_NONE;
LoggingSeverity LogMessage::dbg_sev_ = LS_NONE;
-#endif // !_DEBUG
+#endif
bool LogMessage::log_to_stderr_ = true;
namespace {
@@ -340,7 +340,7 @@
LoggingSeverity severity,
const std::string& tag) {
bool log_to_stderr = log_to_stderr_;
-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) && (!defined(_DEBUG) || defined(NDEBUG))
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) && defined(NDEBUG)
// On the Mac, all stderr output goes to the Console log and causes clutter.
// So in opt builds, don't log to stderr unless the user specifically sets
// a preference to do so.
diff --git a/webrtc/base/logging.h b/webrtc/base/logging.h
index 1208275..e40ca44 100644
--- a/webrtc/base/logging.h
+++ b/webrtc/base/logging.h
@@ -285,7 +285,7 @@
rtc::LogMessage(__FILE__, __LINE__, sev).stream()
// The _F version prefixes the message with the current function name.
-#if (defined(__GNUC__) && defined(_DEBUG)) || defined(WANT_PRETTY_LOG_F)
+#if (defined(__GNUC__) && !defined(NDEBUG)) || defined(WANT_PRETTY_LOG_F)
#define LOG_F(sev) LOG(sev) << __PRETTY_FUNCTION__ << ": "
#define LOG_T_F(sev) LOG(sev) << this << ": " << __PRETTY_FUNCTION__ << ": "
#else
diff --git a/webrtc/base/logging_unittest.cc b/webrtc/base/logging_unittest.cc
index 3719cde..6047361 100644
--- a/webrtc/base/logging_unittest.cc
+++ b/webrtc/base/logging_unittest.cc
@@ -14,7 +14,6 @@
#include "webrtc/base/pathutils.h"
#include "webrtc/base/stream.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/macasyncsocket.cc b/webrtc/base/macasyncsocket.cc
index 1f12500..8f811ea 100644
--- a/webrtc/base/macasyncsocket.cc
+++ b/webrtc/base/macasyncsocket.cc
@@ -112,7 +112,7 @@
SetError(EALREADY);
return SOCKET_ERROR;
}
- if (addr.IsUnresolved()) {
+ if (addr.IsUnresolvedIP()) {
LOG(LS_VERBOSE) << "Resolving addr in MacAsyncSocket::Connect";
resolver_ = new AsyncResolver();
resolver_->SignalWorkDone.connect(this,
diff --git a/webrtc/base/macifaddrs_converter.cc b/webrtc/base/macifaddrs_converter.cc
new file mode 100644
index 0000000..0916cb5
--- /dev/null
+++ b/webrtc/base/macifaddrs_converter.cc
@@ -0,0 +1,281 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <net/if.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/ifaddrs_converter.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/scoped_ptr.h"
+
+#if !defined(WEBRTC_IOS)
+#include <net/if_media.h>
+#include <netinet/in_var.h>
+#else // WEBRTC_IOS
+#define SCOPE6_ID_MAX 16
+
+struct in6_addrlifetime {
+ time_t ia6t_expire; /* valid lifetime expiration time */
+ time_t ia6t_preferred; /* preferred lifetime expiration time */
+ u_int32_t ia6t_vltime; /* valid lifetime */
+ u_int32_t ia6t_pltime; /* prefix lifetime */
+};
+
+struct in6_ifstat {
+ u_quad_t ifs6_in_receive; /* # of total input datagram */
+ u_quad_t ifs6_in_hdrerr; /* # of datagrams with invalid hdr */
+ u_quad_t ifs6_in_toobig; /* # of datagrams exceeded MTU */
+ u_quad_t ifs6_in_noroute; /* # of datagrams with no route */
+ u_quad_t ifs6_in_addrerr; /* # of datagrams with invalid dst */
+ u_quad_t ifs6_in_protounknown; /* # of datagrams with unknown proto */
+ /* NOTE: increment on final dst if */
+ u_quad_t ifs6_in_truncated; /* # of truncated datagrams */
+ u_quad_t ifs6_in_discard; /* # of discarded datagrams */
+ /* NOTE: fragment timeout is not here */
+ u_quad_t ifs6_in_deliver; /* # of datagrams delivered to ULP */
+ /* NOTE: increment on final dst if */
+ u_quad_t ifs6_out_forward; /* # of datagrams forwarded */
+ /* NOTE: increment on outgoing if */
+ u_quad_t ifs6_out_request; /* # of outgoing datagrams from ULP */
+ /* NOTE: does not include forwrads */
+ u_quad_t ifs6_out_discard; /* # of discarded datagrams */
+ u_quad_t ifs6_out_fragok; /* # of datagrams fragmented */
+ u_quad_t ifs6_out_fragfail; /* # of datagrams failed on fragment */
+ u_quad_t ifs6_out_fragcreat; /* # of fragment datagrams */
+ /* NOTE: this is # after fragment */
+ u_quad_t ifs6_reass_reqd; /* # of incoming fragmented packets */
+ /* NOTE: increment on final dst if */
+ u_quad_t ifs6_reass_ok; /* # of reassembled packets */
+ /* NOTE: this is # after reass */
+ /* NOTE: increment on final dst if */
+ u_quad_t ifs6_reass_fail; /* # of reass failures */
+ /* NOTE: may not be packet count */
+ /* NOTE: increment on final dst if */
+ u_quad_t ifs6_in_mcast; /* # of inbound multicast datagrams */
+ u_quad_t ifs6_out_mcast; /* # of outbound multicast datagrams */
+};
+struct icmp6_ifstat {
+ /*
+ * Input statistics
+ */
+ /* ipv6IfIcmpInMsgs, total # of input messages */
+ u_quad_t ifs6_in_msg;
+ /* ipv6IfIcmpInErrors, # of input error messages */
+ u_quad_t ifs6_in_error;
+ /* ipv6IfIcmpInDestUnreachs, # of input dest unreach errors */
+ u_quad_t ifs6_in_dstunreach;
+ /* ipv6IfIcmpInAdminProhibs, # of input admin. prohibited errs */
+ u_quad_t ifs6_in_adminprohib;
+ /* ipv6IfIcmpInTimeExcds, # of input time exceeded errors */
+ u_quad_t ifs6_in_timeexceed;
+ /* ipv6IfIcmpInParmProblems, # of input parameter problem errors */
+ u_quad_t ifs6_in_paramprob;
+ /* ipv6IfIcmpInPktTooBigs, # of input packet too big errors */
+ u_quad_t ifs6_in_pkttoobig;
+ /* ipv6IfIcmpInEchos, # of input echo requests */
+ u_quad_t ifs6_in_echo;
+ /* ipv6IfIcmpInEchoReplies, # of input echo replies */
+ u_quad_t ifs6_in_echoreply;
+ /* ipv6IfIcmpInRouterSolicits, # of input router solicitations */
+ u_quad_t ifs6_in_routersolicit;
+ /* ipv6IfIcmpInRouterAdvertisements, # of input router advertisements */
+ u_quad_t ifs6_in_routeradvert;
+ /* ipv6IfIcmpInNeighborSolicits, # of input neighbor solicitations */
+ u_quad_t ifs6_in_neighborsolicit;
+ /* ipv6IfIcmpInNeighborAdvertisements, # of input neighbor advs. */
+ u_quad_t ifs6_in_neighboradvert;
+ /* ipv6IfIcmpInRedirects, # of input redirects */
+ u_quad_t ifs6_in_redirect;
+ /* ipv6IfIcmpInGroupMembQueries, # of input MLD queries */
+ u_quad_t ifs6_in_mldquery;
+ /* ipv6IfIcmpInGroupMembResponses, # of input MLD reports */
+ u_quad_t ifs6_in_mldreport;
+ /* ipv6IfIcmpInGroupMembReductions, # of input MLD done */
+ u_quad_t ifs6_in_mlddone;
+
+ /*
+ * Output statistics. We should solve unresolved routing problem...
+ */
+ /* ipv6IfIcmpOutMsgs, total # of output messages */
+ u_quad_t ifs6_out_msg;
+ /* ipv6IfIcmpOutErrors, # of output error messages */
+ u_quad_t ifs6_out_error;
+ /* ipv6IfIcmpOutDestUnreachs, # of output dest unreach errors */
+ u_quad_t ifs6_out_dstunreach;
+ /* ipv6IfIcmpOutAdminProhibs, # of output admin. prohibited errs */
+ u_quad_t ifs6_out_adminprohib;
+ /* ipv6IfIcmpOutTimeExcds, # of output time exceeded errors */
+ u_quad_t ifs6_out_timeexceed;
+ /* ipv6IfIcmpOutParmProblems, # of output parameter problem errors */
+ u_quad_t ifs6_out_paramprob;
+ /* ipv6IfIcmpOutPktTooBigs, # of output packet too big errors */
+ u_quad_t ifs6_out_pkttoobig;
+ /* ipv6IfIcmpOutEchos, # of output echo requests */
+ u_quad_t ifs6_out_echo;
+ /* ipv6IfIcmpOutEchoReplies, # of output echo replies */
+ u_quad_t ifs6_out_echoreply;
+ /* ipv6IfIcmpOutRouterSolicits, # of output router solicitations */
+ u_quad_t ifs6_out_routersolicit;
+ /* ipv6IfIcmpOutRouterAdvertisements, # of output router advs. */
+ u_quad_t ifs6_out_routeradvert;
+ /* ipv6IfIcmpOutNeighborSolicits, # of output neighbor solicitations */
+ u_quad_t ifs6_out_neighborsolicit;
+ /* ipv6IfIcmpOutNeighborAdvertisements, # of output neighbor advs. */
+ u_quad_t ifs6_out_neighboradvert;
+ /* ipv6IfIcmpOutRedirects, # of output redirects */
+ u_quad_t ifs6_out_redirect;
+ /* ipv6IfIcmpOutGroupMembQueries, # of output MLD queries */
+ u_quad_t ifs6_out_mldquery;
+ /* ipv6IfIcmpOutGroupMembResponses, # of output MLD reports */
+ u_quad_t ifs6_out_mldreport;
+ /* ipv6IfIcmpOutGroupMembReductions, # of output MLD done */
+ u_quad_t ifs6_out_mlddone;
+};
+
+struct in6_ifreq {
+ char ifr_name[IFNAMSIZ];
+ union {
+ struct sockaddr_in6 ifru_addr;
+ struct sockaddr_in6 ifru_dstaddr;
+ int ifru_flags;
+ int ifru_flags6;
+ int ifru_metric;
+ int ifru_intval;
+ caddr_t ifru_data;
+ struct in6_addrlifetime ifru_lifetime;
+ struct in6_ifstat ifru_stat;
+ struct icmp6_ifstat ifru_icmp6stat;
+ u_int32_t ifru_scope_id[SCOPE6_ID_MAX];
+ } ifr_ifru;
+};
+
+#define SIOCGIFAFLAG_IN6 _IOWR('i', 73, struct in6_ifreq)
+
+#define IN6_IFF_ANYCAST 0x0001 /* anycast address */
+#define IN6_IFF_TENTATIVE 0x0002 /* tentative address */
+#define IN6_IFF_DUPLICATED 0x0004 /* DAD detected duplicate */
+#define IN6_IFF_DETACHED 0x0008 /* may be detached from the link */
+#define IN6_IFF_DEPRECATED 0x0010 /* deprecated address */
+#define IN6_IFF_TEMPORARY 0x0080 /* temporary (anonymous) address. */
+
+#endif // WEBRTC_IOS
+
+namespace rtc {
+
+namespace {
+
+class IPv6AttributesGetter {
+ public:
+ IPv6AttributesGetter();
+ virtual ~IPv6AttributesGetter();
+ bool IsInitialized() const;
+ bool GetIPAttributes(const char* ifname,
+ const sockaddr* sock_addr,
+ int* native_attributes);
+
+ private:
+ // on MAC or IOS, we have to use ioctl with a socket to query an IPv6
+ // interface's attribute.
+ int ioctl_socket_;
+};
+
+IPv6AttributesGetter::IPv6AttributesGetter()
+ : ioctl_socket_(
+ socket(AF_INET6, SOCK_DGRAM, 0 /* unspecified protocol */)) {
+ RTC_DCHECK_GE(ioctl_socket_, 0);
+}
+
+bool IPv6AttributesGetter::IsInitialized() const {
+ return ioctl_socket_ >= 0;
+}
+
+IPv6AttributesGetter::~IPv6AttributesGetter() {
+ if (!IsInitialized()) {
+ return;
+ }
+ close(ioctl_socket_);
+}
+
+bool IPv6AttributesGetter::GetIPAttributes(const char* ifname,
+ const sockaddr* sock_addr,
+ int* native_attributes) {
+ if (!IsInitialized()) {
+ return false;
+ }
+
+ struct in6_ifreq ifr = {};
+ strncpy(ifr.ifr_name, ifname, sizeof(ifr.ifr_name) - 1);
+ memcpy(&ifr.ifr_ifru.ifru_addr, sock_addr, sock_addr->sa_len);
+ int rv = ioctl(ioctl_socket_, SIOCGIFAFLAG_IN6, &ifr);
+ if (rv >= 0) {
+ *native_attributes = ifr.ifr_ifru.ifru_flags;
+ } else {
+ LOG(LS_ERROR) << "ioctl returns " << errno;
+ }
+ return (rv >= 0);
+}
+
+// Converts native IPv6 address attributes to net IPv6 address attributes. If
+// it returns false, the IP address isn't suitable for one-to-one communications
+// applications and should be ignored.
+bool ConvertNativeToIPAttributes(int native_attributes, int* net_attributes) {
+ // For MacOSX, we disallow addresses with attributes IN6_IFF_ANYCASE,
+ // IN6_IFF_DUPLICATED, IN6_IFF_TENTATIVE, and IN6_IFF_DETACHED as these are
+ // still progressing through duplicated address detection (DAD) or are not
+ // suitable for one-to-one communication applications.
+ if (native_attributes & (IN6_IFF_ANYCAST | IN6_IFF_DUPLICATED |
+ IN6_IFF_TENTATIVE | IN6_IFF_DETACHED)) {
+ return false;
+ }
+
+ if (native_attributes & IN6_IFF_TEMPORARY) {
+ *net_attributes |= IPV6_ADDRESS_FLAG_TEMPORARY;
+ }
+
+ if (native_attributes & IN6_IFF_DEPRECATED) {
+ *net_attributes |= IPV6_ADDRESS_FLAG_DEPRECATED;
+ }
+
+ return true;
+}
+
+class MacIfAddrsConverter : public IfAddrsConverter {
+ public:
+ MacIfAddrsConverter() : ip_attribute_getter_(new IPv6AttributesGetter()) {}
+ ~MacIfAddrsConverter() override {}
+
+ bool ConvertNativeAttributesToIPAttributes(const struct ifaddrs* interface,
+ int* ip_attributes) override {
+ int native_attributes;
+ if (!ip_attribute_getter_->GetIPAttributes(
+ interface->ifa_name, interface->ifa_addr, &native_attributes)) {
+ return false;
+ }
+
+ if (!ConvertNativeToIPAttributes(native_attributes, ip_attributes)) {
+ return false;
+ }
+
+ return true;
+ }
+
+ private:
+ rtc::scoped_ptr<IPv6AttributesGetter> ip_attribute_getter_;
+};
+
+} // namespace
+
+IfAddrsConverter* CreateIfAddrsConverter() {
+ return new MacIfAddrsConverter();
+}
+
+} // namespace rtc
diff --git a/webrtc/base/macutils.cc b/webrtc/base/macutils.cc
index 6e436d4..7b1ff47 100644
--- a/webrtc/base/macutils.cc
+++ b/webrtc/base/macutils.cc
@@ -191,10 +191,10 @@
AECreateDesc(typeNull, NULL, 0, &result_data);
OSAScriptError(component, kOSAErrorMessage, typeChar, &result_data);
int len = AEGetDescDataSize(&result_data);
- char* data = (char*) malloc(len);
+ char* data = (char*)malloc(len);
if (data != NULL) {
err = AEGetDescData(&result_data, data, len);
- LOG(LS_ERROR) << "Script error: " << data;
+ LOG(LS_ERROR) << "Script error: " << std::string(data, len);
}
AEDisposeDesc(&script_desc);
AEDisposeDesc(&result_data);
diff --git a/webrtc/base/maybe.h b/webrtc/base/maybe.h
deleted file mode 100644
index df20436..0000000
--- a/webrtc/base/maybe.h
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- * Copyright 2015 The WebRTC Project Authors. All rights reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_BASE_MAYBE_H_
-#define WEBRTC_BASE_MAYBE_H_
-
-#include <algorithm>
-#include <utility>
-
-#include "webrtc/base/checks.h"
-
-namespace rtc {
-
-// Simple std::experimental::optional-wannabe. It either contains a T or not.
-// In order to keep the implementation simple and portable, this implementation
-// actually contains a (default-constructed) T even when it supposedly doesn't
-// contain a value; use e.g. rtc::scoped_ptr<T> instead if that's too
-// expensive.
-//
-// A moved-from Maybe<T> may only be destroyed, and assigned to if T allows
-// being assigned to after having been moved from. Specifically, you may not
-// assume that it just doesn't contain a value anymore.
-//
-// TODO(kwiberg): Get rid of this class when the standard library has
-// std::optional (and we're allowed to use it).
-template <typename T>
-class Maybe final {
- public:
- // Construct an empty Maybe.
- Maybe() : has_value_(false) {}
-
- // Construct a Maybe that contains a value. Note: These are non-explicit, so
- // that a T will implicitly convert to Maybe<T>.
- Maybe(const T& val) : value_(val), has_value_(true) {}
- Maybe(T&& val) : value_(static_cast<T&&>(val)), has_value_(true) {}
-
- // Copy and move constructors.
- // TODO(kwiberg): =default the move constructor when MSVC supports it.
- Maybe(const Maybe&) = default;
- Maybe(Maybe&& m)
- : value_(static_cast<T&&>(m.value_)), has_value_(m.has_value_) {}
-
- // Assignment. Note that we allow assignment from either Maybe<T> or plain T.
- // TODO(kwiberg): =default the move assignment op when MSVC supports it.
- Maybe& operator=(const Maybe&) = default;
- Maybe& operator=(Maybe&& m) {
- value_ = static_cast<T&&>(m.value_);
- has_value_ = m.has_value_;
- return *this;
- }
- Maybe& operator=(const T& val) {
- value_ = val;
- has_value_ = true;
- return *this;
- }
- Maybe& operator=(T&& val) {
- value_ = static_cast<T&&>(val);
- has_value_ = true;
- return *this;
- }
-
- friend void swap(Maybe& m1, Maybe& m2) {
- using std::swap;
- swap(m1.value_, m2.value_);
- swap(m1.has_value_, m2.has_value_);
- }
-
- // Conversion to bool to test if we have a value.
- explicit operator bool() const { return has_value_; }
-
- // Dereferencing. Only allowed if we have a value.
- const T* operator->() const {
- RTC_DCHECK(has_value_);
- return &value_;
- }
- T* operator->() {
- RTC_DCHECK(has_value_);
- return &value_;
- }
- const T& operator*() const {
- RTC_DCHECK(has_value_);
- return value_;
- }
- T& operator*() {
- RTC_DCHECK(has_value_);
- return value_;
- }
-
- // Dereference with a default value in case we don't have a value.
- const T& value_or(const T& default_val) const {
- return has_value_ ? value_ : default_val;
- }
-
- // Equality tests. Two Maybes are equal if they contain equivalent values, or
- // if they're both empty.
- friend bool operator==(const Maybe& m1, const Maybe& m2) {
- return m1.has_value_ && m2.has_value_ ? m1.value_ == m2.value_
- : m1.has_value_ == m2.has_value_;
- }
- friend bool operator!=(const Maybe& m1, const Maybe& m2) {
- return m1.has_value_ && m2.has_value_ ? m1.value_ != m2.value_
- : m1.has_value_ != m2.has_value_;
- }
-
- private:
- // Invariant: Unless *this has been moved from, value_ is default-initialized
- // (or copied or moved from a default-initialized T) if !has_value_.
- T value_;
- bool has_value_;
-};
-
-} // namespace rtc
-
-#endif // WEBRTC_BASE_MAYBE_H_
diff --git a/webrtc/base/maybe_unittest.cc b/webrtc/base/maybe_unittest.cc
deleted file mode 100644
index 73fdc90..0000000
--- a/webrtc/base/maybe_unittest.cc
+++ /dev/null
@@ -1,485 +0,0 @@
-/*
- * Copyright 2015 The WebRTC Project Authors. All rights reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <sstream>
-#include <string>
-#include <utility>
-#include <vector>
-
-#include "webrtc/base/gunit.h"
-#include "webrtc/base/maybe.h"
-
-namespace rtc {
-
-namespace {
-
-// Class whose instances logs various method calls (constructor, destructor,
-// etc.). Each instance has a unique ID (a simple global sequence number) and
-// an origin ID. When a copy is made, the new object gets a fresh ID but copies
-// the origin ID from the original. When a new Logger is created from scratch,
-// it gets a fresh ID, and the origin ID is the same as the ID (default
-// constructor) or given as an argument (explicit constructor).
-class Logger {
- public:
- Logger() : id_(next_id_++), origin_(id_) { Log("default constructor"); }
- explicit Logger(int origin) : id_(next_id_++), origin_(origin) {
- Log("explicit constructor");
- }
- Logger(const Logger& other) : id_(next_id_++), origin_(other.origin_) {
- LogFrom("copy constructor", other);
- }
- Logger(Logger&& other) : id_(next_id_++), origin_(other.origin_) {
- LogFrom("move constructor", other);
- }
- ~Logger() { Log("destructor"); }
- Logger& operator=(const Logger& other) {
- origin_ = other.origin_;
- LogFrom("operator= copy", other);
- return *this;
- }
- Logger& operator=(Logger&& other) {
- origin_ = other.origin_;
- LogFrom("operator= move", other);
- return *this;
- }
- friend void swap(Logger& a, Logger& b) {
- using std::swap;
- swap(a.origin_, b.origin_);
- Log2("swap", a, b);
- }
- friend bool operator==(const Logger& a, const Logger& b) {
- Log2("operator==", a, b);
- return a.origin_ == b.origin_;
- }
- friend bool operator!=(const Logger& a, const Logger& b) {
- Log2("operator!=", a, b);
- return a.origin_ != b.origin_;
- }
- void Foo() { Log("Foo()"); }
- void Foo() const { Log("Foo() const"); }
- static rtc::scoped_ptr<std::vector<std::string>> Setup() {
- auto s = rtc_make_scoped_ptr(new std::vector<std::string>);
- Logger::log_ = s.get();
- Logger::next_id_ = 0;
- return s;
- }
-
- private:
- int id_;
- int origin_;
- static std::vector<std::string>* log_;
- static int next_id_;
- void Log(const char* msg) const {
- std::ostringstream oss;
- oss << id_ << ':' << origin_ << ". " << msg;
- log_->push_back(oss.str());
- }
- void LogFrom(const char* msg, const Logger& other) const {
- std::ostringstream oss;
- oss << id_ << ':' << origin_ << ". " << msg << " (from " << other.id_ << ':'
- << other.origin_ << ")";
- log_->push_back(oss.str());
- }
- static void Log2(const char* msg, const Logger& a, const Logger& b) {
- std::ostringstream oss;
- oss << msg << ' ' << a.id_ << ':' << a.origin_ << ", " << b.id_ << ':'
- << b.origin_;
- log_->push_back(oss.str());
- }
-};
-
-std::vector<std::string>* Logger::log_ = nullptr;
-int Logger::next_id_ = 0;
-
-// Append all the other args to the vector pointed to by the first arg.
-template <typename T>
-void VectorAppend(std::vector<T>* v) {}
-template <typename T, typename... Ts>
-void VectorAppend(std::vector<T>* v, const T& e, Ts... es) {
- v->push_back(e);
- VectorAppend(v, es...);
-}
-
-// Create a vector of strings. Because we're not allowed to use
-// std::initializer_list.
-template <typename... Ts>
-std::vector<std::string> V(Ts... es) {
- std::vector<std::string> strings;
- VectorAppend(&strings, static_cast<std::string>(es)...);
- return strings;
-}
-
-} // namespace
-
-TEST(MaybeTest, TestConstructDefault) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x;
- EXPECT_FALSE(x);
- }
- EXPECT_EQ(V("0:0. default constructor", "0:0. destructor"), *log);
-}
-
-TEST(MaybeTest, TestConstructCopyEmpty) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x;
- EXPECT_FALSE(x);
- auto y = x;
- EXPECT_FALSE(y);
- }
- EXPECT_EQ(V("0:0. default constructor", "1:0. copy constructor (from 0:0)",
- "1:0. destructor", "0:0. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestConstructCopyFull) {
- auto log = Logger::Setup();
- {
- Logger a;
- Maybe<Logger> x = a;
- EXPECT_TRUE(x);
- log->push_back("---");
- auto y = x;
- EXPECT_TRUE(y);
- log->push_back("---");
- }
- EXPECT_EQ(V("0:0. default constructor", "1:0. copy constructor (from 0:0)",
- "---", "2:0. copy constructor (from 1:0)", "---",
- "2:0. destructor", "1:0. destructor", "0:0. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestConstructMoveEmpty) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x;
- EXPECT_FALSE(x);
- auto y = static_cast<Maybe<Logger>&&>(x);
- EXPECT_FALSE(y);
- }
- EXPECT_EQ(V("0:0. default constructor", "1:0. move constructor (from 0:0)",
- "1:0. destructor", "0:0. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestConstructMoveFull) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x = Logger(17);
- EXPECT_TRUE(x);
- log->push_back("---");
- auto y = static_cast<Maybe<Logger>&&>(x);
- EXPECT_TRUE(x);
- EXPECT_TRUE(y);
- log->push_back("---");
- }
- EXPECT_EQ(
- V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
- "0:17. destructor", "---", "2:17. move constructor (from 1:17)", "---",
- "2:17. destructor", "1:17. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestCopyAssignToEmptyFromEmpty) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x, y;
- x = y;
- }
- EXPECT_EQ(
- V("0:0. default constructor", "1:1. default constructor",
- "0:1. operator= copy (from 1:1)", "1:1. destructor", "0:1. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestCopyAssignToFullFromEmpty) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x = Logger(17);
- Maybe<Logger> y;
- log->push_back("---");
- x = y;
- log->push_back("---");
- }
- EXPECT_EQ(
- V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
- "0:17. destructor", "2:2. default constructor", "---",
- "1:2. operator= copy (from 2:2)", "---", "2:2. destructor",
- "1:2. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestCopyAssignToEmptyFromFull) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x;
- Maybe<Logger> y = Logger(17);
- log->push_back("---");
- x = y;
- log->push_back("---");
- }
- EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor",
- "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
- "0:17. operator= copy (from 2:17)", "---", "2:17. destructor",
- "0:17. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestCopyAssignToFullFromFull) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x = Logger(17);
- Maybe<Logger> y = Logger(42);
- log->push_back("---");
- x = y;
- log->push_back("---");
- }
- EXPECT_EQ(
- V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
- "0:17. destructor", "2:42. explicit constructor",
- "3:42. move constructor (from 2:42)", "2:42. destructor", "---",
- "1:42. operator= copy (from 3:42)", "---", "3:42. destructor",
- "1:42. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestCopyAssignToEmptyFromT) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x;
- Logger y(17);
- log->push_back("---");
- x = y;
- log->push_back("---");
- }
- EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor", "---",
- "0:17. operator= copy (from 1:17)", "---", "1:17. destructor",
- "0:17. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestCopyAssignToFullFromT) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x = Logger(17);
- Logger y(42);
- log->push_back("---");
- x = y;
- log->push_back("---");
- }
- EXPECT_EQ(
- V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
- "0:17. destructor", "2:42. explicit constructor", "---",
- "1:42. operator= copy (from 2:42)", "---", "2:42. destructor",
- "1:42. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestMoveAssignToEmptyFromEmpty) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x, y;
- x = static_cast<Maybe<Logger>&&>(y);
- }
- EXPECT_EQ(
- V("0:0. default constructor", "1:1. default constructor",
- "0:1. operator= move (from 1:1)", "1:1. destructor", "0:1. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestMoveAssignToFullFromEmpty) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x = Logger(17);
- Maybe<Logger> y;
- log->push_back("---");
- x = static_cast<Maybe<Logger>&&>(y);
- log->push_back("---");
- }
- EXPECT_EQ(
- V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
- "0:17. destructor", "2:2. default constructor", "---",
- "1:2. operator= move (from 2:2)", "---", "2:2. destructor",
- "1:2. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestMoveAssignToEmptyFromFull) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x;
- Maybe<Logger> y = Logger(17);
- log->push_back("---");
- x = static_cast<Maybe<Logger>&&>(y);
- log->push_back("---");
- }
- EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor",
- "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
- "0:17. operator= move (from 2:17)", "---", "2:17. destructor",
- "0:17. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestMoveAssignToFullFromFull) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x = Logger(17);
- Maybe<Logger> y = Logger(42);
- log->push_back("---");
- x = static_cast<Maybe<Logger>&&>(y);
- log->push_back("---");
- }
- EXPECT_EQ(
- V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
- "0:17. destructor", "2:42. explicit constructor",
- "3:42. move constructor (from 2:42)", "2:42. destructor", "---",
- "1:42. operator= move (from 3:42)", "---", "3:42. destructor",
- "1:42. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestMoveAssignToEmptyFromT) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x;
- Logger y(17);
- log->push_back("---");
- x = static_cast<Logger&&>(y);
- log->push_back("---");
- }
- EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor", "---",
- "0:17. operator= move (from 1:17)", "---", "1:17. destructor",
- "0:17. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestMoveAssignToFullFromT) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x = Logger(17);
- Logger y(42);
- log->push_back("---");
- x = static_cast<Logger&&>(y);
- log->push_back("---");
- }
- EXPECT_EQ(
- V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
- "0:17. destructor", "2:42. explicit constructor", "---",
- "1:42. operator= move (from 2:42)", "---", "2:42. destructor",
- "1:42. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestDereference) {
- auto log = Logger::Setup();
- {
- Maybe<Logger> x = Logger(42);
- const auto& y = x;
- log->push_back("---");
- x->Foo();
- y->Foo();
- static_cast<Maybe<Logger>&&>(x)->Foo();
- static_cast<const Maybe<Logger>&&>(y)->Foo();
- log->push_back("---");
- (*x).Foo();
- (*y).Foo();
- (*static_cast<Maybe<Logger>&&>(x)).Foo();
- (*static_cast<const Maybe<Logger>&&>(y)).Foo();
- log->push_back("---");
- }
- EXPECT_EQ(V("0:42. explicit constructor",
- "1:42. move constructor (from 0:42)", "0:42. destructor", "---",
- "1:42. Foo()", "1:42. Foo() const", "1:42. Foo()",
- "1:42. Foo() const", "---", "1:42. Foo()", "1:42. Foo() const",
- "1:42. Foo()", "1:42. Foo() const", "---", "1:42. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestDereferenceWithDefault) {
- auto log = Logger::Setup();
- {
- const Logger a(17), b(42);
- Maybe<Logger> x(a);
- Maybe<Logger> y;
- log->push_back("-1-");
- EXPECT_EQ(a, x.value_or(Logger(42)));
- log->push_back("-2-");
- EXPECT_EQ(b, y.value_or(Logger(42)));
- log->push_back("-3-");
- EXPECT_EQ(a, Maybe<Logger>(Logger(17)).value_or(b));
- log->push_back("-4-");
- EXPECT_EQ(b, Maybe<Logger>().value_or(b));
- log->push_back("-5-");
- }
- EXPECT_EQ(
- V("0:17. explicit constructor", "1:42. explicit constructor",
- "2:17. copy constructor (from 0:17)", "3:3. default constructor", "-1-",
- "4:42. explicit constructor", "operator== 0:17, 2:17",
- "4:42. destructor", "-2-", "5:42. explicit constructor",
- "operator== 1:42, 5:42", "5:42. destructor", "-3-",
- "6:17. explicit constructor", "7:17. move constructor (from 6:17)",
- "operator== 0:17, 7:17", "7:17. destructor", "6:17. destructor", "-4-",
- "8:8. default constructor", "operator== 1:42, 1:42", "8:8. destructor",
- "-5-", "3:3. destructor", "2:17. destructor", "1:42. destructor",
- "0:17. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestEquality) {
- auto log = Logger::Setup();
- {
- Logger a(17), b(42);
- Maybe<Logger> ma1(a), ma2(a), mb(b), me1, me2;
- log->push_back("---");
- EXPECT_EQ(ma1, ma1);
- EXPECT_EQ(ma1, ma2);
- EXPECT_NE(ma1, mb);
- EXPECT_NE(ma1, me1);
- EXPECT_EQ(me1, me1);
- EXPECT_EQ(me1, me2);
- log->push_back("---");
- }
- EXPECT_EQ(V("0:17. explicit constructor", "1:42. explicit constructor",
- "2:17. copy constructor (from 0:17)",
- "3:17. copy constructor (from 0:17)",
- "4:42. copy constructor (from 1:42)", "5:5. default constructor",
- "6:6. default constructor", "---", "operator== 2:17, 2:17",
- "operator== 2:17, 3:17", "operator!= 2:17, 4:42", "---",
- "6:6. destructor", "5:5. destructor", "4:42. destructor",
- "3:17. destructor", "2:17. destructor", "1:42. destructor",
- "0:17. destructor"),
- *log);
-}
-
-TEST(MaybeTest, TestSwap) {
- auto log = Logger::Setup();
- {
- Logger a(17), b(42);
- Maybe<Logger> x1(a), x2(b), y1(a), y2, z1, z2;
- log->push_back("---");
- swap(x1, x2); // Swap full <-> full.
- swap(y1, y2); // Swap full <-> empty.
- swap(z1, z2); // Swap empty <-> empty.
- log->push_back("---");
- }
- EXPECT_EQ(V("0:17. explicit constructor", "1:42. explicit constructor",
- "2:17. copy constructor (from 0:17)",
- "3:42. copy constructor (from 1:42)",
- "4:17. copy constructor (from 0:17)", "5:5. default constructor",
- "6:6. default constructor", "7:7. default constructor", "---",
- "swap 2:42, 3:17", "swap 4:5, 5:17", "swap 6:7, 7:6", "---",
- "7:6. destructor", "6:7. destructor", "5:17. destructor",
- "4:5. destructor", "3:17. destructor", "2:42. destructor",
- "1:42. destructor", "0:17. destructor"),
- *log);
-}
-
-} // namespace rtc
diff --git a/webrtc/base/messagehandler.h b/webrtc/base/messagehandler.h
index df82b4e..b55b229 100644
--- a/webrtc/base/messagehandler.h
+++ b/webrtc/base/messagehandler.h
@@ -11,6 +11,8 @@
#ifndef WEBRTC_BASE_MESSAGEHANDLER_H_
#define WEBRTC_BASE_MESSAGEHANDLER_H_
+#include <utility>
+
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
@@ -54,8 +56,8 @@
: public MessageHandler {
public:
explicit FunctorMessageHandler(const FunctorT& functor) : functor_(functor) {}
- virtual void OnMessage(Message* msg) { result_ = functor_().Pass(); }
- rtc::scoped_ptr<ReturnT> result() { return result_.Pass(); }
+ virtual void OnMessage(Message* msg) { result_ = std::move(functor_()); }
+ rtc::scoped_ptr<ReturnT> result() { return std::move(result_); }
private:
FunctorT functor_;
diff --git a/webrtc/base/messagequeue_unittest.cc b/webrtc/base/messagequeue_unittest.cc
index 871542d..78024e0 100644
--- a/webrtc/base/messagequeue_unittest.cc
+++ b/webrtc/base/messagequeue_unittest.cc
@@ -16,7 +16,6 @@
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/base/nullsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using namespace rtc;
diff --git a/webrtc/base/nat_unittest.cc b/webrtc/base/nat_unittest.cc
index e967b29..8be1be9 100644
--- a/webrtc/base/nat_unittest.cc
+++ b/webrtc/base/nat_unittest.cc
@@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <algorithm>
#include <string>
#include "webrtc/base/gunit.h"
@@ -20,7 +21,6 @@
#include "webrtc/base/testclient.h"
#include "webrtc/base/asynctcpsocket.h"
#include "webrtc/base/virtualsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using namespace rtc;
@@ -207,6 +207,12 @@
std::vector<Network*> networks;
network_manager.GetNetworks(&networks);
+ networks.erase(std::remove_if(networks.begin(), networks.end(),
+ [](rtc::Network* network) {
+ return rtc::kDefaultNetworkIgnoreMask &
+ network->type();
+ }),
+ networks.end());
if (networks.empty()) {
LOG(LS_WARNING) << "Not enough network adapters for test.";
return;
diff --git a/webrtc/base/natsocketfactory.cc b/webrtc/base/natsocketfactory.cc
index 548a80c..0abd2a1 100644
--- a/webrtc/base/natsocketfactory.cc
+++ b/webrtc/base/natsocketfactory.cc
@@ -10,6 +10,7 @@
#include "webrtc/base/natsocketfactory.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/natserver.h"
#include "webrtc/base/virtualsocketserver.h"
@@ -270,7 +271,7 @@
// Sends the destination address to the server to tell it to connect.
void SendConnectRequest() {
char buf[kNATEncodedIPv6AddressSize];
- size_t length = PackAddressForNAT(buf, ARRAY_SIZE(buf), remote_addr_);
+ size_t length = PackAddressForNAT(buf, arraysize(buf), remote_addr_);
socket_->Send(buf, length);
}
diff --git a/webrtc/base/network.cc b/webrtc/base/network.cc
index 879c1e4..488c475 100644
--- a/webrtc/base/network.cc
+++ b/webrtc/base/network.cc
@@ -24,23 +24,13 @@
#elif !defined(__native_client__)
#include <net/if.h>
#endif
-#include <sys/socket.h>
-#include <sys/utsname.h>
-#include <sys/ioctl.h>
-#include <unistd.h>
-#include <errno.h>
-
-#if defined(WEBRTC_ANDROID)
-#include "webrtc/base/ifaddrs-android.h"
-#elif !defined(__native_client__)
-#include <ifaddrs.h>
-#endif
-
#endif // WEBRTC_POSIX
#if defined(WEBRTC_WIN)
#include "webrtc/base/win32.h"
#include <Iphlpapi.h>
+#elif !defined(__native_client__)
+#include "webrtc/base/ifaddrs_converter.h"
#endif
#include <stdio.h>
@@ -129,7 +119,8 @@
}
}
-bool IsIgnoredIPv6(const IPAddress& ip) {
+#if !defined(__native_client__)
+bool IsIgnoredIPv6(const InterfaceAddress& ip) {
if (ip.family() != AF_INET6) {
return false;
}
@@ -146,11 +137,23 @@
return true;
}
+ // Ignore deprecated IPv6.
+ if (ip.ipv6_flags() & IPV6_ADDRESS_FLAG_DEPRECATED) {
+ return true;
+ }
+
return false;
}
+#endif // !defined(__native_client__)
} // namespace
+// These addresses are used as the targets to find out the default local address
+// on a multi-homed endpoint. They are actually DNS servers.
+const char kPublicIPv4Host[] = "8.8.8.8";
+const char kPublicIPv6Host[] = "2001:4860:4860::8888";
+const int kPublicPort = 53; // DNS port.
+
std::string MakeNetworkKey(const std::string& name, const IPAddress& prefix,
int prefix_length) {
std::ostringstream ost;
@@ -169,6 +172,10 @@
return ENUMERATION_ALLOWED;
}
+bool NetworkManager::GetDefaultLocalAddress(int family, IPAddress* addr) const {
+ return false;
+}
+
NetworkManagerBase::NetworkManagerBase()
: enumeration_permission_(NetworkManager::ENUMERATION_ALLOWED),
max_ipv6_networks_(kMaxIPv6Networks),
@@ -191,6 +198,7 @@
const rtc::IPAddress ipv4_any_address(INADDR_ANY);
ipv4_any_address_network_.reset(
new rtc::Network("any", "any", ipv4_any_address, 0));
+ ipv4_any_address_network_->set_default_local_address_provider(this);
ipv4_any_address_network_->AddIP(ipv4_any_address);
}
networks->push_back(ipv4_any_address_network_.get());
@@ -200,6 +208,7 @@
const rtc::IPAddress ipv6_any_address(in6addr_any);
ipv6_any_address_network_.reset(
new rtc::Network("any", "any", ipv6_any_address, 0));
+ ipv6_any_address_network_->set_default_local_address_provider(this);
ipv6_any_address_network_->AddIP(ipv6_any_address);
}
networks->push_back(ipv6_any_address_network_.get());
@@ -230,20 +239,12 @@
void NetworkManagerBase::MergeNetworkList(const NetworkList& new_networks,
bool* changed,
NetworkManager::Stats* stats) {
+ *changed = false;
// AddressList in this map will track IP addresses for all Networks
// with the same key.
std::map<std::string, AddressList> consolidated_address_list;
NetworkList list(new_networks);
-
- // Result of Network merge. Element in this list should have unique key.
- NetworkList merged_list;
std::sort(list.begin(), list.end(), CompareNetworks);
-
- *changed = false;
-
- if (networks_.size() != list.size())
- *changed = true;
-
// First, build a set of network-keys to the ipaddresses.
for (Network* network : list) {
bool might_add_to_merged_list = false;
@@ -275,6 +276,8 @@
}
// Next, look for existing network objects to re-use.
+ // Result of Network merge. Element in this list should have unique key.
+ NetworkList merged_list;
for (const auto& kv : consolidated_address_list) {
const std::string& key = kv.first;
Network* net = kv.second.net;
@@ -289,17 +292,36 @@
*changed = true;
} else {
// This network exists in the map already. Reset its IP addresses.
- *changed = existing->second->SetIPs(kv.second.ips, *changed);
- merged_list.push_back(existing->second);
- if (existing->second != net) {
+ Network* existing_net = existing->second;
+ *changed = existing_net->SetIPs(kv.second.ips, *changed);
+ merged_list.push_back(existing_net);
+ // If the existing network was not active, networks have changed.
+ if (!existing_net->active()) {
+ *changed = true;
+ }
+ ASSERT(net->active());
+ if (existing_net != net) {
delete net;
}
}
}
- networks_ = merged_list;
+ // It may still happen that the merged list is a subset of |networks_|.
+ // To detect this change, we compare their sizes.
+ if (merged_list.size() != networks_.size()) {
+ *changed = true;
+ }
- // If the network lists changes, we resort it.
+ // If the network list changes, we re-assign |networks_| to the merged list
+ // and re-sort it.
if (*changed) {
+ networks_ = merged_list;
+ // Reset the active states of all networks.
+ for (const auto& kv : networks_map_) {
+ kv.second->set_active(false);
+ }
+ for (Network* network : networks_) {
+ network->set_active(true);
+ }
std::sort(networks_.begin(), networks_.end(), SortNetworks);
// Now network interfaces are sorted, we should set the preference value
// for each of the interfaces we are planning to use.
@@ -321,9 +343,30 @@
}
}
+void NetworkManagerBase::set_default_local_addresses(const IPAddress& ipv4,
+ const IPAddress& ipv6) {
+ if (ipv4.family() == AF_INET) {
+ default_local_ipv4_address_ = ipv4;
+ }
+ if (ipv6.family() == AF_INET6) {
+ default_local_ipv6_address_ = ipv6;
+ }
+}
+
+bool NetworkManagerBase::GetDefaultLocalAddress(int family,
+ IPAddress* ipaddr) const {
+ if (family == AF_INET && !default_local_ipv4_address_.IsNil()) {
+ *ipaddr = default_local_ipv4_address_;
+ return true;
+ } else if (family == AF_INET6 && !default_local_ipv6_address_.IsNil()) {
+ *ipaddr = default_local_ipv6_address_;
+ return true;
+ }
+ return false;
+}
+
BasicNetworkManager::BasicNetworkManager()
: thread_(NULL), sent_first_update_(false), start_count_(0),
- network_ignore_mask_(kDefaultNetworkIgnoreMask),
ignore_non_default_routes_(false) {
}
@@ -346,49 +389,47 @@
#elif defined(WEBRTC_POSIX)
void BasicNetworkManager::ConvertIfAddrs(struct ifaddrs* interfaces,
+ IfAddrsConverter* ifaddrs_converter,
bool include_ignored,
NetworkList* networks) const {
NetworkMap current_networks;
+
for (struct ifaddrs* cursor = interfaces;
cursor != NULL; cursor = cursor->ifa_next) {
IPAddress prefix;
IPAddress mask;
- IPAddress ip;
+ InterfaceAddress ip;
int scope_id = 0;
// Some interfaces may not have address assigned.
- if (!cursor->ifa_addr || !cursor->ifa_netmask)
+ if (!cursor->ifa_addr || !cursor->ifa_netmask) {
continue;
+ }
+ // Skip ones which are down.
+ if (!(cursor->ifa_flags & IFF_RUNNING)) {
+ continue;
+ }
+ // Skip unknown family.
+ if (cursor->ifa_addr->sa_family != AF_INET &&
+ cursor->ifa_addr->sa_family != AF_INET6) {
+ continue;
+ }
+ // Skip IPv6 if not enabled.
+ if (cursor->ifa_addr->sa_family == AF_INET6 && !ipv6_enabled()) {
+ continue;
+ }
+ // Convert to InterfaceAddress.
+ if (!ifaddrs_converter->ConvertIfAddrsToIPAddress(cursor, &ip, &mask)) {
+ continue;
+ }
- switch (cursor->ifa_addr->sa_family) {
- case AF_INET: {
- ip = IPAddress(
- reinterpret_cast<sockaddr_in*>(cursor->ifa_addr)->sin_addr);
- mask = IPAddress(
- reinterpret_cast<sockaddr_in*>(cursor->ifa_netmask)->sin_addr);
- break;
- }
- case AF_INET6: {
- if (ipv6_enabled()) {
- ip = IPAddress(
- reinterpret_cast<sockaddr_in6*>(cursor->ifa_addr)->sin6_addr);
-
- if (IsIgnoredIPv6(ip)) {
- continue;
- }
-
- mask = IPAddress(
- reinterpret_cast<sockaddr_in6*>(cursor->ifa_netmask)->sin6_addr);
- scope_id =
- reinterpret_cast<sockaddr_in6*>(cursor->ifa_addr)->sin6_scope_id;
- break;
- } else {
- continue;
- }
- }
- default: {
+ // Special case for IPv6 address.
+ if (cursor->ifa_addr->sa_family == AF_INET6) {
+ if (IsIgnoredIPv6(ip)) {
continue;
}
+ scope_id =
+ reinterpret_cast<sockaddr_in6*>(cursor->ifa_addr)->sin6_scope_id;
}
int prefix_length = CountIPMaskBits(mask);
@@ -409,14 +450,14 @@
#endif
// TODO(phoglund): Need to recognize other types as well.
scoped_ptr<Network> network(new Network(cursor->ifa_name,
- cursor->ifa_name,
- prefix,
- prefix_length,
- adapter_type));
+ cursor->ifa_name, prefix,
+ prefix_length, adapter_type));
+ network->set_default_local_address_provider(this);
network->set_scope_id(scope_id);
network->AddIP(ip);
network->set_ignored(IsIgnoredNetwork(*network));
if (include_ignored || !network->ignored()) {
+ current_networks[key] = network.get();
networks->push_back(network.release());
}
} else {
@@ -434,7 +475,9 @@
return false;
}
- ConvertIfAddrs(interfaces, include_ignored, networks);
+ rtc::scoped_ptr<IfAddrsConverter> ifaddrs_converter(CreateIfAddrsConverter());
+ ConvertIfAddrs(interfaces, ifaddrs_converter.get(), include_ignored,
+ networks);
freeifaddrs(interfaces);
return true;
@@ -511,14 +554,14 @@
PIP_ADAPTER_PREFIX prefixlist = adapter_addrs->FirstPrefix;
std::string name;
std::string description;
-#ifdef _DEBUG
+#if !defined(NDEBUG)
name = ToUtf8(adapter_addrs->FriendlyName,
wcslen(adapter_addrs->FriendlyName));
#endif
description = ToUtf8(adapter_addrs->Description,
wcslen(adapter_addrs->Description));
for (; address; address = address->Next) {
-#ifndef _DEBUG
+#if defined(NDEBUG)
name = rtc::ToString(count);
#endif
@@ -563,16 +606,15 @@
// TODO(phoglund): Need to recognize other types as well.
adapter_type = ADAPTER_TYPE_LOOPBACK;
}
- scoped_ptr<Network> network(new Network(name,
- description,
- prefix,
- prefix_length,
- adapter_type));
+ scoped_ptr<Network> network(new Network(name, description, prefix,
+ prefix_length, adapter_type));
+ network->set_default_local_address_provider(this);
network->set_scope_id(scope_id);
network->AddIP(ip);
bool ignored = IsIgnoredNetwork(*network);
network->set_ignored(ignored);
if (include_ignored || !network->ignored()) {
+ current_networks[key] = network.get();
networks->push_back(network.release());
}
} else {
@@ -624,9 +666,6 @@
}
}
- if (network_ignore_mask_ & network.type()) {
- return true;
- }
#if defined(WEBRTC_POSIX)
// Filter out VMware/VirtualBox interfaces, typically named vmnet1,
// vmnet8, or vboxnet0.
@@ -724,6 +763,25 @@
}
}
+IPAddress BasicNetworkManager::QueryDefaultLocalAddress(int family) const {
+ ASSERT(thread_ == Thread::Current());
+ ASSERT(thread_->socketserver() != nullptr);
+ ASSERT(family == AF_INET || family == AF_INET6);
+
+ scoped_ptr<AsyncSocket> socket(
+ thread_->socketserver()->CreateAsyncSocket(family, SOCK_DGRAM));
+ if (!socket) {
+ return IPAddress();
+ }
+
+ if (!socket->Connect(
+ SocketAddress(family == AF_INET ? kPublicIPv4Host : kPublicIPv6Host,
+ kPublicPort))) {
+ return IPAddress();
+ }
+ return socket->GetLocalAddress().ipaddr();
+}
+
void BasicNetworkManager::UpdateNetworksOnce() {
if (!start_count_)
return;
@@ -735,7 +793,10 @@
SignalError();
} else {
bool changed;
- MergeNetworkList(list, &changed);
+ NetworkManager::Stats stats;
+ MergeNetworkList(list, &changed, &stats);
+ set_default_local_addresses(QueryDefaultLocalAddress(AF_INET),
+ QueryDefaultLocalAddress(AF_INET6));
if (changed || !sent_first_update_) {
SignalNetworksChanged();
sent_first_update_ = true;
@@ -748,39 +809,45 @@
thread_->PostDelayed(kNetworksUpdateIntervalMs, this, kUpdateNetworksMessage);
}
-void BasicNetworkManager::DumpNetworks(bool include_ignored) {
+void BasicNetworkManager::DumpNetworks() {
NetworkList list;
- CreateNetworks(include_ignored, &list);
+ GetNetworks(&list);
LOG(LS_INFO) << "NetworkManager detected " << list.size() << " networks:";
for (const Network* network : list) {
- if (!network->ignored() || include_ignored) {
- LOG(LS_INFO) << network->ToString() << ": "
- << network->description()
- << ((network->ignored()) ? ", Ignored" : "");
- }
- }
- // Release the network list created previously.
- // Do this in a seperated for loop for better readability.
- for (Network* network : list) {
- delete network;
+ LOG(LS_INFO) << network->ToString() << ": " << network->description()
+ << ", active ? " << network->active()
+ << ((network->ignored()) ? ", Ignored" : "");
}
}
-Network::Network(const std::string& name, const std::string& desc,
- const IPAddress& prefix, int prefix_length)
- : name_(name), description_(desc), prefix_(prefix),
+Network::Network(const std::string& name,
+ const std::string& desc,
+ const IPAddress& prefix,
+ int prefix_length)
+ : name_(name),
+ description_(desc),
+ prefix_(prefix),
prefix_length_(prefix_length),
- key_(MakeNetworkKey(name, prefix, prefix_length)), scope_id_(0),
- ignored_(false), type_(ADAPTER_TYPE_UNKNOWN), preference_(0) {
-}
+ key_(MakeNetworkKey(name, prefix, prefix_length)),
+ scope_id_(0),
+ ignored_(false),
+ type_(ADAPTER_TYPE_UNKNOWN),
+ preference_(0) {}
-Network::Network(const std::string& name, const std::string& desc,
- const IPAddress& prefix, int prefix_length, AdapterType type)
- : name_(name), description_(desc), prefix_(prefix),
+Network::Network(const std::string& name,
+ const std::string& desc,
+ const IPAddress& prefix,
+ int prefix_length,
+ AdapterType type)
+ : name_(name),
+ description_(desc),
+ prefix_(prefix),
prefix_length_(prefix_length),
- key_(MakeNetworkKey(name, prefix, prefix_length)), scope_id_(0),
- ignored_(false), type_(type), preference_(0) {
-}
+ key_(MakeNetworkKey(name, prefix, prefix_length)),
+ scope_id_(0),
+ ignored_(false),
+ type_(type),
+ preference_(0) {}
Network::~Network() = default;
diff --git a/webrtc/base/network.h b/webrtc/base/network.h
index ab3a88d..2f2e1b3 100644
--- a/webrtc/base/network.h
+++ b/webrtc/base/network.h
@@ -28,6 +28,10 @@
namespace rtc {
+extern const char kPublicIPv4Host[];
+extern const char kPublicIPv6Host[];
+
+class IfAddrsConverter;
class Network;
class NetworkMonitorInterface;
class Thread;
@@ -51,9 +55,18 @@
std::string MakeNetworkKey(const std::string& name, const IPAddress& prefix,
int prefix_length);
+class DefaultLocalAddressProvider {
+ public:
+ virtual ~DefaultLocalAddressProvider() = default;
+ // The default local address is the local address used in multi-homed endpoint
+ // when the any address (0.0.0.0 or ::) is used as the local address. It's
+ // important to check the return value as a IP family may not be enabled.
+ virtual bool GetDefaultLocalAddress(int family, IPAddress* ipaddr) const = 0;
+};
+
// Generic network manager interface. It provides list of local
// networks.
-class NetworkManager {
+class NetworkManager : public DefaultLocalAddressProvider {
public:
typedef std::vector<Network*> NetworkList;
@@ -67,7 +80,7 @@
};
NetworkManager();
- virtual ~NetworkManager();
+ ~NetworkManager() override;
// Called when network list is updated.
sigslot::signal0<> SignalNetworksChanged;
@@ -99,8 +112,9 @@
// TODO(guoweis): remove this body when chromium implements this.
virtual void GetAnyAddressNetworks(NetworkList* networks) {}
- // Dumps a list of networks available to LS_INFO.
- virtual void DumpNetworks(bool include_ignored) {}
+ // Dumps the current list of networks in the network manager.
+ virtual void DumpNetworks() {}
+ bool GetDefaultLocalAddress(int family, IPAddress* ipaddr) const override;
struct Stats {
int ipv4_network_count;
@@ -128,6 +142,8 @@
EnumerationPermission enumeration_permission() const override;
+ bool GetDefaultLocalAddress(int family, IPAddress* ipaddr) const override;
+
protected:
typedef std::map<std::string, Network*> NetworkMap;
// Updates |networks_| with the networks listed in |list|. If
@@ -146,6 +162,9 @@
enumeration_permission_ = state;
}
+ void set_default_local_addresses(const IPAddress& ipv4,
+ const IPAddress& ipv6);
+
private:
friend class NetworkTest;
@@ -159,6 +178,9 @@
rtc::scoped_ptr<rtc::Network> ipv4_any_address_network_;
rtc::scoped_ptr<rtc::Network> ipv6_any_address_network_;
+
+ IPAddress default_local_ipv4_address_;
+ IPAddress default_local_ipv6_address_;
};
// Basic implementation of the NetworkManager interface that gets list
@@ -173,8 +195,7 @@
void StartUpdating() override;
void StopUpdating() override;
- // Logs the available networks.
- void DumpNetworks(bool include_ignored) override;
+ void DumpNetworks() override;
// MessageHandler interface.
void OnMessage(Message* msg) override;
@@ -186,18 +207,6 @@
network_ignore_list_ = list;
}
- // Sets the network types to ignore. For instance, calling this with
- // ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
- // loopback interfaces. Set to kDefaultNetworkIgnoreMask by default.
- void set_network_ignore_mask(int network_ignore_mask) {
- // TODO(phoglund): implement support for other types than loopback.
- // See https://code.google.com/p/webrtc/issues/detail?id=4288.
- // Then remove set_network_ignore_list.
- network_ignore_mask_ = network_ignore_mask;
- }
-
- int network_ignore_mask() const { return network_ignore_mask_; }
-
#if defined(WEBRTC_LINUX)
// Sets the flag for ignoring non-default routes.
void set_ignore_non_default_routes(bool value) {
@@ -209,6 +218,7 @@
#if defined(WEBRTC_POSIX)
// Separated from CreateNetworks for tests.
void ConvertIfAddrs(ifaddrs* interfaces,
+ IfAddrsConverter* converter,
bool include_ignored,
NetworkList* networks) const;
#endif // defined(WEBRTC_POSIX)
@@ -220,6 +230,11 @@
// based on the network's property instead of any individual IP.
bool IsIgnoredNetwork(const Network& network) const;
+ // This function connects a UDP socket to a public address and returns the
+ // local address associated it. Since it binds to the "any" address
+ // internally, it returns the default local address on a multi-homed endpoint.
+ IPAddress QueryDefaultLocalAddress(int family) const;
+
private:
friend class NetworkTest;
@@ -239,7 +254,6 @@
bool sent_first_update_;
int start_count_;
std::vector<std::string> network_ignore_list_;
- int network_ignore_mask_;
bool ignore_non_default_routes_;
scoped_ptr<NetworkMonitorInterface> network_monitor_;
};
@@ -247,13 +261,26 @@
// Represents a Unix-type network interface, with a name and single address.
class Network {
public:
- Network(const std::string& name, const std::string& description,
- const IPAddress& prefix, int prefix_length);
+ Network(const std::string& name,
+ const std::string& description,
+ const IPAddress& prefix,
+ int prefix_length);
- Network(const std::string& name, const std::string& description,
- const IPAddress& prefix, int prefix_length, AdapterType type);
+ Network(const std::string& name,
+ const std::string& description,
+ const IPAddress& prefix,
+ int prefix_length,
+ AdapterType type);
~Network();
+ const DefaultLocalAddressProvider* default_local_address_provider() {
+ return default_local_address_provider_;
+ }
+ void set_default_local_address_provider(
+ const DefaultLocalAddressProvider* provider) {
+ default_local_address_provider_ = provider;
+ }
+
// Returns the name of the interface this network is associated wtih.
const std::string& name() const { return name_; }
@@ -319,10 +346,17 @@
int preference() const { return preference_; }
void set_preference(int preference) { preference_ = preference; }
+ // When we enumerate networks and find a previously-seen network is missing,
+ // we do not remove it (because it may be used elsewhere). Instead, we mark
+ // it inactive, so that we can detect network changes properly.
+ bool active() const { return active_; }
+ void set_active(bool active) { active_ = active; }
+
// Debugging description of this network
std::string ToString() const;
private:
+ const DefaultLocalAddressProvider* default_local_address_provider_ = nullptr;
std::string name_;
std::string description_;
IPAddress prefix_;
@@ -333,6 +367,7 @@
bool ignored_;
AdapterType type_;
int preference_;
+ bool active_ = true;
friend class NetworkManager;
};
diff --git a/webrtc/base/network_unittest.cc b/webrtc/base/network_unittest.cc
index 4362221..7dd400b 100644
--- a/webrtc/base/network_unittest.cc
+++ b/webrtc/base/network_unittest.cc
@@ -10,16 +10,14 @@
#include "webrtc/base/network.h"
+#include "webrtc/base/nethelpers.h"
#include "webrtc/base/networkmonitor.h"
#include <vector>
#if defined(WEBRTC_POSIX)
#include <sys/types.h>
-#if !defined(WEBRTC_ANDROID)
-#include <ifaddrs.h>
-#else
-#include "webrtc/base/ifaddrs-android.h"
-#endif
-#endif
+#include <net/if.h>
+#include "webrtc/base/ifaddrs_converter.h"
+#endif // defined(WEBRTC_POSIX)
#include "webrtc/base/gunit.h"
#if defined(WEBRTC_WIN)
#include "webrtc/base/logging.h" // For LOG_GLE
@@ -27,6 +25,8 @@
namespace rtc {
+namespace {
+
class FakeNetworkMonitor : public NetworkMonitorBase {
public:
void Start() override {}
@@ -41,6 +41,8 @@
}
};
+} // namespace
+
class NetworkTest : public testing::Test, public sigslot::has_slots<> {
public:
NetworkTest() : callback_called_(false) {}
@@ -88,7 +90,51 @@
struct ifaddrs* interfaces,
bool include_ignored,
NetworkManager::NetworkList* networks) {
- network_manager.ConvertIfAddrs(interfaces, include_ignored, networks);
+ // Use the base IfAddrsConverter for test cases.
+ rtc::scoped_ptr<IfAddrsConverter> ifaddrs_converter(new IfAddrsConverter());
+ network_manager.ConvertIfAddrs(interfaces, ifaddrs_converter.get(),
+ include_ignored, networks);
+ }
+
+ struct sockaddr_in6* CreateIpv6Addr(const std::string& ip_string,
+ uint32_t scope_id) {
+ struct sockaddr_in6* ipv6_addr = new struct sockaddr_in6;
+ memset(ipv6_addr, 0, sizeof(struct sockaddr_in6));
+ ipv6_addr->sin6_family = AF_INET6;
+ ipv6_addr->sin6_scope_id = scope_id;
+ IPAddress ip;
+ IPFromString(ip_string, &ip);
+ ipv6_addr->sin6_addr = ip.ipv6_address();
+ return ipv6_addr;
+ }
+
+ // Pointers created here need to be released via ReleaseIfAddrs.
+ struct ifaddrs* AddIpv6Address(struct ifaddrs* list,
+ char* if_name,
+ const std::string& ipv6_address,
+ const std::string& ipv6_netmask,
+ uint32_t scope_id) {
+ struct ifaddrs* if_addr = new struct ifaddrs;
+ memset(if_addr, 0, sizeof(struct ifaddrs));
+ if_addr->ifa_name = if_name;
+ if_addr->ifa_addr = reinterpret_cast<struct sockaddr*>(
+ CreateIpv6Addr(ipv6_address, scope_id));
+ if_addr->ifa_netmask =
+ reinterpret_cast<struct sockaddr*>(CreateIpv6Addr(ipv6_netmask, 0));
+ if_addr->ifa_next = list;
+ if_addr->ifa_flags = IFF_RUNNING;
+ return if_addr;
+ }
+
+ void ReleaseIfAddrs(struct ifaddrs* list) {
+ struct ifaddrs* if_addr = list;
+ while (if_addr != nullptr) {
+ struct ifaddrs* next_addr = if_addr->ifa_next;
+ delete if_addr->ifa_addr;
+ delete if_addr->ifa_netmask;
+ delete if_addr;
+ if_addr = next_addr;
+ }
}
#endif // defined(WEBRTC_POSIX)
@@ -96,6 +142,12 @@
bool callback_called_;
};
+class TestBasicNetworkManager : public BasicNetworkManager {
+ public:
+ using BasicNetworkManager::QueryDefaultLocalAddress;
+ using BasicNetworkManager::set_default_local_addresses;
+};
+
// Test that the Network ctor works properly.
TEST_F(NetworkTest, TestNetworkConstruct) {
Network ipv4_network1("test_eth0", "Test Network Adapter 1",
@@ -107,26 +159,6 @@
EXPECT_FALSE(ipv4_network1.ignored());
}
-// Tests that our ignore function works properly.
-TEST_F(NetworkTest, TestIsIgnoredNetworkIgnoresOnlyLoopbackByDefault) {
- Network ipv4_network1("test_eth0", "Test Network Adapter 1",
- IPAddress(0x12345600U), 24, ADAPTER_TYPE_ETHERNET);
- Network ipv4_network2("test_wlan0", "Test Network Adapter 2",
- IPAddress(0x12345601U), 16, ADAPTER_TYPE_WIFI);
- Network ipv4_network3("test_cell0", "Test Network Adapter 3",
- IPAddress(0x12345602U), 16, ADAPTER_TYPE_CELLULAR);
- Network ipv4_network4("test_vpn0", "Test Network Adapter 4",
- IPAddress(0x12345603U), 16, ADAPTER_TYPE_VPN);
- Network ipv4_network5("test_lo", "Test Network Adapter 5",
- IPAddress(0x12345604U), 16, ADAPTER_TYPE_LOOPBACK);
- BasicNetworkManager network_manager;
- EXPECT_FALSE(IsIgnoredNetwork(network_manager, ipv4_network1));
- EXPECT_FALSE(IsIgnoredNetwork(network_manager, ipv4_network2));
- EXPECT_FALSE(IsIgnoredNetwork(network_manager, ipv4_network3));
- EXPECT_FALSE(IsIgnoredNetwork(network_manager, ipv4_network4));
- EXPECT_TRUE(IsIgnoredNetwork(network_manager, ipv4_network5));
-}
-
TEST_F(NetworkTest, TestIsIgnoredNetworkIgnoresIPsStartingWith0) {
Network ipv4_network1("test_eth0", "Test Network Adapter 1",
IPAddress(0x12345600U), 24, ADAPTER_TYPE_ETHERNET);
@@ -137,21 +169,6 @@
EXPECT_TRUE(IsIgnoredNetwork(network_manager, ipv4_network2));
}
-TEST_F(NetworkTest, TestIsIgnoredNetworkIgnoresNetworksAccordingToIgnoreMask) {
- Network ipv4_network1("test_eth0", "Test Network Adapter 1",
- IPAddress(0x12345600U), 24, ADAPTER_TYPE_ETHERNET);
- Network ipv4_network2("test_wlan0", "Test Network Adapter 2",
- IPAddress(0x12345601U), 16, ADAPTER_TYPE_WIFI);
- Network ipv4_network3("test_cell0", "Test Network Adapter 3",
- IPAddress(0x12345602U), 16, ADAPTER_TYPE_CELLULAR);
- BasicNetworkManager network_manager;
- network_manager.set_network_ignore_mask(
- ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK | ADAPTER_TYPE_WIFI);
- EXPECT_TRUE(IsIgnoredNetwork(network_manager, ipv4_network1));
- EXPECT_TRUE(IsIgnoredNetwork(network_manager, ipv4_network2));
- EXPECT_FALSE(IsIgnoredNetwork(network_manager, ipv4_network3));
-}
-
// TODO(phoglund): Remove when ignore list goes away.
TEST_F(NetworkTest, TestIgnoreList) {
Network ignore_me("ignore_me", "Ignore me please!",
@@ -583,10 +600,13 @@
}
}
-// Test that DumpNetworks works.
-TEST_F(NetworkTest, TestDumpNetworks) {
+// Test that DumpNetworks does not crash.
+TEST_F(NetworkTest, TestCreateAndDumpNetworks) {
BasicNetworkManager manager;
- manager.DumpNetworks(true);
+ NetworkManager::NetworkList list = GetNetworks(manager, true);
+ bool changed;
+ MergeNetworkList(manager, list, &changed);
+ manager.DumpNetworks();
}
// Test that we can toggle IPv6 on and off.
@@ -693,6 +713,40 @@
CallConvertIfAddrs(manager, &list, true, &result);
EXPECT_TRUE(result.empty());
}
+
+// Verify that if there are two addresses on one interface, only one network
+// is generated.
+TEST_F(NetworkTest, TestConvertIfAddrsMultiAddressesOnOneInterface) {
+ char if_name[20] = "rmnet0";
+ ifaddrs* list = nullptr;
+ list = AddIpv6Address(list, if_name, "1000:2000:3000:4000:0:0:0:1",
+ "FFFF:FFFF:FFFF:FFFF::", 0);
+ list = AddIpv6Address(list, if_name, "1000:2000:3000:4000:0:0:0:2",
+ "FFFF:FFFF:FFFF:FFFF::", 0);
+ NetworkManager::NetworkList result;
+ BasicNetworkManager manager;
+ CallConvertIfAddrs(manager, list, true, &result);
+ EXPECT_EQ(1U, result.size());
+ bool changed;
+ // This ensures we release the objects created in CallConvertIfAddrs.
+ MergeNetworkList(manager, result, &changed);
+ ReleaseIfAddrs(list);
+}
+
+TEST_F(NetworkTest, TestConvertIfAddrsNotRunning) {
+ ifaddrs list;
+ memset(&list, 0, sizeof(list));
+ list.ifa_name = const_cast<char*>("test_iface");
+ sockaddr ifa_addr;
+ sockaddr ifa_netmask;
+ list.ifa_addr = &ifa_addr;
+ list.ifa_netmask = &ifa_netmask;
+
+ NetworkManager::NetworkList result;
+ BasicNetworkManager manager;
+ CallConvertIfAddrs(manager, &list, true, &result);
+ EXPECT_TRUE(result.empty());
+}
#endif // defined(WEBRTC_POSIX)
#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
@@ -776,6 +830,49 @@
EXPECT_EQ(list2[0]->GetIPs()[1], ip2);
}
+// Test that MergeNetworkList successfully detects the change if
+// a network becomes inactive and then active again.
+TEST_F(NetworkTest, TestMergeNetworkListWithInactiveNetworks) {
+ BasicNetworkManager manager;
+ Network network1("test_wifi", "Test Network Adapter 1",
+ IPAddress(0x12345600U), 24);
+ Network network2("test_eth0", "Test Network Adapter 2",
+ IPAddress(0x00010000U), 16);
+ network1.AddIP(IPAddress(0x12345678));
+ network2.AddIP(IPAddress(0x00010004));
+ NetworkManager::NetworkList list;
+ Network* net1 = new Network(network1);
+ list.push_back(net1);
+ bool changed;
+ MergeNetworkList(manager, list, &changed);
+ EXPECT_TRUE(changed);
+ list.clear();
+ manager.GetNetworks(&list);
+ ASSERT_EQ(1U, list.size());
+ EXPECT_EQ(net1, list[0]);
+
+ list.clear();
+ Network* net2 = new Network(network2);
+ list.push_back(net2);
+ MergeNetworkList(manager, list, &changed);
+ EXPECT_TRUE(changed);
+ list.clear();
+ manager.GetNetworks(&list);
+ ASSERT_EQ(1U, list.size());
+ EXPECT_EQ(net2, list[0]);
+
+ // Now network1 is inactive. Try to merge it again.
+ list.clear();
+ list.push_back(new Network(network1));
+ MergeNetworkList(manager, list, &changed);
+ EXPECT_TRUE(changed);
+ list.clear();
+ manager.GetNetworks(&list);
+ ASSERT_EQ(1U, list.size());
+ EXPECT_TRUE(list[0]->active());
+ EXPECT_EQ(net1, list[0]);
+}
+
// Test that the filtering logic follows the defined ruleset in network.h.
TEST_F(NetworkTest, TestIPv6Selection) {
InterfaceAddress ip;
@@ -842,4 +939,35 @@
NetworkMonitorFactory::ReleaseFactory(factory);
}
+TEST_F(NetworkTest, DefaultLocalAddress) {
+ TestBasicNetworkManager manager;
+ manager.StartUpdating();
+ IPAddress ip;
+
+ // GetDefaultLocalAddress should return false when not set.
+ EXPECT_FALSE(manager.GetDefaultLocalAddress(AF_INET, &ip));
+ EXPECT_FALSE(manager.GetDefaultLocalAddress(AF_INET6, &ip));
+
+ // Make sure we can query default local address when an address for such
+ // address family exists.
+ std::vector<Network*> networks;
+ manager.GetNetworks(&networks);
+ for (auto& network : networks) {
+ if (network->GetBestIP().family() == AF_INET) {
+ EXPECT_TRUE(manager.QueryDefaultLocalAddress(AF_INET) != IPAddress());
+ } else if (network->GetBestIP().family() == AF_INET6) {
+ EXPECT_TRUE(manager.QueryDefaultLocalAddress(AF_INET6) != IPAddress());
+ }
+ }
+
+ // GetDefaultLocalAddress should return the valid default address after set.
+ manager.set_default_local_addresses(GetLoopbackIP(AF_INET),
+ GetLoopbackIP(AF_INET6));
+ EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET, &ip));
+ EXPECT_EQ(ip, GetLoopbackIP(AF_INET));
+ EXPECT_TRUE(manager.GetDefaultLocalAddress(AF_INET6, &ip));
+ EXPECT_EQ(ip, GetLoopbackIP(AF_INET6));
+ manager.StopUpdating();
+}
+
} // namespace rtc
diff --git a/webrtc/base/nullsocketserver_unittest.cc b/webrtc/base/nullsocketserver_unittest.cc
index 2aa38b4..4f22c38 100644
--- a/webrtc/base/nullsocketserver_unittest.cc
+++ b/webrtc/base/nullsocketserver_unittest.cc
@@ -10,7 +10,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/nullsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/objc/NSString+StdString.h b/webrtc/base/objc/NSString+StdString.h
new file mode 100644
index 0000000..8bf6cc9
--- /dev/null
+++ b/webrtc/base/objc/NSString+StdString.h
@@ -0,0 +1,26 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include <string>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface NSString (StdString)
+
+@property(nonatomic, readonly) std::string stdString;
+
++ (std::string)stdStringForString:(NSString *)nsString;
++ (NSString *)stringForStdString:(const std::string&)stdString;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/webrtc/base/objc/NSString+StdString.mm b/webrtc/base/objc/NSString+StdString.mm
new file mode 100644
index 0000000..3210ff0
--- /dev/null
+++ b/webrtc/base/objc/NSString+StdString.mm
@@ -0,0 +1,33 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "NSString+StdString.h"
+
+@implementation NSString (StdString)
+
+- (std::string)stdString {
+ return [NSString stdStringForString:self];
+}
+
++ (std::string)stdStringForString:(NSString *)nsString {
+ NSData *charData = [nsString dataUsingEncoding:NSUTF8StringEncoding];
+ return std::string(reinterpret_cast<const char *>(charData.bytes),
+ charData.length);
+}
+
++ (NSString *)stringForStdString:(const std::string&)stdString {
+ // std::string may contain null termination character so we construct
+ // using length.
+ return [[NSString alloc] initWithBytes:stdString.data()
+ length:stdString.length()
+ encoding:NSUTF8StringEncoding];
+}
+
+@end
diff --git a/webrtc/base/objc/OWNERS b/webrtc/base/objc/OWNERS
new file mode 100644
index 0000000..cd06158
--- /dev/null
+++ b/webrtc/base/objc/OWNERS
@@ -0,0 +1 @@
+tkchin@webrtc.org
diff --git a/webrtc/base/objc/RTCCameraPreviewView.h b/webrtc/base/objc/RTCCameraPreviewView.h
new file mode 100644
index 0000000..03e94c2
--- /dev/null
+++ b/webrtc/base/objc/RTCCameraPreviewView.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+@class AVCaptureSession;
+@class RTCAVFoundationVideoSource;
+
+/** RTCCameraPreviewView is a view that renders local video from an
+ * AVCaptureSession.
+ */
+@interface RTCCameraPreviewView : UIView
+
+/** The capture session being rendered in the view. Capture session
+ * is assigned to AVCaptureVideoPreviewLayer async in the same
+ * queue that the AVCaptureSession is started/stopped.
+ */
+@property(nonatomic, strong) AVCaptureSession *captureSession;
+
+@end
diff --git a/webrtc/base/objc/RTCCameraPreviewView.m b/webrtc/base/objc/RTCCameraPreviewView.m
new file mode 100644
index 0000000..5a57483
--- /dev/null
+++ b/webrtc/base/objc/RTCCameraPreviewView.m
@@ -0,0 +1,47 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if !defined(__has_feature) || !__has_feature(objc_arc)
+#error "This file requires ARC support."
+#endif
+
+#import "webrtc/base/objc/RTCCameraPreviewView.h"
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "webrtc/base/objc/RTCDispatcher.h"
+
+@implementation RTCCameraPreviewView
+
+@synthesize captureSession = _captureSession;
+
++ (Class)layerClass {
+ return [AVCaptureVideoPreviewLayer class];
+}
+
+- (void)setCaptureSession:(AVCaptureSession *)captureSession {
+ if (_captureSession == captureSession) {
+ return;
+ }
+ _captureSession = captureSession;
+ AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer];
+ [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+ block:^{
+ previewLayer.session = captureSession;
+ }];
+}
+
+#pragma mark - Private
+
+- (AVCaptureVideoPreviewLayer *)previewLayer {
+ return (AVCaptureVideoPreviewLayer *)self.layer;
+}
+
+@end
diff --git a/webrtc/base/objc/RTCDispatcher.h b/webrtc/base/objc/RTCDispatcher.h
new file mode 100644
index 0000000..c32b93d
--- /dev/null
+++ b/webrtc/base/objc/RTCDispatcher.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
+ // Main dispatcher queue.
+ RTCDispatcherTypeMain,
+ // Used for starting/stopping AVCaptureSession, and assigning
+ // capture session to AVCaptureVideoPreviewLayer.
+ RTCDispatcherTypeCaptureSession,
+};
+
+/** Dispatcher that asynchronously dispatches blocks to a specific
+ * shared dispatch queue.
+ */
+@interface RTCDispatcher : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Dispatch the block asynchronously on the queue for dispatchType.
+ * @param dispatchType The queue type to dispatch on.
+ * @param block The block to dispatch asynchronously.
+ */
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType
+ block:(dispatch_block_t)block;
+
+@end
diff --git a/webrtc/base/objc/RTCDispatcher.m b/webrtc/base/objc/RTCDispatcher.m
new file mode 100644
index 0000000..065705a
--- /dev/null
+++ b/webrtc/base/objc/RTCDispatcher.m
@@ -0,0 +1,46 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDispatcher.h"
+
+static dispatch_queue_t kCaptureSessionQueue = nil;
+
+@implementation RTCDispatcher {
+ dispatch_queue_t _captureSessionQueue;
+}
+
++ (void)initialize {
+ static dispatch_once_t onceToken;
+ dispatch_once(&onceToken, ^{
+ kCaptureSessionQueue = dispatch_queue_create(
+ "org.webrtc.RTCDispatcherCaptureSession",
+ DISPATCH_QUEUE_SERIAL);
+ });
+}
+
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType
+ block:(dispatch_block_t)block {
+ dispatch_queue_t queue = [self dispatchQueueForType:dispatchType];
+ dispatch_async(queue, block);
+}
+
+#pragma mark - Private
+
++ (dispatch_queue_t)dispatchQueueForType:(RTCDispatcherQueueType)dispatchType {
+ switch (dispatchType) {
+ case RTCDispatcherTypeMain:
+ return dispatch_get_main_queue();
+ case RTCDispatcherTypeCaptureSession:
+ return kCaptureSessionQueue;
+ }
+}
+
+@end
+
diff --git a/webrtc/base/objc/RTCLogging.h b/webrtc/base/objc/RTCLogging.h
new file mode 100644
index 0000000..19fade5
--- /dev/null
+++ b/webrtc/base/objc/RTCLogging.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+// Subset of rtc::LoggingSeverity.
+typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
+ kRTCLoggingSeverityVerbose,
+ kRTCLoggingSeverityInfo,
+ kRTCLoggingSeverityWarning,
+ kRTCLoggingSeverityError,
+};
+
+#if defined(__cplusplus)
+extern "C" void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
+extern "C" void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
+extern "C" NSString* RTCFileName(const char* filePath);
+#else
+
+// Wrapper for C++ LOG(sev) macros.
+// Logs the log string to the webrtc logstream for the given severity.
+extern void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
+
+// Wrapper for rtc::LogMessage::LogToDebug.
+// Sets the minimum severity to be logged to console.
+extern void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
+
+// Returns the filename with the path prefix removed.
+extern NSString* RTCFileName(const char* filePath);
+
+#endif
+
+// Some convenience macros.
+
+#define RTCLogString(format, ...) \
+ [NSString stringWithFormat:@"(%@:%d %s): " format, \
+ RTCFileName(__FILE__), \
+ __LINE__, \
+ __FUNCTION__, \
+ ##__VA_ARGS__]
+
+#define RTCLogFormat(severity, format, ...) \
+ do { \
+ NSString* log_string = RTCLogString(format, ##__VA_ARGS__); \
+ RTCLogEx(severity, log_string); \
+ } while (false)
+
+#define RTCLogVerbose(format, ...) \
+ RTCLogFormat(kRTCLoggingSeverityVerbose, format, ##__VA_ARGS__) \
+
+#define RTCLogInfo(format, ...) \
+ RTCLogFormat(kRTCLoggingSeverityInfo, format, ##__VA_ARGS__) \
+
+#define RTCLogWarning(format, ...) \
+ RTCLogFormat(kRTCLoggingSeverityWarning, format, ##__VA_ARGS__) \
+
+#define RTCLogError(format, ...) \
+ RTCLogFormat(kRTCLoggingSeverityError, format, ##__VA_ARGS__) \
+
+#if !defined(NDEBUG)
+#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
+#else
+#define RTCLogDebug(format, ...) \
+ do { \
+ } while (false)
+#endif
+
+#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
diff --git a/webrtc/base/objc/RTCLogging.mm b/webrtc/base/objc/RTCLogging.mm
new file mode 100644
index 0000000..e9afe72
--- /dev/null
+++ b/webrtc/base/objc/RTCLogging.mm
@@ -0,0 +1,47 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCLogging.h"
+
+#include "webrtc/base/logging.h"
+
+rtc::LoggingSeverity RTCGetNativeLoggingSeverity(RTCLoggingSeverity severity) {
+ switch (severity) {
+ case kRTCLoggingSeverityVerbose:
+ return rtc::LS_VERBOSE;
+ case kRTCLoggingSeverityInfo:
+ return rtc::LS_INFO;
+ case kRTCLoggingSeverityWarning:
+ return rtc::LS_WARNING;
+ case kRTCLoggingSeverityError:
+ return rtc::LS_ERROR;
+ }
+}
+
+void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string) {
+ if (log_string.length) {
+ const char* utf8_string = log_string.UTF8String;
+ LOG_V(RTCGetNativeLoggingSeverity(severity)) << utf8_string;
+ }
+}
+
+void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity) {
+ rtc::LogMessage::LogToDebug(RTCGetNativeLoggingSeverity(severity));
+}
+
+NSString* RTCFileName(const char* file_path) {
+ NSString* ns_file_path =
+ [[NSString alloc] initWithBytesNoCopy:const_cast<char*>(file_path)
+ length:strlen(file_path)
+ encoding:NSUTF8StringEncoding
+ freeWhenDone:NO];
+ return ns_file_path.lastPathComponent;
+}
+
diff --git a/webrtc/base/openssladapter.cc b/webrtc/base/openssladapter.cc
index c906ebb..1f5fbbc 100644
--- a/webrtc/base/openssladapter.cc
+++ b/webrtc/base/openssladapter.cc
@@ -31,6 +31,7 @@
#include "config.h"
#endif // HAVE_CONFIG_H
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/openssl.h"
@@ -835,7 +836,7 @@
return ok;
}
-#if _DEBUG
+#if !defined(NDEBUG)
// We only use this for tracing and so it is only needed in debug mode
@@ -864,11 +865,11 @@
}
}
-#endif // _DEBUG
+#endif
int
OpenSSLAdapter::SSLVerifyCallback(int ok, X509_STORE_CTX* store) {
-#if _DEBUG
+#if !defined(NDEBUG)
if (!ok) {
char data[256];
X509* cert = X509_STORE_CTX_get_current_cert(store);
@@ -915,7 +916,7 @@
bool OpenSSLAdapter::ConfigureTrustedRootCertificates(SSL_CTX* ctx) {
// Add the root cert that we care about to the SSL context
int count_of_added_certs = 0;
- for (int i = 0; i < ARRAY_SIZE(kSSLCertCertificateList); i++) {
+ for (size_t i = 0; i < arraysize(kSSLCertCertificateList); i++) {
const unsigned char* cert_buffer = kSSLCertCertificateList[i];
size_t cert_buffer_len = kSSLCertCertificateSizeList[i];
X509* cert = d2i_X509(NULL, &cert_buffer,
@@ -949,7 +950,7 @@
return NULL;
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
SSL_CTX_set_info_callback(ctx, SSLInfoCallback);
#endif
diff --git a/webrtc/base/openssladapter.h b/webrtc/base/openssladapter.h
index 3dcb1c5..cdf45e6 100644
--- a/webrtc/base/openssladapter.h
+++ b/webrtc/base/openssladapter.h
@@ -67,9 +67,9 @@
static bool VerifyServerName(SSL* ssl, const char* host,
bool ignore_bad_cert);
bool SSLPostConnectionCheck(SSL* ssl, const char* host);
-#if _DEBUG
+#if !defined(NDEBUG)
static void SSLInfoCallback(const SSL* s, int where, int ret);
-#endif // !_DEBUG
+#endif
static int SSLVerifyCallback(int ok, X509_STORE_CTX* store);
static VerificationCallback custom_verify_callback_;
friend class OpenSSLStreamAdapter; // for custom_verify_callback_;
diff --git a/webrtc/base/opensslidentity.cc b/webrtc/base/opensslidentity.cc
index feda674..7185571 100644
--- a/webrtc/base/opensslidentity.cc
+++ b/webrtc/base/opensslidentity.cc
@@ -96,6 +96,7 @@
X509* x509 = NULL;
BIGNUM* serial_number = NULL;
X509_NAME* name = NULL;
+ time_t epoch_off = 0; // Time offset since epoch.
if ((x509=X509_new()) == NULL)
goto error;
@@ -130,8 +131,8 @@
!X509_set_issuer_name(x509, name))
goto error;
- if (!X509_gmtime_adj(X509_get_notBefore(x509), params.not_before) ||
- !X509_gmtime_adj(X509_get_notAfter(x509), params.not_after))
+ if (!X509_time_adj(X509_get_notBefore(x509), params.not_before, &epoch_off) ||
+ !X509_time_adj(X509_get_notAfter(x509), params.not_after, &epoch_off))
goto error;
if (!X509_sign(x509, pkey, EVP_sha256()))
@@ -186,7 +187,7 @@
#endif
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
// Print a certificate to the log, for debugging.
static void PrintCert(X509* x509) {
BIO* temp_memory_bio = BIO_new(BIO_s_mem());
@@ -215,7 +216,7 @@
LogSSLErrors("Generating certificate");
return NULL;
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
PrintCert(x509);
#endif
OpenSSLCertificate* ret = new OpenSSLCertificate(x509);
@@ -373,6 +374,22 @@
#endif
}
+// Documented in sslidentity.h.
+int64_t OpenSSLCertificate::CertificateExpirationTime() const {
+ ASN1_TIME* expire_time = X509_get_notAfter(x509_);
+ bool long_format;
+
+ if (expire_time->type == V_ASN1_UTCTIME) {
+ long_format = false;
+ } else if (expire_time->type == V_ASN1_GENERALIZEDTIME) {
+ long_format = true;
+ } else {
+ return -1;
+ }
+
+ return ASN1TimeToSec(expire_time->data, expire_time->length, long_format);
+}
+
OpenSSLIdentity::OpenSSLIdentity(OpenSSLKeyPair* key_pair,
OpenSSLCertificate* certificate)
: key_pair_(key_pair), certificate_(certificate) {
@@ -401,8 +418,9 @@
SSLIdentityParams params;
params.key_params = key_params;
params.common_name = common_name;
- params.not_before = CERTIFICATE_WINDOW;
- params.not_after = CERTIFICATE_LIFETIME;
+ time_t now = time(NULL);
+ params.not_before = now + CERTIFICATE_WINDOW;
+ params.not_after = now + CERTIFICATE_LIFETIME;
return GenerateInternal(params);
}
diff --git a/webrtc/base/opensslidentity.h b/webrtc/base/opensslidentity.h
index f957ef2..c8aa69a 100644
--- a/webrtc/base/opensslidentity.h
+++ b/webrtc/base/opensslidentity.h
@@ -87,6 +87,8 @@
bool GetSignatureDigestAlgorithm(std::string* algorithm) const override;
bool GetChain(SSLCertChain** chain) const override;
+ int64_t CertificateExpirationTime() const override;
+
private:
void AddReference() const;
diff --git a/webrtc/base/opensslstreamadapter.cc b/webrtc/base/opensslstreamadapter.cc
index 67ed5db..7563f17 100644
--- a/webrtc/base/opensslstreamadapter.cc
+++ b/webrtc/base/opensslstreamadapter.cc
@@ -43,17 +43,19 @@
#endif
#ifdef HAVE_DTLS_SRTP
-// SRTP cipher suite table
+// SRTP cipher suite table. |internal_name| is used to construct a
+// colon-separated profile strings which is needed by
+// SSL_CTX_set_tlsext_use_srtp().
struct SrtpCipherMapEntry {
- const char* external_name;
const char* internal_name;
+ const int id;
};
// This isn't elegant, but it's better than an external reference
static SrtpCipherMapEntry SrtpCipherMap[] = {
- {CS_AES_CM_128_HMAC_SHA1_80, "SRTP_AES128_CM_SHA1_80"},
- {CS_AES_CM_128_HMAC_SHA1_32, "SRTP_AES128_CM_SHA1_32"},
- {NULL, NULL}};
+ {"SRTP_AES128_CM_SHA1_80", SRTP_AES128_CM_SHA1_80},
+ {"SRTP_AES128_CM_SHA1_32", SRTP_AES128_CM_SHA1_32},
+ {nullptr, 0}};
#endif
#ifndef OPENSSL_IS_BORINGSSL
@@ -158,10 +160,12 @@
static int kDefaultSslEcCipher12 =
static_cast<uint16_t>(TLS1_CK_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256);
// Fallback cipher for DTLS 1.2 if hardware-accelerated AES-GCM is unavailable.
+// TODO(davidben): Switch to the standardized CHACHA20_POLY1305 variant when
+// available.
static int kDefaultSslCipher12NoAesGcm =
- static_cast<uint16_t>(TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305);
+ static_cast<uint16_t>(TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305_OLD);
static int kDefaultSslEcCipher12NoAesGcm =
- static_cast<uint16_t>(TLS1_CK_ECDHE_ECDSA_CHACHA20_POLY1305);
+ static_cast<uint16_t>(TLS1_CK_ECDHE_ECDSA_CHACHA20_POLY1305_OLD);
#else // !OPENSSL_IS_BORINGSSL
// OpenSSL sorts differently than BoringSSL, so the default cipher doesn't
// change between TLS 1.0 and TLS 1.2 with the current setup.
@@ -297,12 +301,13 @@
: SSLStreamAdapter(stream),
state_(SSL_NONE),
role_(SSL_CLIENT),
- ssl_read_needs_write_(false), ssl_write_needs_read_(false),
- ssl_(NULL), ssl_ctx_(NULL),
+ ssl_read_needs_write_(false),
+ ssl_write_needs_read_(false),
+ ssl_(NULL),
+ ssl_ctx_(NULL),
custom_verification_succeeded_(false),
ssl_mode_(SSL_MODE_TLS),
- ssl_max_version_(SSL_PROTOCOL_TLS_11) {
-}
+ ssl_max_version_(SSL_PROTOCOL_TLS_12) {}
OpenSSLStreamAdapter::~OpenSSLStreamAdapter() {
Cleanup();
@@ -348,9 +353,9 @@
return true;
}
-std::string OpenSSLStreamAdapter::GetSslCipherSuiteName(int cipher) {
+std::string OpenSSLStreamAdapter::SslCipherSuiteToName(int cipher_suite) {
#ifdef OPENSSL_IS_BORINGSSL
- const SSL_CIPHER* ssl_cipher = SSL_get_cipher_by_value(cipher);
+ const SSL_CIPHER* ssl_cipher = SSL_get_cipher_by_value(cipher_suite);
if (!ssl_cipher) {
return std::string();
}
@@ -361,7 +366,7 @@
#else
for (const SslCipherMapEntry* entry = kSslCipherMap; entry->rfc_name;
++entry) {
- if (cipher == entry->openssl_id) {
+ if (cipher_suite == static_cast<int>(entry->openssl_id)) {
return entry->rfc_name;
}
}
@@ -369,7 +374,7 @@
#endif
}
-bool OpenSSLStreamAdapter::GetSslCipherSuite(int* cipher) {
+bool OpenSSLStreamAdapter::GetSslCipherSuite(int* cipher_suite) {
if (state_ != SSL_CONNECTED)
return false;
@@ -378,7 +383,7 @@
return false;
}
- *cipher = static_cast<uint16_t>(SSL_CIPHER_get_id(current_cipher));
+ *cipher_suite = static_cast<uint16_t>(SSL_CIPHER_get_id(current_cipher));
return true;
}
@@ -405,20 +410,20 @@
#endif
}
-bool OpenSSLStreamAdapter::SetDtlsSrtpCiphers(
- const std::vector<std::string>& ciphers) {
+bool OpenSSLStreamAdapter::SetDtlsSrtpCryptoSuites(
+ const std::vector<int>& ciphers) {
#ifdef HAVE_DTLS_SRTP
std::string internal_ciphers;
if (state_ != SSL_NONE)
return false;
- for (std::vector<std::string>::const_iterator cipher = ciphers.begin();
+ for (std::vector<int>::const_iterator cipher = ciphers.begin();
cipher != ciphers.end(); ++cipher) {
bool found = false;
- for (SrtpCipherMapEntry *entry = SrtpCipherMap; entry->internal_name;
+ for (SrtpCipherMapEntry* entry = SrtpCipherMap; entry->internal_name;
++entry) {
- if (*cipher == entry->external_name) {
+ if (*cipher == entry->id) {
found = true;
if (!internal_ciphers.empty())
internal_ciphers += ":";
@@ -443,7 +448,7 @@
#endif
}
-bool OpenSSLStreamAdapter::GetDtlsSrtpCipher(std::string* cipher) {
+bool OpenSSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) {
#ifdef HAVE_DTLS_SRTP
ASSERT(state_ == SSL_CONNECTED);
if (state_ != SSL_CONNECTED)
@@ -455,17 +460,9 @@
if (!srtp_profile)
return false;
- for (SrtpCipherMapEntry *entry = SrtpCipherMap;
- entry->internal_name; ++entry) {
- if (!strcmp(entry->internal_name, srtp_profile->name)) {
- *cipher = entry->external_name;
- return true;
- }
- }
-
- ASSERT(false); // This should never happen
-
- return false;
+ *crypto_suite = srtp_profile->id;
+ ASSERT(!SrtpCryptoSuiteToName(*crypto_suite).empty());
+ return true;
#else
return false;
#endif
@@ -994,7 +991,7 @@
return NULL;
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
SSL_CTX_set_info_callback(ctx, OpenSSLAdapter::SSLInfoCallback);
#endif
diff --git a/webrtc/base/opensslstreamadapter.h b/webrtc/base/opensslstreamadapter.h
index 0f3ded9..e57b2a3 100644
--- a/webrtc/base/opensslstreamadapter.h
+++ b/webrtc/base/opensslstreamadapter.h
@@ -88,7 +88,7 @@
StreamState GetState() const override;
// TODO(guoweis): Move this away from a static class method.
- static std::string GetSslCipherSuiteName(int cipher);
+ static std::string SslCipherSuiteToName(int crypto_suite);
bool GetSslCipherSuite(int* cipher) override;
@@ -101,8 +101,8 @@
size_t result_len) override;
// DTLS-SRTP interface
- bool SetDtlsSrtpCiphers(const std::vector<std::string>& ciphers) override;
- bool GetDtlsSrtpCipher(std::string* cipher) override;
+ bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites) override;
+ bool GetDtlsSrtpCryptoSuite(int* crypto_suite) override;
// Capabilities interfaces
static bool HaveDtls();
diff --git a/webrtc/base/optional.h b/webrtc/base/optional.h
new file mode 100644
index 0000000..b8071e6
--- /dev/null
+++ b/webrtc/base/optional.h
@@ -0,0 +1,139 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_OPTIONAL_H_
+#define WEBRTC_BASE_OPTIONAL_H_
+
+#include <algorithm>
+#include <utility>
+
+#include "webrtc/base/checks.h"
+
+namespace rtc {
+
+// Simple std::experimental::optional-wannabe. It either contains a T or not.
+// In order to keep the implementation simple and portable, this implementation
+// actually contains a (default-constructed) T even when it supposedly doesn't
+// contain a value; use e.g. rtc::scoped_ptr<T> instead if that's too
+// expensive.
+//
+// A moved-from Optional<T> may only be destroyed, and assigned to if T allows
+// being assigned to after having been moved from. Specifically, you may not
+// assume that it just doesn't contain a value anymore.
+//
+// Examples of good places to use Optional:
+//
+// - As a class or struct member, when the member doesn't always have a value:
+// struct Prisoner {
+// std::string name;
+// Optional<int> cell_number; // Empty if not currently incarcerated.
+// };
+//
+// - As a return value for functions that may fail to return a value on all
+// allowed inputs. For example, a function that searches an array might
+// return an Optional<size_t> (the index where it found the element, or
+// nothing if it didn't find it); and a function that parses numbers might
+// return Optional<double> (the parsed number, or nothing if parsing failed).
+//
+// Examples of bad places to use Optional:
+//
+// - As a return value for functions that may fail because of disallowed
+// inputs. For example, a string length function should not return
+// Optional<size_t> so that it can return nothing in case the caller passed
+// it a null pointer; the function should probably use RTC_[D]CHECK instead,
+// and return plain size_t.
+//
+// - As a return value for functions that may fail to return a value on all
+// allowed inputs, but need to tell the caller what went wrong. Returning
+// Optional<double> when parsing a single number as in the example above
+// might make sense, but any larger parse job is probably going to need to
+// tell the caller what the problem was, not just that there was one.
+//
+// TODO(kwiberg): Get rid of this class when the standard library has
+// std::optional (and we're allowed to use it).
+template <typename T>
+class Optional final {
+ public:
+ // Construct an empty Optional.
+ Optional() : has_value_(false) {}
+
+ // Construct an Optional that contains a value.
+ explicit Optional(const T& val) : value_(val), has_value_(true) {}
+ explicit Optional(T&& val) : value_(std::move(val)), has_value_(true) {}
+
+ // Copy and move constructors.
+ // TODO(kwiberg): =default the move constructor when MSVC supports it.
+ Optional(const Optional&) = default;
+ Optional(Optional&& m)
+ : value_(std::move(m.value_)), has_value_(m.has_value_) {}
+
+ // Assignment.
+ // TODO(kwiberg): =default the move assignment op when MSVC supports it.
+ Optional& operator=(const Optional&) = default;
+ Optional& operator=(Optional&& m) {
+ value_ = std::move(m.value_);
+ has_value_ = m.has_value_;
+ return *this;
+ }
+
+ friend void swap(Optional& m1, Optional& m2) {
+ using std::swap;
+ swap(m1.value_, m2.value_);
+ swap(m1.has_value_, m2.has_value_);
+ }
+
+ // Conversion to bool to test if we have a value.
+ explicit operator bool() const { return has_value_; }
+
+ // Dereferencing. Only allowed if we have a value.
+ const T* operator->() const {
+ RTC_DCHECK(has_value_);
+ return &value_;
+ }
+ T* operator->() {
+ RTC_DCHECK(has_value_);
+ return &value_;
+ }
+ const T& operator*() const {
+ RTC_DCHECK(has_value_);
+ return value_;
+ }
+ T& operator*() {
+ RTC_DCHECK(has_value_);
+ return value_;
+ }
+
+ // Dereference with a default value in case we don't have a value.
+ const T& value_or(const T& default_val) const {
+ return has_value_ ? value_ : default_val;
+ }
+
+ // Equality tests. Two Optionals are equal if they contain equivalent values,
+ // or
+ // if they're both empty.
+ friend bool operator==(const Optional& m1, const Optional& m2) {
+ return m1.has_value_ && m2.has_value_ ? m1.value_ == m2.value_
+ : m1.has_value_ == m2.has_value_;
+ }
+ friend bool operator!=(const Optional& m1, const Optional& m2) {
+ return m1.has_value_ && m2.has_value_ ? m1.value_ != m2.value_
+ : m1.has_value_ != m2.has_value_;
+ }
+
+ private:
+ // Invariant: Unless *this has been moved from, value_ is default-initialized
+ // (or copied or moved from a default-initialized T) if !has_value_.
+ T value_;
+ bool has_value_;
+};
+
+} // namespace rtc
+
+#endif // WEBRTC_BASE_OPTIONAL_H_
diff --git a/webrtc/base/optional_unittest.cc b/webrtc/base/optional_unittest.cc
new file mode 100644
index 0000000..eabf091
--- /dev/null
+++ b/webrtc/base/optional_unittest.cc
@@ -0,0 +1,489 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <sstream>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/optional.h"
+
+namespace rtc {
+
+namespace {
+
+// Class whose instances logs various method calls (constructor, destructor,
+// etc.). Each instance has a unique ID (a simple global sequence number) and
+// an origin ID. When a copy is made, the new object gets a fresh ID but copies
+// the origin ID from the original. When a new Logger is created from scratch,
+// it gets a fresh ID, and the origin ID is the same as the ID (default
+// constructor) or given as an argument (explicit constructor).
+class Logger {
+ public:
+ Logger() : id_(next_id_++), origin_(id_) { Log("default constructor"); }
+ explicit Logger(int origin) : id_(next_id_++), origin_(origin) {
+ Log("explicit constructor");
+ }
+ Logger(const Logger& other) : id_(next_id_++), origin_(other.origin_) {
+ LogFrom("copy constructor", other);
+ }
+ Logger(Logger&& other) : id_(next_id_++), origin_(other.origin_) {
+ LogFrom("move constructor", other);
+ }
+ ~Logger() { Log("destructor"); }
+ Logger& operator=(const Logger& other) {
+ origin_ = other.origin_;
+ LogFrom("operator= copy", other);
+ return *this;
+ }
+ Logger& operator=(Logger&& other) {
+ origin_ = other.origin_;
+ LogFrom("operator= move", other);
+ return *this;
+ }
+ friend void swap(Logger& a, Logger& b) {
+ using std::swap;
+ swap(a.origin_, b.origin_);
+ Log2("swap", a, b);
+ }
+ friend bool operator==(const Logger& a, const Logger& b) {
+ Log2("operator==", a, b);
+ return a.origin_ == b.origin_;
+ }
+ friend bool operator!=(const Logger& a, const Logger& b) {
+ Log2("operator!=", a, b);
+ return a.origin_ != b.origin_;
+ }
+ void Foo() { Log("Foo()"); }
+ void Foo() const { Log("Foo() const"); }
+ static rtc::scoped_ptr<std::vector<std::string>> Setup() {
+ auto s = rtc_make_scoped_ptr(new std::vector<std::string>);
+ Logger::log_ = s.get();
+ Logger::next_id_ = 0;
+ return s;
+ }
+
+ private:
+ int id_;
+ int origin_;
+ static std::vector<std::string>* log_;
+ static int next_id_;
+ void Log(const char* msg) const {
+ std::ostringstream oss;
+ oss << id_ << ':' << origin_ << ". " << msg;
+ log_->push_back(oss.str());
+ }
+ void LogFrom(const char* msg, const Logger& other) const {
+ std::ostringstream oss;
+ oss << id_ << ':' << origin_ << ". " << msg << " (from " << other.id_ << ':'
+ << other.origin_ << ")";
+ log_->push_back(oss.str());
+ }
+ static void Log2(const char* msg, const Logger& a, const Logger& b) {
+ std::ostringstream oss;
+ oss << msg << ' ' << a.id_ << ':' << a.origin_ << ", " << b.id_ << ':'
+ << b.origin_;
+ log_->push_back(oss.str());
+ }
+};
+
+std::vector<std::string>* Logger::log_ = nullptr;
+int Logger::next_id_ = 0;
+
+// Append all the other args to the vector pointed to by the first arg.
+template <typename T>
+void VectorAppend(std::vector<T>* v) {}
+template <typename T, typename... Ts>
+void VectorAppend(std::vector<T>* v, const T& e, Ts... es) {
+ v->push_back(e);
+ VectorAppend(v, es...);
+}
+
+// Create a vector of strings. Because we're not allowed to use
+// std::initializer_list.
+template <typename... Ts>
+std::vector<std::string> V(Ts... es) {
+ std::vector<std::string> strings;
+ VectorAppend(&strings, static_cast<std::string>(es)...);
+ return strings;
+}
+
+} // namespace
+
+TEST(OptionalTest, TestConstructDefault) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x;
+ EXPECT_FALSE(x);
+ }
+ EXPECT_EQ(V("0:0. default constructor", "0:0. destructor"), *log);
+}
+
+TEST(OptionalTest, TestConstructCopyEmpty) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x;
+ EXPECT_FALSE(x);
+ auto y = x;
+ EXPECT_FALSE(y);
+ }
+ EXPECT_EQ(V("0:0. default constructor", "1:0. copy constructor (from 0:0)",
+ "1:0. destructor", "0:0. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestConstructCopyFull) {
+ auto log = Logger::Setup();
+ {
+ Logger a;
+ Optional<Logger> x(a);
+ EXPECT_TRUE(x);
+ log->push_back("---");
+ auto y = x;
+ EXPECT_TRUE(y);
+ log->push_back("---");
+ }
+ EXPECT_EQ(V("0:0. default constructor", "1:0. copy constructor (from 0:0)",
+ "---", "2:0. copy constructor (from 1:0)", "---",
+ "2:0. destructor", "1:0. destructor", "0:0. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestConstructMoveEmpty) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x;
+ EXPECT_FALSE(x);
+ auto y = std::move(x);
+ EXPECT_FALSE(y);
+ }
+ EXPECT_EQ(V("0:0. default constructor", "1:0. move constructor (from 0:0)",
+ "1:0. destructor", "0:0. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestConstructMoveFull) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x(Logger(17));
+ EXPECT_TRUE(x);
+ log->push_back("---");
+ auto y = std::move(x);
+ EXPECT_TRUE(x);
+ EXPECT_TRUE(y);
+ log->push_back("---");
+ }
+ EXPECT_EQ(
+ V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+ "0:17. destructor", "---", "2:17. move constructor (from 1:17)", "---",
+ "2:17. destructor", "1:17. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToEmptyFromEmpty) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x, y;
+ x = y;
+ }
+ EXPECT_EQ(
+ V("0:0. default constructor", "1:1. default constructor",
+ "0:1. operator= copy (from 1:1)", "1:1. destructor", "0:1. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToFullFromEmpty) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x(Logger(17));
+ Optional<Logger> y;
+ log->push_back("---");
+ x = y;
+ log->push_back("---");
+ }
+ EXPECT_EQ(
+ V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+ "0:17. destructor", "2:2. default constructor", "---",
+ "1:2. operator= copy (from 2:2)", "---", "2:2. destructor",
+ "1:2. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToEmptyFromFull) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x;
+ Optional<Logger> y(Logger(17));
+ log->push_back("---");
+ x = y;
+ log->push_back("---");
+ }
+ EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor",
+ "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
+ "0:17. operator= copy (from 2:17)", "---", "2:17. destructor",
+ "0:17. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToFullFromFull) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x(Logger(17));
+ Optional<Logger> y(Logger(42));
+ log->push_back("---");
+ x = y;
+ log->push_back("---");
+ }
+ EXPECT_EQ(
+ V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+ "0:17. destructor", "2:42. explicit constructor",
+ "3:42. move constructor (from 2:42)", "2:42. destructor", "---",
+ "1:42. operator= copy (from 3:42)", "---", "3:42. destructor",
+ "1:42. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToEmptyFromT) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x;
+ Logger y(17);
+ log->push_back("---");
+ x = Optional<Logger>(y);
+ log->push_back("---");
+ }
+ EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor", "---",
+ "2:17. copy constructor (from 1:17)",
+ "0:17. operator= move (from 2:17)", "2:17. destructor", "---",
+ "1:17. destructor", "0:17. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestCopyAssignToFullFromT) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x(Logger(17));
+ Logger y(42);
+ log->push_back("---");
+ x = Optional<Logger>(y);
+ log->push_back("---");
+ }
+ EXPECT_EQ(
+ V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+ "0:17. destructor", "2:42. explicit constructor", "---",
+ "3:42. copy constructor (from 2:42)",
+ "1:42. operator= move (from 3:42)", "3:42. destructor", "---",
+ "2:42. destructor", "1:42. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToEmptyFromEmpty) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x, y;
+ x = std::move(y);
+ }
+ EXPECT_EQ(
+ V("0:0. default constructor", "1:1. default constructor",
+ "0:1. operator= move (from 1:1)", "1:1. destructor", "0:1. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToFullFromEmpty) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x(Logger(17));
+ Optional<Logger> y;
+ log->push_back("---");
+ x = std::move(y);
+ log->push_back("---");
+ }
+ EXPECT_EQ(
+ V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+ "0:17. destructor", "2:2. default constructor", "---",
+ "1:2. operator= move (from 2:2)", "---", "2:2. destructor",
+ "1:2. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToEmptyFromFull) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x;
+ Optional<Logger> y(Logger(17));
+ log->push_back("---");
+ x = std::move(y);
+ log->push_back("---");
+ }
+ EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor",
+ "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
+ "0:17. operator= move (from 2:17)", "---", "2:17. destructor",
+ "0:17. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToFullFromFull) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x(Logger(17));
+ Optional<Logger> y(Logger(42));
+ log->push_back("---");
+ x = std::move(y);
+ log->push_back("---");
+ }
+ EXPECT_EQ(
+ V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+ "0:17. destructor", "2:42. explicit constructor",
+ "3:42. move constructor (from 2:42)", "2:42. destructor", "---",
+ "1:42. operator= move (from 3:42)", "---", "3:42. destructor",
+ "1:42. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToEmptyFromT) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x;
+ Logger y(17);
+ log->push_back("---");
+ x = Optional<Logger>(std::move(y));
+ log->push_back("---");
+ }
+ EXPECT_EQ(V("0:0. default constructor", "1:17. explicit constructor", "---",
+ "2:17. move constructor (from 1:17)",
+ "0:17. operator= move (from 2:17)", "2:17. destructor", "---",
+ "1:17. destructor", "0:17. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestMoveAssignToFullFromT) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x(Logger(17));
+ Logger y(42);
+ log->push_back("---");
+ x = Optional<Logger>(std::move(y));
+ log->push_back("---");
+ }
+ EXPECT_EQ(
+ V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
+ "0:17. destructor", "2:42. explicit constructor", "---",
+ "3:42. move constructor (from 2:42)",
+ "1:42. operator= move (from 3:42)", "3:42. destructor", "---",
+ "2:42. destructor", "1:42. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestDereference) {
+ auto log = Logger::Setup();
+ {
+ Optional<Logger> x(Logger(42));
+ const auto& y = x;
+ log->push_back("---");
+ x->Foo();
+ y->Foo();
+ std::move(x)->Foo();
+ std::move(y)->Foo();
+ log->push_back("---");
+ (*x).Foo();
+ (*y).Foo();
+ (*std::move(x)).Foo();
+ (*std::move(y)).Foo();
+ log->push_back("---");
+ }
+ EXPECT_EQ(V("0:42. explicit constructor",
+ "1:42. move constructor (from 0:42)", "0:42. destructor", "---",
+ "1:42. Foo()", "1:42. Foo() const", "1:42. Foo()",
+ "1:42. Foo() const", "---", "1:42. Foo()", "1:42. Foo() const",
+ "1:42. Foo()", "1:42. Foo() const", "---", "1:42. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestDereferenceWithDefault) {
+ auto log = Logger::Setup();
+ {
+ const Logger a(17), b(42);
+ Optional<Logger> x(a);
+ Optional<Logger> y;
+ log->push_back("-1-");
+ EXPECT_EQ(a, x.value_or(Logger(42)));
+ log->push_back("-2-");
+ EXPECT_EQ(b, y.value_or(Logger(42)));
+ log->push_back("-3-");
+ EXPECT_EQ(a, Optional<Logger>(Logger(17)).value_or(b));
+ log->push_back("-4-");
+ EXPECT_EQ(b, Optional<Logger>().value_or(b));
+ log->push_back("-5-");
+ }
+ EXPECT_EQ(
+ V("0:17. explicit constructor", "1:42. explicit constructor",
+ "2:17. copy constructor (from 0:17)", "3:3. default constructor", "-1-",
+ "4:42. explicit constructor", "operator== 0:17, 2:17",
+ "4:42. destructor", "-2-", "5:42. explicit constructor",
+ "operator== 1:42, 5:42", "5:42. destructor", "-3-",
+ "6:17. explicit constructor", "7:17. move constructor (from 6:17)",
+ "operator== 0:17, 7:17", "7:17. destructor", "6:17. destructor", "-4-",
+ "8:8. default constructor", "operator== 1:42, 1:42", "8:8. destructor",
+ "-5-", "3:3. destructor", "2:17. destructor", "1:42. destructor",
+ "0:17. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestEquality) {
+ auto log = Logger::Setup();
+ {
+ Logger a(17), b(42);
+ Optional<Logger> ma1(a), ma2(a), mb(b), me1, me2;
+ log->push_back("---");
+ EXPECT_EQ(ma1, ma1);
+ EXPECT_EQ(ma1, ma2);
+ EXPECT_NE(ma1, mb);
+ EXPECT_NE(ma1, me1);
+ EXPECT_EQ(me1, me1);
+ EXPECT_EQ(me1, me2);
+ log->push_back("---");
+ }
+ EXPECT_EQ(V("0:17. explicit constructor", "1:42. explicit constructor",
+ "2:17. copy constructor (from 0:17)",
+ "3:17. copy constructor (from 0:17)",
+ "4:42. copy constructor (from 1:42)", "5:5. default constructor",
+ "6:6. default constructor", "---", "operator== 2:17, 2:17",
+ "operator== 2:17, 3:17", "operator!= 2:17, 4:42", "---",
+ "6:6. destructor", "5:5. destructor", "4:42. destructor",
+ "3:17. destructor", "2:17. destructor", "1:42. destructor",
+ "0:17. destructor"),
+ *log);
+}
+
+TEST(OptionalTest, TestSwap) {
+ auto log = Logger::Setup();
+ {
+ Logger a(17), b(42);
+ Optional<Logger> x1(a), x2(b), y1(a), y2, z1, z2;
+ log->push_back("---");
+ swap(x1, x2); // Swap full <-> full.
+ swap(y1, y2); // Swap full <-> empty.
+ swap(z1, z2); // Swap empty <-> empty.
+ log->push_back("---");
+ }
+ EXPECT_EQ(V("0:17. explicit constructor", "1:42. explicit constructor",
+ "2:17. copy constructor (from 0:17)",
+ "3:42. copy constructor (from 1:42)",
+ "4:17. copy constructor (from 0:17)", "5:5. default constructor",
+ "6:6. default constructor", "7:7. default constructor", "---",
+ "swap 2:42, 3:17", "swap 4:5, 5:17", "swap 6:7, 7:6", "---",
+ "7:6. destructor", "6:7. destructor", "5:17. destructor",
+ "4:5. destructor", "3:17. destructor", "2:42. destructor",
+ "1:42. destructor", "0:17. destructor"),
+ *log);
+}
+
+} // namespace rtc
diff --git a/webrtc/base/physicalsocketserver.cc b/webrtc/base/physicalsocketserver.cc
index 86abcf2..3e45452 100644
--- a/webrtc/base/physicalsocketserver.cc
+++ b/webrtc/base/physicalsocketserver.cc
@@ -39,11 +39,11 @@
#include <algorithm>
#include <map>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
-#include "webrtc/base/nethelpers.h"
#include "webrtc/base/physicalsocketserver.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/base/winping.h"
@@ -96,463 +96,669 @@
static const int ICMP_PING_TIMEOUT_MILLIS = 10000u;
#endif
-class PhysicalSocket : public AsyncSocket, public sigslot::has_slots<> {
- public:
- PhysicalSocket(PhysicalSocketServer* ss, SOCKET s = INVALID_SOCKET)
- : ss_(ss), s_(s), enabled_events_(0), error_(0),
- state_((s == INVALID_SOCKET) ? CS_CLOSED : CS_CONNECTED),
- resolver_(NULL) {
+PhysicalSocket::PhysicalSocket(PhysicalSocketServer* ss, SOCKET s)
+ : ss_(ss), s_(s), enabled_events_(0), error_(0),
+ state_((s == INVALID_SOCKET) ? CS_CLOSED : CS_CONNECTED),
+ resolver_(nullptr) {
#if defined(WEBRTC_WIN)
- // EnsureWinsockInit() ensures that winsock is initialized. The default
- // version of this function doesn't do anything because winsock is
- // initialized by constructor of a static object. If neccessary libjingle
- // users can link it with a different version of this function by replacing
- // win32socketinit.cc. See win32socketinit.cc for more details.
- EnsureWinsockInit();
+ // EnsureWinsockInit() ensures that winsock is initialized. The default
+ // version of this function doesn't do anything because winsock is
+ // initialized by constructor of a static object. If neccessary libjingle
+ // users can link it with a different version of this function by replacing
+ // win32socketinit.cc. See win32socketinit.cc for more details.
+ EnsureWinsockInit();
#endif
- if (s_ != INVALID_SOCKET) {
- enabled_events_ = DE_READ | DE_WRITE;
+ if (s_ != INVALID_SOCKET) {
+ enabled_events_ = DE_READ | DE_WRITE;
- int type = SOCK_STREAM;
- socklen_t len = sizeof(type);
- VERIFY(0 == getsockopt(s_, SOL_SOCKET, SO_TYPE, (SockOptArg)&type, &len));
- udp_ = (SOCK_DGRAM == type);
- }
- }
-
- ~PhysicalSocket() override {
- Close();
- }
-
- // Creates the underlying OS socket (same as the "socket" function).
- virtual bool Create(int family, int type) {
- Close();
- s_ = ::socket(family, type, 0);
+ int type = SOCK_STREAM;
+ socklen_t len = sizeof(type);
+ VERIFY(0 == getsockopt(s_, SOL_SOCKET, SO_TYPE, (SockOptArg)&type, &len));
udp_ = (SOCK_DGRAM == type);
- UpdateLastError();
- if (udp_)
- enabled_events_ = DE_READ | DE_WRITE;
- return s_ != INVALID_SOCKET;
}
+}
- SocketAddress GetLocalAddress() const override {
- sockaddr_storage addr_storage = {0};
- socklen_t addrlen = sizeof(addr_storage);
- sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
- int result = ::getsockname(s_, addr, &addrlen);
- SocketAddress address;
- if (result >= 0) {
- SocketAddressFromSockAddrStorage(addr_storage, &address);
- } else {
- LOG(LS_WARNING) << "GetLocalAddress: unable to get local addr, socket="
- << s_;
- }
- return address;
+PhysicalSocket::~PhysicalSocket() {
+ Close();
+}
+
+bool PhysicalSocket::Create(int family, int type) {
+ Close();
+ s_ = ::socket(family, type, 0);
+ udp_ = (SOCK_DGRAM == type);
+ UpdateLastError();
+ if (udp_)
+ enabled_events_ = DE_READ | DE_WRITE;
+ return s_ != INVALID_SOCKET;
+}
+
+SocketAddress PhysicalSocket::GetLocalAddress() const {
+ sockaddr_storage addr_storage = {0};
+ socklen_t addrlen = sizeof(addr_storage);
+ sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+ int result = ::getsockname(s_, addr, &addrlen);
+ SocketAddress address;
+ if (result >= 0) {
+ SocketAddressFromSockAddrStorage(addr_storage, &address);
+ } else {
+ LOG(LS_WARNING) << "GetLocalAddress: unable to get local addr, socket="
+ << s_;
}
+ return address;
+}
- SocketAddress GetRemoteAddress() const override {
- sockaddr_storage addr_storage = {0};
- socklen_t addrlen = sizeof(addr_storage);
- sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
- int result = ::getpeername(s_, addr, &addrlen);
- SocketAddress address;
- if (result >= 0) {
- SocketAddressFromSockAddrStorage(addr_storage, &address);
- } else {
- LOG(LS_WARNING) << "GetRemoteAddress: unable to get remote addr, socket="
- << s_;
- }
- return address;
+SocketAddress PhysicalSocket::GetRemoteAddress() const {
+ sockaddr_storage addr_storage = {0};
+ socklen_t addrlen = sizeof(addr_storage);
+ sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+ int result = ::getpeername(s_, addr, &addrlen);
+ SocketAddress address;
+ if (result >= 0) {
+ SocketAddressFromSockAddrStorage(addr_storage, &address);
+ } else {
+ LOG(LS_WARNING) << "GetRemoteAddress: unable to get remote addr, socket="
+ << s_;
}
+ return address;
+}
- int Bind(const SocketAddress& bind_addr) override {
- sockaddr_storage addr_storage;
- size_t len = bind_addr.ToSockAddrStorage(&addr_storage);
- sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
- int err = ::bind(s_, addr, static_cast<int>(len));
- UpdateLastError();
-#ifdef _DEBUG
- if (0 == err) {
- dbg_addr_ = "Bound @ ";
- dbg_addr_.append(GetLocalAddress().ToString());
- }
-#endif // _DEBUG
- return err;
+int PhysicalSocket::Bind(const SocketAddress& bind_addr) {
+ sockaddr_storage addr_storage;
+ size_t len = bind_addr.ToSockAddrStorage(&addr_storage);
+ sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+ int err = ::bind(s_, addr, static_cast<int>(len));
+ UpdateLastError();
+#if !defined(NDEBUG)
+ if (0 == err) {
+ dbg_addr_ = "Bound @ ";
+ dbg_addr_.append(GetLocalAddress().ToString());
}
+#endif
+ return err;
+}
- int Connect(const SocketAddress& addr) override {
- // TODO: Implicit creation is required to reconnect...
- // ...but should we make it more explicit?
- if (state_ != CS_CLOSED) {
- SetError(EALREADY);
- return SOCKET_ERROR;
- }
- if (addr.IsUnresolved()) {
- LOG(LS_VERBOSE) << "Resolving addr in PhysicalSocket::Connect";
- resolver_ = new AsyncResolver();
- resolver_->SignalDone.connect(this, &PhysicalSocket::OnResolveResult);
- resolver_->Start(addr);
- state_ = CS_CONNECTING;
- return 0;
- }
-
- return DoConnect(addr);
+int PhysicalSocket::Connect(const SocketAddress& addr) {
+ // TODO(pthatcher): Implicit creation is required to reconnect...
+ // ...but should we make it more explicit?
+ if (state_ != CS_CLOSED) {
+ SetError(EALREADY);
+ return SOCKET_ERROR;
}
-
- int DoConnect(const SocketAddress& connect_addr) {
- if ((s_ == INVALID_SOCKET) &&
- !Create(connect_addr.family(), SOCK_STREAM)) {
- return SOCKET_ERROR;
- }
- sockaddr_storage addr_storage;
- size_t len = connect_addr.ToSockAddrStorage(&addr_storage);
- sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
- int err = ::connect(s_, addr, static_cast<int>(len));
- UpdateLastError();
- if (err == 0) {
- state_ = CS_CONNECTED;
- } else if (IsBlockingError(GetError())) {
- state_ = CS_CONNECTING;
- enabled_events_ |= DE_CONNECT;
- } else {
- return SOCKET_ERROR;
- }
-
- enabled_events_ |= DE_READ | DE_WRITE;
+ if (addr.IsUnresolvedIP()) {
+ LOG(LS_VERBOSE) << "Resolving addr in PhysicalSocket::Connect";
+ resolver_ = new AsyncResolver();
+ resolver_->SignalDone.connect(this, &PhysicalSocket::OnResolveResult);
+ resolver_->Start(addr);
+ state_ = CS_CONNECTING;
return 0;
}
- int GetError() const override {
- CritScope cs(&crit_);
- return error_;
+ return DoConnect(addr);
+}
+
+int PhysicalSocket::DoConnect(const SocketAddress& connect_addr) {
+ if ((s_ == INVALID_SOCKET) &&
+ !Create(connect_addr.family(), SOCK_STREAM)) {
+ return SOCKET_ERROR;
+ }
+ sockaddr_storage addr_storage;
+ size_t len = connect_addr.ToSockAddrStorage(&addr_storage);
+ sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+ int err = ::connect(s_, addr, static_cast<int>(len));
+ UpdateLastError();
+ if (err == 0) {
+ state_ = CS_CONNECTED;
+ } else if (IsBlockingError(GetError())) {
+ state_ = CS_CONNECTING;
+ enabled_events_ |= DE_CONNECT;
+ } else {
+ return SOCKET_ERROR;
}
- void SetError(int error) override {
- CritScope cs(&crit_);
- error_ = error;
- }
+ enabled_events_ |= DE_READ | DE_WRITE;
+ return 0;
+}
- ConnState GetState() const override { return state_; }
+int PhysicalSocket::GetError() const {
+ CritScope cs(&crit_);
+ return error_;
+}
- int GetOption(Option opt, int* value) override {
- int slevel;
- int sopt;
- if (TranslateOption(opt, &slevel, &sopt) == -1)
- return -1;
- socklen_t optlen = sizeof(*value);
- int ret = ::getsockopt(s_, slevel, sopt, (SockOptArg)value, &optlen);
- if (ret != -1 && opt == OPT_DONTFRAGMENT) {
+void PhysicalSocket::SetError(int error) {
+ CritScope cs(&crit_);
+ error_ = error;
+}
+
+AsyncSocket::ConnState PhysicalSocket::GetState() const {
+ return state_;
+}
+
+int PhysicalSocket::GetOption(Option opt, int* value) {
+ int slevel;
+ int sopt;
+ if (TranslateOption(opt, &slevel, &sopt) == -1)
+ return -1;
+ socklen_t optlen = sizeof(*value);
+ int ret = ::getsockopt(s_, slevel, sopt, (SockOptArg)value, &optlen);
+ if (ret != -1 && opt == OPT_DONTFRAGMENT) {
#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
- *value = (*value != IP_PMTUDISC_DONT) ? 1 : 0;
+ *value = (*value != IP_PMTUDISC_DONT) ? 1 : 0;
#endif
- }
- return ret;
}
+ return ret;
+}
- int SetOption(Option opt, int value) override {
- int slevel;
- int sopt;
- if (TranslateOption(opt, &slevel, &sopt) == -1)
- return -1;
- if (opt == OPT_DONTFRAGMENT) {
+int PhysicalSocket::SetOption(Option opt, int value) {
+ int slevel;
+ int sopt;
+ if (TranslateOption(opt, &slevel, &sopt) == -1)
+ return -1;
+ if (opt == OPT_DONTFRAGMENT) {
#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
- value = (value) ? IP_PMTUDISC_DO : IP_PMTUDISC_DONT;
+ value = (value) ? IP_PMTUDISC_DO : IP_PMTUDISC_DONT;
#endif
- }
- return ::setsockopt(s_, slevel, sopt, (SockOptArg)&value, sizeof(value));
}
+ return ::setsockopt(s_, slevel, sopt, (SockOptArg)&value, sizeof(value));
+}
- int Send(const void* pv, size_t cb) override {
- int sent = ::send(s_, reinterpret_cast<const char *>(pv), (int)cb,
+int PhysicalSocket::Send(const void* pv, size_t cb) {
+ int sent = ::send(s_, reinterpret_cast<const char *>(pv), (int)cb,
#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
- // Suppress SIGPIPE. Without this, attempting to send on a socket whose
- // other end is closed will result in a SIGPIPE signal being raised to
- // our process, which by default will terminate the process, which we
- // don't want. By specifying this flag, we'll just get the error EPIPE
- // instead and can handle the error gracefully.
- MSG_NOSIGNAL
+ // Suppress SIGPIPE. Without this, attempting to send on a socket whose
+ // other end is closed will result in a SIGPIPE signal being raised to
+ // our process, which by default will terminate the process, which we
+ // don't want. By specifying this flag, we'll just get the error EPIPE
+ // instead and can handle the error gracefully.
+ MSG_NOSIGNAL
#else
- 0
+ 0
#endif
- );
- UpdateLastError();
- MaybeRemapSendError();
- // We have seen minidumps where this may be false.
- ASSERT(sent <= static_cast<int>(cb));
- if ((sent < 0) && IsBlockingError(GetError())) {
- enabled_events_ |= DE_WRITE;
- }
- return sent;
+ );
+ UpdateLastError();
+ MaybeRemapSendError();
+ // We have seen minidumps where this may be false.
+ ASSERT(sent <= static_cast<int>(cb));
+ if ((sent < 0) && IsBlockingError(GetError())) {
+ enabled_events_ |= DE_WRITE;
}
+ return sent;
+}
- int SendTo(const void* buffer,
- size_t length,
- const SocketAddress& addr) override {
- sockaddr_storage saddr;
- size_t len = addr.ToSockAddrStorage(&saddr);
- int sent = ::sendto(
- s_, static_cast<const char *>(buffer), static_cast<int>(length),
+int PhysicalSocket::SendTo(const void* buffer,
+ size_t length,
+ const SocketAddress& addr) {
+ sockaddr_storage saddr;
+ size_t len = addr.ToSockAddrStorage(&saddr);
+ int sent = ::sendto(
+ s_, static_cast<const char *>(buffer), static_cast<int>(length),
#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
- // Suppress SIGPIPE. See above for explanation.
- MSG_NOSIGNAL,
+ // Suppress SIGPIPE. See above for explanation.
+ MSG_NOSIGNAL,
#else
- 0,
+ 0,
#endif
- reinterpret_cast<sockaddr*>(&saddr), static_cast<int>(len));
- UpdateLastError();
- MaybeRemapSendError();
- // We have seen minidumps where this may be false.
- ASSERT(sent <= static_cast<int>(length));
- if ((sent < 0) && IsBlockingError(GetError())) {
- enabled_events_ |= DE_WRITE;
- }
- return sent;
+ reinterpret_cast<sockaddr*>(&saddr), static_cast<int>(len));
+ UpdateLastError();
+ MaybeRemapSendError();
+ // We have seen minidumps where this may be false.
+ ASSERT(sent <= static_cast<int>(length));
+ if ((sent < 0) && IsBlockingError(GetError())) {
+ enabled_events_ |= DE_WRITE;
}
+ return sent;
+}
- int Recv(void* buffer, size_t length) override {
- int received = ::recv(s_, static_cast<char*>(buffer),
- static_cast<int>(length), 0);
- if ((received == 0) && (length != 0)) {
- // Note: on graceful shutdown, recv can return 0. In this case, we
- // pretend it is blocking, and then signal close, so that simplifying
- // assumptions can be made about Recv.
- LOG(LS_WARNING) << "EOF from socket; deferring close event";
- // Must turn this back on so that the select() loop will notice the close
- // event.
- enabled_events_ |= DE_READ;
- SetError(EWOULDBLOCK);
- return SOCKET_ERROR;
- }
- UpdateLastError();
- int error = GetError();
- bool success = (received >= 0) || IsBlockingError(error);
- if (udp_ || success) {
- enabled_events_ |= DE_READ;
- }
- if (!success) {
- LOG_F(LS_VERBOSE) << "Error = " << error;
- }
- return received;
+int PhysicalSocket::Recv(void* buffer, size_t length) {
+ int received = ::recv(s_, static_cast<char*>(buffer),
+ static_cast<int>(length), 0);
+ if ((received == 0) && (length != 0)) {
+ // Note: on graceful shutdown, recv can return 0. In this case, we
+ // pretend it is blocking, and then signal close, so that simplifying
+ // assumptions can be made about Recv.
+ LOG(LS_WARNING) << "EOF from socket; deferring close event";
+ // Must turn this back on so that the select() loop will notice the close
+ // event.
+ enabled_events_ |= DE_READ;
+ SetError(EWOULDBLOCK);
+ return SOCKET_ERROR;
}
-
- int RecvFrom(void* buffer, size_t length, SocketAddress* out_addr) override {
- sockaddr_storage addr_storage;
- socklen_t addr_len = sizeof(addr_storage);
- sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
- int received = ::recvfrom(s_, static_cast<char*>(buffer),
- static_cast<int>(length), 0, addr, &addr_len);
- UpdateLastError();
- if ((received >= 0) && (out_addr != NULL))
- SocketAddressFromSockAddrStorage(addr_storage, out_addr);
- int error = GetError();
- bool success = (received >= 0) || IsBlockingError(error);
- if (udp_ || success) {
- enabled_events_ |= DE_READ;
- }
- if (!success) {
- LOG_F(LS_VERBOSE) << "Error = " << error;
- }
- return received;
+ UpdateLastError();
+ int error = GetError();
+ bool success = (received >= 0) || IsBlockingError(error);
+ if (udp_ || success) {
+ enabled_events_ |= DE_READ;
}
-
- int Listen(int backlog) override {
- int err = ::listen(s_, backlog);
- UpdateLastError();
- if (err == 0) {
- state_ = CS_CONNECTING;
- enabled_events_ |= DE_ACCEPT;
-#ifdef _DEBUG
- dbg_addr_ = "Listening @ ";
- dbg_addr_.append(GetLocalAddress().ToString());
-#endif // _DEBUG
- }
- return err;
+ if (!success) {
+ LOG_F(LS_VERBOSE) << "Error = " << error;
}
+ return received;
+}
- AsyncSocket* Accept(SocketAddress* out_addr) override {
- sockaddr_storage addr_storage;
- socklen_t addr_len = sizeof(addr_storage);
- sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
- SOCKET s = ::accept(s_, addr, &addr_len);
- UpdateLastError();
- if (s == INVALID_SOCKET)
- return NULL;
+int PhysicalSocket::RecvFrom(void* buffer,
+ size_t length,
+ SocketAddress* out_addr) {
+ sockaddr_storage addr_storage;
+ socklen_t addr_len = sizeof(addr_storage);
+ sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+ int received = ::recvfrom(s_, static_cast<char*>(buffer),
+ static_cast<int>(length), 0, addr, &addr_len);
+ UpdateLastError();
+ if ((received >= 0) && (out_addr != nullptr))
+ SocketAddressFromSockAddrStorage(addr_storage, out_addr);
+ int error = GetError();
+ bool success = (received >= 0) || IsBlockingError(error);
+ if (udp_ || success) {
+ enabled_events_ |= DE_READ;
+ }
+ if (!success) {
+ LOG_F(LS_VERBOSE) << "Error = " << error;
+ }
+ return received;
+}
+
+int PhysicalSocket::Listen(int backlog) {
+ int err = ::listen(s_, backlog);
+ UpdateLastError();
+ if (err == 0) {
+ state_ = CS_CONNECTING;
enabled_events_ |= DE_ACCEPT;
- if (out_addr != NULL)
- SocketAddressFromSockAddrStorage(addr_storage, out_addr);
- return ss_->WrapSocket(s);
+#if !defined(NDEBUG)
+ dbg_addr_ = "Listening @ ";
+ dbg_addr_.append(GetLocalAddress().ToString());
+#endif
}
+ return err;
+}
- int Close() override {
- if (s_ == INVALID_SOCKET)
- return 0;
- int err = ::closesocket(s_);
- UpdateLastError();
- s_ = INVALID_SOCKET;
- state_ = CS_CLOSED;
- enabled_events_ = 0;
- if (resolver_) {
- resolver_->Destroy(false);
- resolver_ = NULL;
- }
- return err;
+AsyncSocket* PhysicalSocket::Accept(SocketAddress* out_addr) {
+ // Always re-subscribe DE_ACCEPT to make sure new incoming connections will
+ // trigger an event even if DoAccept returns an error here.
+ enabled_events_ |= DE_ACCEPT;
+ sockaddr_storage addr_storage;
+ socklen_t addr_len = sizeof(addr_storage);
+ sockaddr* addr = reinterpret_cast<sockaddr*>(&addr_storage);
+ SOCKET s = DoAccept(s_, addr, &addr_len);
+ UpdateLastError();
+ if (s == INVALID_SOCKET)
+ return nullptr;
+ if (out_addr != nullptr)
+ SocketAddressFromSockAddrStorage(addr_storage, out_addr);
+ return ss_->WrapSocket(s);
+}
+
+int PhysicalSocket::Close() {
+ if (s_ == INVALID_SOCKET)
+ return 0;
+ int err = ::closesocket(s_);
+ UpdateLastError();
+ s_ = INVALID_SOCKET;
+ state_ = CS_CLOSED;
+ enabled_events_ = 0;
+ if (resolver_) {
+ resolver_->Destroy(false);
+ resolver_ = nullptr;
}
+ return err;
+}
- int EstimateMTU(uint16_t* mtu) override {
- SocketAddress addr = GetRemoteAddress();
- if (addr.IsAnyIP()) {
- SetError(ENOTCONN);
- return -1;
- }
+int PhysicalSocket::EstimateMTU(uint16_t* mtu) {
+ SocketAddress addr = GetRemoteAddress();
+ if (addr.IsAnyIP()) {
+ SetError(ENOTCONN);
+ return -1;
+ }
#if defined(WEBRTC_WIN)
- // Gets the interface MTU (TTL=1) for the interface used to reach |addr|.
- WinPing ping;
- if (!ping.IsValid()) {
+ // Gets the interface MTU (TTL=1) for the interface used to reach |addr|.
+ WinPing ping;
+ if (!ping.IsValid()) {
+ SetError(EINVAL); // can't think of a better error ID
+ return -1;
+ }
+ int header_size = ICMP_HEADER_SIZE;
+ if (addr.family() == AF_INET6) {
+ header_size += IPV6_HEADER_SIZE;
+ } else if (addr.family() == AF_INET) {
+ header_size += IP_HEADER_SIZE;
+ }
+
+ for (int level = 0; PACKET_MAXIMUMS[level + 1] > 0; ++level) {
+ int32_t size = PACKET_MAXIMUMS[level] - header_size;
+ WinPing::PingResult result = ping.Ping(addr.ipaddr(), size,
+ ICMP_PING_TIMEOUT_MILLIS,
+ 1, false);
+ if (result == WinPing::PING_FAIL) {
SetError(EINVAL); // can't think of a better error ID
return -1;
+ } else if (result != WinPing::PING_TOO_LARGE) {
+ *mtu = PACKET_MAXIMUMS[level];
+ return 0;
}
- int header_size = ICMP_HEADER_SIZE;
- if (addr.family() == AF_INET6) {
- header_size += IPV6_HEADER_SIZE;
- } else if (addr.family() == AF_INET) {
- header_size += IP_HEADER_SIZE;
- }
+ }
- for (int level = 0; PACKET_MAXIMUMS[level + 1] > 0; ++level) {
- int32_t size = PACKET_MAXIMUMS[level] - header_size;
- WinPing::PingResult result = ping.Ping(addr.ipaddr(), size,
- ICMP_PING_TIMEOUT_MILLIS,
- 1, false);
- if (result == WinPing::PING_FAIL) {
- SetError(EINVAL); // can't think of a better error ID
- return -1;
- } else if (result != WinPing::PING_TOO_LARGE) {
- *mtu = PACKET_MAXIMUMS[level];
- return 0;
- }
- }
-
- ASSERT(false);
- return -1;
+ ASSERT(false);
+ return -1;
#elif defined(WEBRTC_MAC)
- // No simple way to do this on Mac OS X.
- // SIOCGIFMTU would work if we knew which interface would be used, but
- // figuring that out is pretty complicated. For now we'll return an error
- // and let the caller pick a default MTU.
- SetError(EINVAL);
- return -1;
+ // No simple way to do this on Mac OS X.
+ // SIOCGIFMTU would work if we knew which interface would be used, but
+ // figuring that out is pretty complicated. For now we'll return an error
+ // and let the caller pick a default MTU.
+ SetError(EINVAL);
+ return -1;
#elif defined(WEBRTC_LINUX)
- // Gets the path MTU.
- int value;
- socklen_t vlen = sizeof(value);
- int err = getsockopt(s_, IPPROTO_IP, IP_MTU, &value, &vlen);
- if (err < 0) {
- UpdateLastError();
- return err;
- }
+ // Gets the path MTU.
+ int value;
+ socklen_t vlen = sizeof(value);
+ int err = getsockopt(s_, IPPROTO_IP, IP_MTU, &value, &vlen);
+ if (err < 0) {
+ UpdateLastError();
+ return err;
+ }
- ASSERT((0 <= value) && (value <= 65536));
- *mtu = value;
- return 0;
+ ASSERT((0 <= value) && (value <= 65536));
+ *mtu = value;
+ return 0;
#elif defined(__native_client__)
- // Most socket operations, including this, will fail in NaCl's sandbox.
- error_ = EACCES;
- return -1;
+ // Most socket operations, including this, will fail in NaCl's sandbox.
+ error_ = EACCES;
+ return -1;
#endif
+}
+
+
+SOCKET PhysicalSocket::DoAccept(SOCKET socket,
+ sockaddr* addr,
+ socklen_t* addrlen) {
+ return ::accept(socket, addr, addrlen);
+}
+
+void PhysicalSocket::OnResolveResult(AsyncResolverInterface* resolver) {
+ if (resolver != resolver_) {
+ return;
}
- SocketServer* socketserver() { return ss_; }
-
- protected:
- void OnResolveResult(AsyncResolverInterface* resolver) {
- if (resolver != resolver_) {
- return;
- }
-
- int error = resolver_->GetError();
- if (error == 0) {
- error = DoConnect(resolver_->address());
- } else {
- Close();
- }
-
- if (error) {
- SetError(error);
- SignalCloseEvent(this, error);
- }
+ int error = resolver_->GetError();
+ if (error == 0) {
+ error = DoConnect(resolver_->address());
+ } else {
+ Close();
}
- void UpdateLastError() {
- SetError(LAST_SYSTEM_ERROR);
+ if (error) {
+ SetError(error);
+ SignalCloseEvent(this, error);
}
+}
- void MaybeRemapSendError() {
+void PhysicalSocket::UpdateLastError() {
+ SetError(LAST_SYSTEM_ERROR);
+}
+
+void PhysicalSocket::MaybeRemapSendError() {
#if defined(WEBRTC_MAC)
- // https://developer.apple.com/library/mac/documentation/Darwin/
- // Reference/ManPages/man2/sendto.2.html
- // ENOBUFS - The output queue for a network interface is full.
- // This generally indicates that the interface has stopped sending,
- // but may be caused by transient congestion.
- if (GetError() == ENOBUFS) {
- SetError(EWOULDBLOCK);
- }
-#endif
+ // https://developer.apple.com/library/mac/documentation/Darwin/
+ // Reference/ManPages/man2/sendto.2.html
+ // ENOBUFS - The output queue for a network interface is full.
+ // This generally indicates that the interface has stopped sending,
+ // but may be caused by transient congestion.
+ if (GetError() == ENOBUFS) {
+ SetError(EWOULDBLOCK);
}
+#endif
+}
- static int TranslateOption(Option opt, int* slevel, int* sopt) {
- switch (opt) {
- case OPT_DONTFRAGMENT:
+int PhysicalSocket::TranslateOption(Option opt, int* slevel, int* sopt) {
+ switch (opt) {
+ case OPT_DONTFRAGMENT:
#if defined(WEBRTC_WIN)
- *slevel = IPPROTO_IP;
- *sopt = IP_DONTFRAGMENT;
- break;
+ *slevel = IPPROTO_IP;
+ *sopt = IP_DONTFRAGMENT;
+ break;
#elif defined(WEBRTC_MAC) || defined(BSD) || defined(__native_client__)
- LOG(LS_WARNING) << "Socket::OPT_DONTFRAGMENT not supported.";
- return -1;
+ LOG(LS_WARNING) << "Socket::OPT_DONTFRAGMENT not supported.";
+ return -1;
#elif defined(WEBRTC_POSIX)
- *slevel = IPPROTO_IP;
- *sopt = IP_MTU_DISCOVER;
- break;
+ *slevel = IPPROTO_IP;
+ *sopt = IP_MTU_DISCOVER;
+ break;
#endif
- case OPT_RCVBUF:
- *slevel = SOL_SOCKET;
- *sopt = SO_RCVBUF;
- break;
- case OPT_SNDBUF:
- *slevel = SOL_SOCKET;
- *sopt = SO_SNDBUF;
- break;
- case OPT_NODELAY:
- *slevel = IPPROTO_TCP;
- *sopt = TCP_NODELAY;
- break;
- case OPT_DSCP:
- LOG(LS_WARNING) << "Socket::OPT_DSCP not supported.";
- return -1;
- case OPT_RTP_SENDTIME_EXTN_ID:
- return -1; // No logging is necessary as this not a OS socket option.
- default:
- ASSERT(false);
- return -1;
- }
- return 0;
+ case OPT_RCVBUF:
+ *slevel = SOL_SOCKET;
+ *sopt = SO_RCVBUF;
+ break;
+ case OPT_SNDBUF:
+ *slevel = SOL_SOCKET;
+ *sopt = SO_SNDBUF;
+ break;
+ case OPT_NODELAY:
+ *slevel = IPPROTO_TCP;
+ *sopt = TCP_NODELAY;
+ break;
+ case OPT_DSCP:
+ LOG(LS_WARNING) << "Socket::OPT_DSCP not supported.";
+ return -1;
+ case OPT_RTP_SENDTIME_EXTN_ID:
+ return -1; // No logging is necessary as this not a OS socket option.
+ default:
+ ASSERT(false);
+ return -1;
}
+ return 0;
+}
- PhysicalSocketServer* ss_;
- SOCKET s_;
- uint8_t enabled_events_;
- bool udp_;
- int error_;
- // Protects |error_| that is accessed from different threads.
- mutable CriticalSection crit_;
- ConnState state_;
- AsyncResolver* resolver_;
+SocketDispatcher::SocketDispatcher(PhysicalSocketServer *ss)
+#if defined(WEBRTC_WIN)
+ : PhysicalSocket(ss), id_(0), signal_close_(false)
+#else
+ : PhysicalSocket(ss)
+#endif
+{
+}
-#ifdef _DEBUG
- std::string dbg_addr_;
-#endif // _DEBUG;
-};
+SocketDispatcher::SocketDispatcher(SOCKET s, PhysicalSocketServer *ss)
+#if defined(WEBRTC_WIN)
+ : PhysicalSocket(ss, s), id_(0), signal_close_(false)
+#else
+ : PhysicalSocket(ss, s)
+#endif
+{
+}
+
+SocketDispatcher::~SocketDispatcher() {
+ Close();
+}
+
+bool SocketDispatcher::Initialize() {
+ ASSERT(s_ != INVALID_SOCKET);
+ // Must be a non-blocking
+#if defined(WEBRTC_WIN)
+ u_long argp = 1;
+ ioctlsocket(s_, FIONBIO, &argp);
+#elif defined(WEBRTC_POSIX)
+ fcntl(s_, F_SETFL, fcntl(s_, F_GETFL, 0) | O_NONBLOCK);
+#endif
+ ss_->Add(this);
+ return true;
+}
+
+bool SocketDispatcher::Create(int type) {
+ return Create(AF_INET, type);
+}
+
+bool SocketDispatcher::Create(int family, int type) {
+ // Change the socket to be non-blocking.
+ if (!PhysicalSocket::Create(family, type))
+ return false;
+
+ if (!Initialize())
+ return false;
+
+#if defined(WEBRTC_WIN)
+ do { id_ = ++next_id_; } while (id_ == 0);
+#endif
+ return true;
+}
+
+#if defined(WEBRTC_WIN)
+
+WSAEVENT SocketDispatcher::GetWSAEvent() {
+ return WSA_INVALID_EVENT;
+}
+
+SOCKET SocketDispatcher::GetSocket() {
+ return s_;
+}
+
+bool SocketDispatcher::CheckSignalClose() {
+ if (!signal_close_)
+ return false;
+
+ char ch;
+ if (recv(s_, &ch, 1, MSG_PEEK) > 0)
+ return false;
+
+ state_ = CS_CLOSED;
+ signal_close_ = false;
+ SignalCloseEvent(this, signal_err_);
+ return true;
+}
+
+int SocketDispatcher::next_id_ = 0;
+
+#elif defined(WEBRTC_POSIX)
+
+int SocketDispatcher::GetDescriptor() {
+ return s_;
+}
+
+bool SocketDispatcher::IsDescriptorClosed() {
+ // We don't have a reliable way of distinguishing end-of-stream
+ // from readability. So test on each readable call. Is this
+ // inefficient? Probably.
+ char ch;
+ ssize_t res = ::recv(s_, &ch, 1, MSG_PEEK);
+ if (res > 0) {
+ // Data available, so not closed.
+ return false;
+ } else if (res == 0) {
+ // EOF, so closed.
+ return true;
+ } else { // error
+ switch (errno) {
+ // Returned if we've already closed s_.
+ case EBADF:
+ // Returned during ungraceful peer shutdown.
+ case ECONNRESET:
+ return true;
+ default:
+ // Assume that all other errors are just blocking errors, meaning the
+ // connection is still good but we just can't read from it right now.
+ // This should only happen when connecting (and at most once), because
+ // in all other cases this function is only called if the file
+ // descriptor is already known to be in the readable state. However,
+ // it's not necessary a problem if we spuriously interpret a
+ // "connection lost"-type error as a blocking error, because typically
+ // the next recv() will get EOF, so we'll still eventually notice that
+ // the socket is closed.
+ LOG_ERR(LS_WARNING) << "Assuming benign blocking error";
+ return false;
+ }
+ }
+}
+
+#endif // WEBRTC_POSIX
+
+uint32_t SocketDispatcher::GetRequestedEvents() {
+ return enabled_events_;
+}
+
+void SocketDispatcher::OnPreEvent(uint32_t ff) {
+ if ((ff & DE_CONNECT) != 0)
+ state_ = CS_CONNECTED;
+
+#if defined(WEBRTC_WIN)
+ // We set CS_CLOSED from CheckSignalClose.
+#elif defined(WEBRTC_POSIX)
+ if ((ff & DE_CLOSE) != 0)
+ state_ = CS_CLOSED;
+#endif
+}
+
+#if defined(WEBRTC_WIN)
+
+void SocketDispatcher::OnEvent(uint32_t ff, int err) {
+ int cache_id = id_;
+ // Make sure we deliver connect/accept first. Otherwise, consumers may see
+ // something like a READ followed by a CONNECT, which would be odd.
+ if (((ff & DE_CONNECT) != 0) && (id_ == cache_id)) {
+ if (ff != DE_CONNECT)
+ LOG(LS_VERBOSE) << "Signalled with DE_CONNECT: " << ff;
+ enabled_events_ &= ~DE_CONNECT;
+#if !defined(NDEBUG)
+ dbg_addr_ = "Connected @ ";
+ dbg_addr_.append(GetRemoteAddress().ToString());
+#endif
+ SignalConnectEvent(this);
+ }
+ if (((ff & DE_ACCEPT) != 0) && (id_ == cache_id)) {
+ enabled_events_ &= ~DE_ACCEPT;
+ SignalReadEvent(this);
+ }
+ if ((ff & DE_READ) != 0) {
+ enabled_events_ &= ~DE_READ;
+ SignalReadEvent(this);
+ }
+ if (((ff & DE_WRITE) != 0) && (id_ == cache_id)) {
+ enabled_events_ &= ~DE_WRITE;
+ SignalWriteEvent(this);
+ }
+ if (((ff & DE_CLOSE) != 0) && (id_ == cache_id)) {
+ signal_close_ = true;
+ signal_err_ = err;
+ }
+}
+
+#elif defined(WEBRTC_POSIX)
+
+void SocketDispatcher::OnEvent(uint32_t ff, int err) {
+ // Make sure we deliver connect/accept first. Otherwise, consumers may see
+ // something like a READ followed by a CONNECT, which would be odd.
+ if ((ff & DE_CONNECT) != 0) {
+ enabled_events_ &= ~DE_CONNECT;
+ SignalConnectEvent(this);
+ }
+ if ((ff & DE_ACCEPT) != 0) {
+ enabled_events_ &= ~DE_ACCEPT;
+ SignalReadEvent(this);
+ }
+ if ((ff & DE_READ) != 0) {
+ enabled_events_ &= ~DE_READ;
+ SignalReadEvent(this);
+ }
+ if ((ff & DE_WRITE) != 0) {
+ enabled_events_ &= ~DE_WRITE;
+ SignalWriteEvent(this);
+ }
+ if ((ff & DE_CLOSE) != 0) {
+ // The socket is now dead to us, so stop checking it.
+ enabled_events_ = 0;
+ SignalCloseEvent(this, err);
+ }
+}
+
+#endif // WEBRTC_POSIX
+
+int SocketDispatcher::Close() {
+ if (s_ == INVALID_SOCKET)
+ return 0;
+
+#if defined(WEBRTC_WIN)
+ id_ = 0;
+ signal_close_ = false;
+#endif
+ ss_->Remove(this);
+ return PhysicalSocket::Close();
+}
#if defined(WEBRTC_POSIX)
class EventDispatcher : public Dispatcher {
@@ -628,8 +834,8 @@
// Returns true if the given signal number is set.
bool IsSignalSet(int signum) const {
- ASSERT(signum < ARRAY_SIZE(received_signal_));
- if (signum < ARRAY_SIZE(received_signal_)) {
+ ASSERT(signum < static_cast<int>(arraysize(received_signal_)));
+ if (signum < static_cast<int>(arraysize(received_signal_))) {
return received_signal_[signum];
} else {
return false;
@@ -638,8 +844,8 @@
// Clears the given signal number.
void ClearSignal(int signum) {
- ASSERT(signum < ARRAY_SIZE(received_signal_));
- if (signum < ARRAY_SIZE(received_signal_)) {
+ ASSERT(signum < static_cast<int>(arraysize(received_signal_)));
+ if (signum < static_cast<int>(arraysize(received_signal_))) {
received_signal_[signum] = false;
}
}
@@ -654,7 +860,7 @@
// user-level state of the process, since the handler could be executed at any
// time on any thread.
void OnPosixSignalReceived(int signum) {
- if (signum >= ARRAY_SIZE(received_signal_)) {
+ if (signum >= static_cast<int>(arraysize(received_signal_))) {
// We don't have space in our array for this.
return;
}
@@ -790,116 +996,6 @@
PhysicalSocketServer *owner_;
};
-class SocketDispatcher : public Dispatcher, public PhysicalSocket {
- public:
- explicit SocketDispatcher(PhysicalSocketServer *ss) : PhysicalSocket(ss) {
- }
- SocketDispatcher(SOCKET s, PhysicalSocketServer *ss) : PhysicalSocket(ss, s) {
- }
-
- ~SocketDispatcher() override {
- Close();
- }
-
- bool Initialize() {
- ss_->Add(this);
- fcntl(s_, F_SETFL, fcntl(s_, F_GETFL, 0) | O_NONBLOCK);
- return true;
- }
-
- virtual bool Create(int type) {
- return Create(AF_INET, type);
- }
-
- bool Create(int family, int type) override {
- // Change the socket to be non-blocking.
- if (!PhysicalSocket::Create(family, type))
- return false;
-
- return Initialize();
- }
-
- int GetDescriptor() override { return s_; }
-
- bool IsDescriptorClosed() override {
- // We don't have a reliable way of distinguishing end-of-stream
- // from readability. So test on each readable call. Is this
- // inefficient? Probably.
- char ch;
- ssize_t res = ::recv(s_, &ch, 1, MSG_PEEK);
- if (res > 0) {
- // Data available, so not closed.
- return false;
- } else if (res == 0) {
- // EOF, so closed.
- return true;
- } else { // error
- switch (errno) {
- // Returned if we've already closed s_.
- case EBADF:
- // Returned during ungraceful peer shutdown.
- case ECONNRESET:
- return true;
- default:
- // Assume that all other errors are just blocking errors, meaning the
- // connection is still good but we just can't read from it right now.
- // This should only happen when connecting (and at most once), because
- // in all other cases this function is only called if the file
- // descriptor is already known to be in the readable state. However,
- // it's not necessary a problem if we spuriously interpret a
- // "connection lost"-type error as a blocking error, because typically
- // the next recv() will get EOF, so we'll still eventually notice that
- // the socket is closed.
- LOG_ERR(LS_WARNING) << "Assuming benign blocking error";
- return false;
- }
- }
- }
-
- uint32_t GetRequestedEvents() override { return enabled_events_; }
-
- void OnPreEvent(uint32_t ff) override {
- if ((ff & DE_CONNECT) != 0)
- state_ = CS_CONNECTED;
- if ((ff & DE_CLOSE) != 0)
- state_ = CS_CLOSED;
- }
-
- void OnEvent(uint32_t ff, int err) override {
- // Make sure we deliver connect/accept first. Otherwise, consumers may see
- // something like a READ followed by a CONNECT, which would be odd.
- if ((ff & DE_CONNECT) != 0) {
- enabled_events_ &= ~DE_CONNECT;
- SignalConnectEvent(this);
- }
- if ((ff & DE_ACCEPT) != 0) {
- enabled_events_ &= ~DE_ACCEPT;
- SignalReadEvent(this);
- }
- if ((ff & DE_READ) != 0) {
- enabled_events_ &= ~DE_READ;
- SignalReadEvent(this);
- }
- if ((ff & DE_WRITE) != 0) {
- enabled_events_ &= ~DE_WRITE;
- SignalWriteEvent(this);
- }
- if ((ff & DE_CLOSE) != 0) {
- // The socket is now dead to us, so stop checking it.
- enabled_events_ = 0;
- SignalCloseEvent(this, err);
- }
- }
-
- int Close() override {
- if (s_ == INVALID_SOCKET)
- return 0;
-
- ss_->Remove(this);
- return PhysicalSocket::Close();
- }
-};
-
class FileDispatcher: public Dispatcher, public AsyncFile {
public:
FileDispatcher(int fd, PhysicalSocketServer *ss) : ss_(ss), fd_(fd) {
@@ -1013,130 +1109,6 @@
PhysicalSocketServer* ss_;
WSAEVENT hev_;
};
-
-class SocketDispatcher : public Dispatcher, public PhysicalSocket {
- public:
- static int next_id_;
- int id_;
- bool signal_close_;
- int signal_err_;
-
- SocketDispatcher(PhysicalSocketServer* ss)
- : PhysicalSocket(ss),
- id_(0),
- signal_close_(false) {
- }
-
- SocketDispatcher(SOCKET s, PhysicalSocketServer* ss)
- : PhysicalSocket(ss, s),
- id_(0),
- signal_close_(false) {
- }
-
- virtual ~SocketDispatcher() {
- Close();
- }
-
- bool Initialize() {
- ASSERT(s_ != INVALID_SOCKET);
- // Must be a non-blocking
- u_long argp = 1;
- ioctlsocket(s_, FIONBIO, &argp);
- ss_->Add(this);
- return true;
- }
-
- virtual bool Create(int type) {
- return Create(AF_INET, type);
- }
-
- virtual bool Create(int family, int type) {
- // Create socket
- if (!PhysicalSocket::Create(family, type))
- return false;
-
- if (!Initialize())
- return false;
-
- do { id_ = ++next_id_; } while (id_ == 0);
- return true;
- }
-
- virtual int Close() {
- if (s_ == INVALID_SOCKET)
- return 0;
-
- id_ = 0;
- signal_close_ = false;
- ss_->Remove(this);
- return PhysicalSocket::Close();
- }
-
- virtual uint32_t GetRequestedEvents() { return enabled_events_; }
-
- virtual void OnPreEvent(uint32_t ff) {
- if ((ff & DE_CONNECT) != 0)
- state_ = CS_CONNECTED;
- // We set CS_CLOSED from CheckSignalClose.
- }
-
- virtual void OnEvent(uint32_t ff, int err) {
- int cache_id = id_;
- // Make sure we deliver connect/accept first. Otherwise, consumers may see
- // something like a READ followed by a CONNECT, which would be odd.
- if (((ff & DE_CONNECT) != 0) && (id_ == cache_id)) {
- if (ff != DE_CONNECT)
- LOG(LS_VERBOSE) << "Signalled with DE_CONNECT: " << ff;
- enabled_events_ &= ~DE_CONNECT;
-#ifdef _DEBUG
- dbg_addr_ = "Connected @ ";
- dbg_addr_.append(GetRemoteAddress().ToString());
-#endif // _DEBUG
- SignalConnectEvent(this);
- }
- if (((ff & DE_ACCEPT) != 0) && (id_ == cache_id)) {
- enabled_events_ &= ~DE_ACCEPT;
- SignalReadEvent(this);
- }
- if ((ff & DE_READ) != 0) {
- enabled_events_ &= ~DE_READ;
- SignalReadEvent(this);
- }
- if (((ff & DE_WRITE) != 0) && (id_ == cache_id)) {
- enabled_events_ &= ~DE_WRITE;
- SignalWriteEvent(this);
- }
- if (((ff & DE_CLOSE) != 0) && (id_ == cache_id)) {
- signal_close_ = true;
- signal_err_ = err;
- }
- }
-
- virtual WSAEVENT GetWSAEvent() {
- return WSA_INVALID_EVENT;
- }
-
- virtual SOCKET GetSocket() {
- return s_;
- }
-
- virtual bool CheckSignalClose() {
- if (!signal_close_)
- return false;
-
- char ch;
- if (recv(s_, &ch, 1, MSG_PEEK) > 0)
- return false;
-
- state_ = CS_CLOSED;
- signal_close_ = false;
- SignalCloseEvent(this, signal_err_);
- return true;
- }
-};
-
-int SocketDispatcher::next_id_ = 0;
-
#endif // WEBRTC_WIN
// Sets the value of a boolean value to false when signaled.
@@ -1189,7 +1161,7 @@
return socket;
} else {
delete socket;
- return 0;
+ return nullptr;
}
}
@@ -1203,7 +1175,7 @@
return dispatcher;
} else {
delete dispatcher;
- return 0;
+ return nullptr;
}
}
@@ -1213,7 +1185,7 @@
return dispatcher;
} else {
delete dispatcher;
- return 0;
+ return nullptr;
}
}
@@ -1342,7 +1314,7 @@
int errcode = 0;
// Reap any error code, which can be signaled through reads or writes.
- // TODO: Should we set errcode if getsockopt fails?
+ // TODO(pthatcher): Should we set errcode if getsockopt fails?
if (FD_ISSET(fd, &fdsRead) || FD_ISSET(fd, &fdsWrite)) {
socklen_t len = sizeof(errcode);
::getsockopt(fd, SOL_SOCKET, SO_ERROR, &errcode, &len);
@@ -1351,7 +1323,7 @@
// Check readable descriptors. If we're waiting on an accept, signal
// that. Otherwise we're waiting for data, check to see if we're
// readable or really closed.
- // TODO: Only peek at TCP descriptors.
+ // TODO(pthatcher): Only peek at TCP descriptors.
if (FD_ISSET(fd, &fdsRead)) {
FD_CLR(fd, &fdsRead);
if (pdispatcher->GetRequestedEvents() & DE_ACCEPT) {
@@ -1525,7 +1497,7 @@
if (dw == WSA_WAIT_FAILED) {
// Failed?
- // TODO: need a better strategy than this!
+ // TODO(pthatcher): need a better strategy than this!
WSAGetLastError();
ASSERT(false);
return false;
diff --git a/webrtc/base/physicalsocketserver.h b/webrtc/base/physicalsocketserver.h
index af09e0b..ae1f10f 100644
--- a/webrtc/base/physicalsocketserver.h
+++ b/webrtc/base/physicalsocketserver.h
@@ -14,6 +14,7 @@
#include <vector>
#include "webrtc/base/asyncfile.h"
+#include "webrtc/base/nethelpers.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/socketserver.h"
#include "webrtc/base/criticalsection.h"
@@ -115,6 +116,107 @@
#endif
};
+class PhysicalSocket : public AsyncSocket, public sigslot::has_slots<> {
+ public:
+ PhysicalSocket(PhysicalSocketServer* ss, SOCKET s = INVALID_SOCKET);
+ ~PhysicalSocket() override;
+
+ // Creates the underlying OS socket (same as the "socket" function).
+ virtual bool Create(int family, int type);
+
+ SocketAddress GetLocalAddress() const override;
+ SocketAddress GetRemoteAddress() const override;
+
+ int Bind(const SocketAddress& bind_addr) override;
+ int Connect(const SocketAddress& addr) override;
+
+ int GetError() const override;
+ void SetError(int error) override;
+
+ ConnState GetState() const override;
+
+ int GetOption(Option opt, int* value) override;
+ int SetOption(Option opt, int value) override;
+
+ int Send(const void* pv, size_t cb) override;
+ int SendTo(const void* buffer,
+ size_t length,
+ const SocketAddress& addr) override;
+
+ int Recv(void* buffer, size_t length) override;
+ int RecvFrom(void* buffer, size_t length, SocketAddress* out_addr) override;
+
+ int Listen(int backlog) override;
+ AsyncSocket* Accept(SocketAddress* out_addr) override;
+
+ int Close() override;
+
+ int EstimateMTU(uint16_t* mtu) override;
+
+ SocketServer* socketserver() { return ss_; }
+
+ protected:
+ int DoConnect(const SocketAddress& connect_addr);
+
+ // Make virtual so ::accept can be overwritten in tests.
+ virtual SOCKET DoAccept(SOCKET socket, sockaddr* addr, socklen_t* addrlen);
+
+ void OnResolveResult(AsyncResolverInterface* resolver);
+
+ void UpdateLastError();
+ void MaybeRemapSendError();
+
+ static int TranslateOption(Option opt, int* slevel, int* sopt);
+
+ PhysicalSocketServer* ss_;
+ SOCKET s_;
+ uint8_t enabled_events_;
+ bool udp_;
+ mutable CriticalSection crit_;
+ int error_ GUARDED_BY(crit_);
+ ConnState state_;
+ AsyncResolver* resolver_;
+
+#if !defined(NDEBUG)
+ std::string dbg_addr_;
+#endif
+};
+
+class SocketDispatcher : public Dispatcher, public PhysicalSocket {
+ public:
+ explicit SocketDispatcher(PhysicalSocketServer *ss);
+ SocketDispatcher(SOCKET s, PhysicalSocketServer *ss);
+ ~SocketDispatcher() override;
+
+ bool Initialize();
+
+ virtual bool Create(int type);
+ bool Create(int family, int type) override;
+
+#if defined(WEBRTC_WIN)
+ WSAEVENT GetWSAEvent() override;
+ SOCKET GetSocket() override;
+ bool CheckSignalClose() override;
+#elif defined(WEBRTC_POSIX)
+ int GetDescriptor() override;
+ bool IsDescriptorClosed() override;
+#endif
+
+ uint32_t GetRequestedEvents() override;
+ void OnPreEvent(uint32_t ff) override;
+ void OnEvent(uint32_t ff, int err) override;
+
+ int Close() override;
+
+#if defined(WEBRTC_WIN)
+ private:
+ static int next_id_;
+ int id_;
+ bool signal_close_;
+ int signal_err_;
+#endif // WEBRTC_WIN
+};
+
} // namespace rtc
#endif // WEBRTC_BASE_PHYSICALSOCKETSERVER_H__
diff --git a/webrtc/base/physicalsocketserver_unittest.cc b/webrtc/base/physicalsocketserver_unittest.cc
index 6c7be9f..a2fde80 100644
--- a/webrtc/base/physicalsocketserver_unittest.cc
+++ b/webrtc/base/physicalsocketserver_unittest.cc
@@ -18,13 +18,85 @@
#include "webrtc/base/socket_unittest.h"
#include "webrtc/base/testutils.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
-class PhysicalSocketTest : public SocketTest {
+class PhysicalSocketTest;
+
+class FakeSocketDispatcher : public SocketDispatcher {
+ public:
+ explicit FakeSocketDispatcher(PhysicalSocketServer* ss)
+ : SocketDispatcher(ss) {
+ }
+
+ protected:
+ SOCKET DoAccept(SOCKET socket, sockaddr* addr, socklen_t* addrlen) override;
};
+class FakePhysicalSocketServer : public PhysicalSocketServer {
+ public:
+ explicit FakePhysicalSocketServer(PhysicalSocketTest* test)
+ : test_(test) {
+ }
+
+ AsyncSocket* CreateAsyncSocket(int type) override {
+ SocketDispatcher* dispatcher = new FakeSocketDispatcher(this);
+ if (dispatcher->Create(type)) {
+ return dispatcher;
+ } else {
+ delete dispatcher;
+ return nullptr;
+ }
+ }
+
+ AsyncSocket* CreateAsyncSocket(int family, int type) override {
+ SocketDispatcher* dispatcher = new FakeSocketDispatcher(this);
+ if (dispatcher->Create(family, type)) {
+ return dispatcher;
+ } else {
+ delete dispatcher;
+ return nullptr;
+ }
+ }
+
+ PhysicalSocketTest* GetTest() const { return test_; }
+
+ private:
+ PhysicalSocketTest* test_;
+};
+
+class PhysicalSocketTest : public SocketTest {
+ public:
+ // Set flag to simluate failures when calling "::accept" on a AsyncSocket.
+ void SetFailAccept(bool fail) { fail_accept_ = fail; }
+ bool FailAccept() const { return fail_accept_; }
+
+ protected:
+ PhysicalSocketTest()
+ : server_(new FakePhysicalSocketServer(this)),
+ scope_(server_.get()),
+ fail_accept_(false) {
+ }
+
+ void ConnectInternalAcceptError(const IPAddress& loopback);
+
+ rtc::scoped_ptr<FakePhysicalSocketServer> server_;
+ SocketServerScope scope_;
+ bool fail_accept_;
+};
+
+SOCKET FakeSocketDispatcher::DoAccept(SOCKET socket,
+ sockaddr* addr,
+ socklen_t* addrlen) {
+ FakePhysicalSocketServer* ss =
+ static_cast<FakePhysicalSocketServer*>(socketserver());
+ if (ss->GetTest()->FailAccept()) {
+ return INVALID_SOCKET;
+ }
+
+ return SocketDispatcher::DoAccept(socket, addr, addrlen);
+}
+
TEST_F(PhysicalSocketTest, TestConnectIPv4) {
SocketTest::TestConnectIPv4();
}
@@ -51,6 +123,92 @@
SocketTest::TestConnectFailIPv4();
}
+void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) {
+ testing::StreamSink sink;
+ SocketAddress accept_addr;
+
+ // Create two clients.
+ scoped_ptr<AsyncSocket> client1(server_->CreateAsyncSocket(loopback.family(),
+ SOCK_STREAM));
+ sink.Monitor(client1.get());
+ EXPECT_EQ(AsyncSocket::CS_CLOSED, client1->GetState());
+ EXPECT_PRED1(IsUnspecOrEmptyIP, client1->GetLocalAddress().ipaddr());
+
+ scoped_ptr<AsyncSocket> client2(server_->CreateAsyncSocket(loopback.family(),
+ SOCK_STREAM));
+ sink.Monitor(client2.get());
+ EXPECT_EQ(AsyncSocket::CS_CLOSED, client2->GetState());
+ EXPECT_PRED1(IsUnspecOrEmptyIP, client2->GetLocalAddress().ipaddr());
+
+ // Create server and listen.
+ scoped_ptr<AsyncSocket> server(
+ server_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
+ sink.Monitor(server.get());
+ EXPECT_EQ(0, server->Bind(SocketAddress(loopback, 0)));
+ EXPECT_EQ(0, server->Listen(5));
+ EXPECT_EQ(AsyncSocket::CS_CONNECTING, server->GetState());
+
+ // Ensure no pending server connections, since we haven't done anything yet.
+ EXPECT_FALSE(sink.Check(server.get(), testing::SSE_READ));
+ EXPECT_TRUE(nullptr == server->Accept(&accept_addr));
+ EXPECT_TRUE(accept_addr.IsNil());
+
+ // Attempt first connect to listening socket.
+ EXPECT_EQ(0, client1->Connect(server->GetLocalAddress()));
+ EXPECT_FALSE(client1->GetLocalAddress().IsNil());
+ EXPECT_NE(server->GetLocalAddress(), client1->GetLocalAddress());
+
+ // Client is connecting, outcome not yet determined.
+ EXPECT_EQ(AsyncSocket::CS_CONNECTING, client1->GetState());
+ EXPECT_FALSE(sink.Check(client1.get(), testing::SSE_OPEN));
+ EXPECT_FALSE(sink.Check(client1.get(), testing::SSE_CLOSE));
+
+ // Server has pending connection, try to accept it (will fail).
+ EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
+ // Simulate "::accept" returning an error.
+ SetFailAccept(true);
+ scoped_ptr<AsyncSocket> accepted(server->Accept(&accept_addr));
+ EXPECT_FALSE(accepted);
+ ASSERT_TRUE(accept_addr.IsNil());
+
+ // Ensure no more pending server connections.
+ EXPECT_FALSE(sink.Check(server.get(), testing::SSE_READ));
+ EXPECT_TRUE(nullptr == server->Accept(&accept_addr));
+ EXPECT_TRUE(accept_addr.IsNil());
+
+ // Attempt second connect to listening socket.
+ EXPECT_EQ(0, client2->Connect(server->GetLocalAddress()));
+ EXPECT_FALSE(client2->GetLocalAddress().IsNil());
+ EXPECT_NE(server->GetLocalAddress(), client2->GetLocalAddress());
+
+ // Client is connecting, outcome not yet determined.
+ EXPECT_EQ(AsyncSocket::CS_CONNECTING, client2->GetState());
+ EXPECT_FALSE(sink.Check(client2.get(), testing::SSE_OPEN));
+ EXPECT_FALSE(sink.Check(client2.get(), testing::SSE_CLOSE));
+
+ // Server has pending connection, try to accept it (will succeed).
+ EXPECT_TRUE_WAIT((sink.Check(server.get(), testing::SSE_READ)), kTimeout);
+ SetFailAccept(false);
+ scoped_ptr<AsyncSocket> accepted2(server->Accept(&accept_addr));
+ ASSERT_TRUE(accepted2);
+ EXPECT_FALSE(accept_addr.IsNil());
+ EXPECT_EQ(accepted2->GetRemoteAddress(), accept_addr);
+}
+
+TEST_F(PhysicalSocketTest, TestConnectAcceptErrorIPv4) {
+ ConnectInternalAcceptError(kIPv4Loopback);
+}
+
+// Crashes on Linux. See webrtc:4923.
+#if defined(WEBRTC_LINUX)
+#define MAYBE_TestConnectAcceptErrorIPv6 DISABLED_TestConnectAcceptErrorIPv6
+#else
+#define MAYBE_TestConnectAcceptErrorIPv6 TestConnectAcceptErrorIPv6
+#endif
+TEST_F(PhysicalSocketTest, MAYBE_TestConnectAcceptErrorIPv6) {
+ ConnectInternalAcceptError(kIPv6Loopback);
+}
+
// Crashes on Linux. See webrtc:4923.
#if defined(WEBRTC_LINUX)
#define MAYBE_TestConnectFailIPv6 DISABLED_TestConnectFailIPv6
@@ -215,8 +373,11 @@
// https://code.google.com/p/webrtc/issues/detail?id=4958
// TODO(deadbeef): Enable again once test is reimplemented to be unflaky.
// Also disable for ASan.
+// Disabled on Android: https://code.google.com/p/webrtc/issues/detail?id=4364
+// Disabled on Linux: https://bugs.chromium.org/p/webrtc/issues/detail?id=5233
#if defined(THREAD_SANITIZER) || defined(MEMORY_SANITIZER) || \
- defined(ADDRESS_SANITIZER)
+ defined(ADDRESS_SANITIZER) || defined(WEBRTC_ANDROID) || \
+ defined(WEBRTC_LINUX)
#define MAYBE_TestUdpReadyToSendIPv4 DISABLED_TestUdpReadyToSendIPv4
#else
#define MAYBE_TestUdpReadyToSendIPv4 TestUdpReadyToSendIPv4
diff --git a/webrtc/base/platform_thread.cc b/webrtc/base/platform_thread.cc
index 4167392..05b7a25 100644
--- a/webrtc/base/platform_thread.cc
+++ b/webrtc/base/platform_thread.cc
@@ -10,8 +10,6 @@
#include "webrtc/base/platform_thread.h"
-#include <string.h>
-
#include "webrtc/base/checks.h"
#if defined(WEBRTC_LINUX)
@@ -58,7 +56,6 @@
}
void SetCurrentThreadName(const char* name) {
- RTC_DCHECK(strlen(name) < 64);
#if defined(WEBRTC_WIN)
struct {
DWORD dwType;
@@ -79,4 +76,175 @@
#endif
}
+namespace {
+#if defined(WEBRTC_WIN)
+void CALLBACK RaiseFlag(ULONG_PTR param) {
+ *reinterpret_cast<bool*>(param) = true;
+}
+#else
+struct ThreadAttributes {
+ ThreadAttributes() { pthread_attr_init(&attr); }
+ ~ThreadAttributes() { pthread_attr_destroy(&attr); }
+ pthread_attr_t* operator&() { return &attr; }
+ pthread_attr_t attr;
+};
+#endif // defined(WEBRTC_WIN)
+}
+
+PlatformThread::PlatformThread(ThreadRunFunction func,
+ void* obj,
+ const char* thread_name)
+ : run_function_(func),
+ obj_(obj),
+ name_(thread_name ? thread_name : "webrtc"),
+#if defined(WEBRTC_WIN)
+ stop_(false),
+ thread_(NULL) {
+#else
+ stop_event_(false, false),
+ thread_(0) {
+#endif // defined(WEBRTC_WIN)
+ RTC_DCHECK(func);
+ RTC_DCHECK(name_.length() < 64);
+}
+
+PlatformThread::~PlatformThread() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+#if defined(WEBRTC_WIN)
+ RTC_DCHECK(!thread_);
+#endif // defined(WEBRTC_WIN)
+}
+
+#if defined(WEBRTC_WIN)
+DWORD WINAPI PlatformThread::StartThread(void* param) {
+ static_cast<PlatformThread*>(param)->Run();
+ return 0;
+}
+#else
+void* PlatformThread::StartThread(void* param) {
+ static_cast<PlatformThread*>(param)->Run();
+ return 0;
+}
+#endif // defined(WEBRTC_WIN)
+
+void PlatformThread::Start() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(!thread_) << "Thread already started?";
+#if defined(WEBRTC_WIN)
+ stop_ = false;
+
+ // See bug 2902 for background on STACK_SIZE_PARAM_IS_A_RESERVATION.
+ // Set the reserved stack stack size to 1M, which is the default on Windows
+ // and Linux.
+ DWORD thread_id;
+ thread_ = ::CreateThread(NULL, 1024 * 1024, &StartThread, this,
+ STACK_SIZE_PARAM_IS_A_RESERVATION, &thread_id);
+ RTC_CHECK(thread_) << "CreateThread failed";
+#else
+ ThreadAttributes attr;
+ // Set the stack stack size to 1M.
+ pthread_attr_setstacksize(&attr, 1024 * 1024);
+ RTC_CHECK_EQ(0, pthread_create(&thread_, &attr, &StartThread, this));
+#endif // defined(WEBRTC_WIN)
+}
+
+bool PlatformThread::IsRunning() const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+#if defined(WEBRTC_WIN)
+ return thread_ != nullptr;
+#else
+ return thread_ != 0;
+#endif // defined(WEBRTC_WIN)
+}
+
+void PlatformThread::Stop() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (!IsRunning())
+ return;
+
+#if defined(WEBRTC_WIN)
+ // Set stop_ to |true| on the worker thread.
+ QueueUserAPC(&RaiseFlag, thread_, reinterpret_cast<ULONG_PTR>(&stop_));
+ WaitForSingleObject(thread_, INFINITE);
+ CloseHandle(thread_);
+ thread_ = nullptr;
+#else
+ stop_event_.Set();
+ RTC_CHECK_EQ(0, pthread_join(thread_, nullptr));
+ thread_ = 0;
+#endif // defined(WEBRTC_WIN)
+}
+
+void PlatformThread::Run() {
+ if (!name_.empty())
+ rtc::SetCurrentThreadName(name_.c_str());
+ do {
+ // The interface contract of Start/Stop is that for a successfull call to
+ // Start, there should be at least one call to the run function. So we
+ // call the function before checking |stop_|.
+ if (!run_function_(obj_))
+ break;
+#if defined(WEBRTC_WIN)
+ // Alertable sleep to permit RaiseFlag to run and update |stop_|.
+ SleepEx(0, true);
+ } while (!stop_);
+#else
+ } while (!stop_event_.Wait(0));
+#endif // defined(WEBRTC_WIN)
+}
+
+bool PlatformThread::SetPriority(ThreadPriority priority) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK(IsRunning());
+#if defined(WEBRTC_WIN)
+ return SetThreadPriority(thread_, priority) != FALSE;
+#elif defined(__native_client__)
+ // Setting thread priorities is not supported in NaCl.
+ return true;
+#elif defined(WEBRTC_CHROMIUM_BUILD) && defined(WEBRTC_LINUX)
+ // TODO(tommi): Switch to the same mechanism as Chromium uses for changing
+ // thread priorities.
+ return true;
+#else
+#ifdef WEBRTC_THREAD_RR
+ const int policy = SCHED_RR;
+#else
+ const int policy = SCHED_FIFO;
+#endif
+ const int min_prio = sched_get_priority_min(policy);
+ const int max_prio = sched_get_priority_max(policy);
+ if (min_prio == -1 || max_prio == -1) {
+ return false;
+ }
+
+ if (max_prio - min_prio <= 2)
+ return false;
+
+ // Convert webrtc priority to system priorities:
+ sched_param param;
+ const int top_prio = max_prio - 1;
+ const int low_prio = min_prio + 1;
+ switch (priority) {
+ case kLowPriority:
+ param.sched_priority = low_prio;
+ break;
+ case kNormalPriority:
+ // The -1 ensures that the kHighPriority is always greater or equal to
+ // kNormalPriority.
+ param.sched_priority = (low_prio + top_prio - 1) / 2;
+ break;
+ case kHighPriority:
+ param.sched_priority = std::max(top_prio - 2, low_prio);
+ break;
+ case kHighestPriority:
+ param.sched_priority = std::max(top_prio - 1, low_prio);
+ break;
+ case kRealtimePriority:
+ param.sched_priority = top_prio;
+ break;
+ }
+ return pthread_setschedparam(thread_, policy, ¶m) == 0;
+#endif // defined(WEBRTC_WIN)
+}
+
} // namespace rtc
diff --git a/webrtc/base/platform_thread.h b/webrtc/base/platform_thread.h
index 50033b3..53465e4 100644
--- a/webrtc/base/platform_thread.h
+++ b/webrtc/base/platform_thread.h
@@ -11,24 +11,16 @@
#ifndef WEBRTC_BASE_PLATFORM_THREAD_H_
#define WEBRTC_BASE_PLATFORM_THREAD_H_
-#if defined(WEBRTC_WIN)
-#include <winsock2.h>
-#include <windows.h>
-#elif defined(WEBRTC_POSIX)
-#include <pthread.h>
-#include <unistd.h>
-#endif
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/platform_thread_types.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
namespace rtc {
-#if defined(WEBRTC_WIN)
-typedef DWORD PlatformThreadId;
-typedef DWORD PlatformThreadRef;
-#elif defined(WEBRTC_POSIX)
-typedef pid_t PlatformThreadId;
-typedef pthread_t PlatformThreadRef;
-#endif
-
PlatformThreadId CurrentThreadId();
PlatformThreadRef CurrentThreadRef();
@@ -38,6 +30,71 @@
// Sets the current thread name.
void SetCurrentThreadName(const char* name);
+// Callback function that the spawned thread will enter once spawned.
+// A return value of false is interpreted as that the function has no
+// more work to do and that the thread can be released.
+typedef bool (*ThreadRunFunction)(void*);
+
+enum ThreadPriority {
+#ifdef WEBRTC_WIN
+ kLowPriority = THREAD_PRIORITY_BELOW_NORMAL,
+ kNormalPriority = THREAD_PRIORITY_NORMAL,
+ kHighPriority = THREAD_PRIORITY_ABOVE_NORMAL,
+ kHighestPriority = THREAD_PRIORITY_HIGHEST,
+ kRealtimePriority = THREAD_PRIORITY_TIME_CRITICAL
+#else
+ kLowPriority = 1,
+ kNormalPriority = 2,
+ kHighPriority = 3,
+ kHighestPriority = 4,
+ kRealtimePriority = 5
+#endif
+};
+
+// Represents a simple worker thread. The implementation must be assumed
+// to be single threaded, meaning that all methods of the class, must be
+// called from the same thread, including instantiation.
+class PlatformThread {
+ public:
+ PlatformThread(ThreadRunFunction func, void* obj, const char* thread_name);
+ virtual ~PlatformThread();
+
+ // Spawns a thread and tries to set thread priority according to the priority
+ // from when CreateThread was called.
+ void Start();
+
+ bool IsRunning() const;
+
+ // Stops (joins) the spawned thread.
+ void Stop();
+
+ // Set the priority of the thread. Must be called when thread is running.
+ bool SetPriority(ThreadPriority priority);
+
+ private:
+ void Run();
+
+ ThreadRunFunction const run_function_;
+ void* const obj_;
+ // TODO(pbos): Make sure call sites use string literals and update to a const
+ // char* instead of a std::string.
+ const std::string name_;
+ rtc::ThreadChecker thread_checker_;
+#if defined(WEBRTC_WIN)
+ static DWORD WINAPI StartThread(void* param);
+
+ bool stop_;
+ HANDLE thread_;
+#else
+ static void* StartThread(void* param);
+
+ rtc::Event stop_event_;
+
+ pthread_t thread_;
+#endif // defined(WEBRTC_WIN)
+ RTC_DISALLOW_COPY_AND_ASSIGN(PlatformThread);
+};
+
} // namespace rtc
#endif // WEBRTC_BASE_PLATFORM_THREAD_H_
diff --git a/webrtc/base/platform_thread_types.h b/webrtc/base/platform_thread_types.h
new file mode 100644
index 0000000..546fffd
--- /dev/null
+++ b/webrtc/base/platform_thread_types.h
@@ -0,0 +1,32 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_
+#define WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_
+
+#if defined(WEBRTC_WIN)
+#include <winsock2.h>
+#include <windows.h>
+#elif defined(WEBRTC_POSIX)
+#include <pthread.h>
+#include <unistd.h>
+#endif
+
+namespace rtc {
+#if defined(WEBRTC_WIN)
+typedef DWORD PlatformThreadId;
+typedef DWORD PlatformThreadRef;
+#elif defined(WEBRTC_POSIX)
+typedef pid_t PlatformThreadId;
+typedef pthread_t PlatformThreadRef;
+#endif
+} // namespace rtc
+
+#endif // WEBRTC_BASE_PLATFORM_THREAD_TYPES_H_
diff --git a/webrtc/base/platform_thread_unittest.cc b/webrtc/base/platform_thread_unittest.cc
new file mode 100644
index 0000000..f9db8e3
--- /dev/null
+++ b/webrtc/base/platform_thread_unittest.cc
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/base/platform_thread.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+
+namespace webrtc {
+
+// Function that does nothing, and reports success.
+bool NullRunFunction(void* obj) {
+ SleepMs(0); // Hand over timeslice, prevents busy looping.
+ return true;
+}
+
+TEST(PlatformThreadTest, StartStop) {
+ rtc::PlatformThread thread(&NullRunFunction, nullptr, "PlatformThreadTest");
+ thread.Start();
+ thread.Stop();
+}
+
+// Function that sets a boolean.
+bool SetFlagRunFunction(void* obj) {
+ bool* obj_as_bool = static_cast<bool*>(obj);
+ *obj_as_bool = true;
+ SleepMs(0); // Hand over timeslice, prevents busy looping.
+ return true;
+}
+
+TEST(PlatformThreadTest, RunFunctionIsCalled) {
+ bool flag = false;
+ rtc::PlatformThread thread(&SetFlagRunFunction, &flag, "RunFunctionIsCalled");
+ thread.Start();
+
+ // At this point, the flag may be either true or false.
+ thread.Stop();
+
+ // We expect the thread to have run at least once.
+ EXPECT_TRUE(flag);
+}
+
+} // namespace webrtc
diff --git a/webrtc/base/proxy_unittest.cc b/webrtc/base/proxy_unittest.cc
index 03dc154..d8a523f 100644
--- a/webrtc/base/proxy_unittest.cc
+++ b/webrtc/base/proxy_unittest.cc
@@ -17,7 +17,6 @@
#include "webrtc/base/testclient.h"
#include "webrtc/base/testechoserver.h"
#include "webrtc/base/virtualsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using rtc::Socket;
using rtc::Thread;
diff --git a/webrtc/base/proxydetect.cc b/webrtc/base/proxydetect.cc
index b144d20..30959ca 100644
--- a/webrtc/base/proxydetect.cc
+++ b/webrtc/base/proxydetect.cc
@@ -34,6 +34,7 @@
#include <map>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/httpcommon.h"
#include "webrtc/base/httpcommon-inl.h"
@@ -222,7 +223,7 @@
uint32_t mask = (m == 0) ? 0 : (~0UL) << (32 - m);
SocketAddress addr(url.host(), 0);
// TODO: Support IPv6 proxyitems. This code block is IPv4 only anyway.
- return !addr.IsUnresolved() &&
+ return !addr.IsUnresolvedIP() &&
((addr.ipaddr().v4AddressAsHostOrderInteger() & mask) == (ip & mask));
}
@@ -398,7 +399,7 @@
}
char buffer[NAME_MAX + 1];
if (0 != FSRefMakePath(&fr, reinterpret_cast<uint8_t*>(buffer),
- ARRAY_SIZE(buffer))) {
+ arraysize(buffer))) {
LOG(LS_ERROR) << "FSRefMakePath failed";
return false;
}
diff --git a/webrtc/base/random.cc b/webrtc/base/random.cc
new file mode 100644
index 0000000..14a9faf
--- /dev/null
+++ b/webrtc/base/random.cc
@@ -0,0 +1,86 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+#include "webrtc/base/random.h"
+
+#include <math.h>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+Random::Random(uint64_t seed) {
+ RTC_DCHECK(seed != 0x0ull);
+ state_ = seed;
+}
+
+uint32_t Random::Rand(uint32_t t) {
+ // Casting the output to 32 bits will give an almost uniform number.
+ // Pr[x=0] = (2^32-1) / (2^64-1)
+ // Pr[x=k] = 2^32 / (2^64-1) for k!=0
+ // Uniform would be Pr[x=k] = 2^32 / 2^64 for all 32-bit integers k.
+ uint32_t x = NextOutput();
+ // If x / 2^32 is uniform on [0,1), then x / 2^32 * (t+1) is uniform on
+ // the interval [0,t+1), so the integer part is uniform on [0,t].
+ uint64_t result = x * (static_cast<uint64_t>(t) + 1);
+ result >>= 32;
+ return result;
+}
+
+uint32_t Random::Rand(uint32_t low, uint32_t high) {
+ RTC_DCHECK(low <= high);
+ return Rand(high - low) + low;
+}
+
+int32_t Random::Rand(int32_t low, int32_t high) {
+ RTC_DCHECK(low <= high);
+ // We rely on subtraction (and addition) to be the same for signed and
+ // unsigned numbers in two-complement representation. Thus, although
+ // high - low might be negative as an int, it is the correct difference
+ // when interpreted as an unsigned.
+ return Rand(high - low) + low;
+}
+
+template <>
+float Random::Rand<float>() {
+ double result = NextOutput() - 1;
+ result = result / 0xFFFFFFFFFFFFFFFEull;
+ return static_cast<float>(result);
+}
+
+template <>
+double Random::Rand<double>() {
+ double result = NextOutput() - 1;
+ result = result / 0xFFFFFFFFFFFFFFFEull;
+ return result;
+}
+
+template <>
+bool Random::Rand<bool>() {
+ return Rand(0, 1) == 1;
+}
+
+double Random::Gaussian(double mean, double standard_deviation) {
+ // Creating a Normal distribution variable from two independent uniform
+ // variables based on the Box-Muller transform, which is defined on the
+ // interval (0, 1]. Note that we rely on NextOutput to generate integers
+ // in the range [1, 2^64-1]. Normally this behavior is a bit frustrating,
+ // but here it is exactly what we need.
+ const double kPi = 3.14159265358979323846;
+ double u1 = static_cast<double>(NextOutput()) / 0xFFFFFFFFFFFFFFFFull;
+ double u2 = static_cast<double>(NextOutput()) / 0xFFFFFFFFFFFFFFFFull;
+ return mean + standard_deviation * sqrt(-2 * log(u1)) * cos(2 * kPi * u2);
+}
+
+double Random::Exponential(double lambda) {
+ double uniform = Rand<double>();
+ return -log(uniform) / lambda;
+}
+
+} // namespace webrtc
diff --git a/webrtc/base/random.h b/webrtc/base/random.h
new file mode 100644
index 0000000..647b84c
--- /dev/null
+++ b/webrtc/base/random.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_BASE_RANDOM_H_
+#define WEBRTC_BASE_RANDOM_H_
+
+#include <limits>
+
+#include "webrtc/typedefs.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+class Random {
+ public:
+ explicit Random(uint64_t seed);
+
+ // Return pseudo-random integer of the specified type.
+ // We need to limit the size to 32 bits to keep the output close to uniform.
+ template <typename T>
+ T Rand() {
+ static_assert(std::numeric_limits<T>::is_integer &&
+ std::numeric_limits<T>::radix == 2 &&
+ std::numeric_limits<T>::digits <= 32,
+ "Rand is only supported for built-in integer types that are "
+ "32 bits or smaller.");
+ return static_cast<T>(NextOutput());
+ }
+
+ // Uniformly distributed pseudo-random number in the interval [0, t].
+ uint32_t Rand(uint32_t t);
+
+ // Uniformly distributed pseudo-random number in the interval [low, high].
+ uint32_t Rand(uint32_t low, uint32_t high);
+
+ // Uniformly distributed pseudo-random number in the interval [low, high].
+ int32_t Rand(int32_t low, int32_t high);
+
+ // Normal Distribution.
+ double Gaussian(double mean, double standard_deviation);
+
+ // Exponential Distribution.
+ double Exponential(double lambda);
+
+ private:
+ // Outputs a nonzero 64-bit random number.
+ uint64_t NextOutput() {
+ state_ ^= state_ >> 12;
+ state_ ^= state_ << 25;
+ state_ ^= state_ >> 27;
+ RTC_DCHECK(state_ != 0x0ULL);
+ return state_ * 2685821657736338717ull;
+ }
+
+ uint64_t state_;
+
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Random);
+};
+
+// Return pseudo-random number in the interval [0.0, 1.0).
+template <>
+float Random::Rand<float>();
+
+// Return pseudo-random number in the interval [0.0, 1.0).
+template <>
+double Random::Rand<double>();
+
+// Return pseudo-random boolean value.
+template <>
+bool Random::Rand<bool>();
+
+} // namespace webrtc
+
+#endif // WEBRTC_BASE_RANDOM_H_
diff --git a/webrtc/base/random_unittest.cc b/webrtc/base/random_unittest.cc
new file mode 100644
index 0000000..febae1c
--- /dev/null
+++ b/webrtc/base/random_unittest.cc
@@ -0,0 +1,302 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <math.h>
+
+#include <limits>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/random.h"
+
+namespace webrtc {
+
+namespace {
+// Computes the positive remainder of x/n.
+template <typename T>
+T fdiv_remainder(T x, T n) {
+ RTC_CHECK_GE(n, static_cast<T>(0));
+ T remainder = x % n;
+ if (remainder < 0)
+ remainder += n;
+ return remainder;
+}
+} // namespace
+
+// Sample a number of random integers of type T. Divide them into buckets
+// based on the remainder when dividing by bucket_count and check that each
+// bucket gets roughly the expected number of elements.
+template <typename T>
+void UniformBucketTest(T bucket_count, int samples, Random* prng) {
+ std::vector<int> buckets(bucket_count, 0);
+
+ uint64_t total_values = 1ull << (std::numeric_limits<T>::digits +
+ std::numeric_limits<T>::is_signed);
+ T upper_limit =
+ std::numeric_limits<T>::max() -
+ static_cast<T>(total_values % static_cast<uint64_t>(bucket_count));
+ ASSERT_GT(upper_limit, std::numeric_limits<T>::max() / 2);
+
+ for (int i = 0; i < samples; i++) {
+ T sample;
+ do {
+ // We exclude a few numbers from the range so that it is divisible by
+ // the number of buckets. If we are unlucky and hit one of the excluded
+ // numbers we just resample. Note that if the number of buckets is a
+ // power of 2, then we don't have to exclude anything.
+ sample = prng->Rand<T>();
+ } while (sample > upper_limit);
+ buckets[fdiv_remainder(sample, bucket_count)]++;
+ }
+
+ for (T i = 0; i < bucket_count; i++) {
+ // Expect the result to be within 3 standard deviations of the mean.
+ EXPECT_NEAR(buckets[i], samples / bucket_count,
+ 3 * sqrt(samples / bucket_count));
+ }
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestSignedChar) {
+ Random prng(7297352569824ull);
+ UniformBucketTest<signed char>(64, 640000, &prng);
+ UniformBucketTest<signed char>(11, 440000, &prng);
+ UniformBucketTest<signed char>(3, 270000, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestUnsignedChar) {
+ Random prng(7297352569824ull);
+ UniformBucketTest<unsigned char>(64, 640000, &prng);
+ UniformBucketTest<unsigned char>(11, 440000, &prng);
+ UniformBucketTest<unsigned char>(3, 270000, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestSignedShort) {
+ Random prng(7297352569824ull);
+ UniformBucketTest<int16_t>(64, 640000, &prng);
+ UniformBucketTest<int16_t>(11, 440000, &prng);
+ UniformBucketTest<int16_t>(3, 270000, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestUnsignedShort) {
+ Random prng(7297352569824ull);
+ UniformBucketTest<uint16_t>(64, 640000, &prng);
+ UniformBucketTest<uint16_t>(11, 440000, &prng);
+ UniformBucketTest<uint16_t>(3, 270000, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestSignedInt) {
+ Random prng(7297352569824ull);
+ UniformBucketTest<signed int>(64, 640000, &prng);
+ UniformBucketTest<signed int>(11, 440000, &prng);
+ UniformBucketTest<signed int>(3, 270000, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, BucketTestUnsignedInt) {
+ Random prng(7297352569824ull);
+ UniformBucketTest<unsigned int>(64, 640000, &prng);
+ UniformBucketTest<unsigned int>(11, 440000, &prng);
+ UniformBucketTest<unsigned int>(3, 270000, &prng);
+}
+
+// The range of the random numbers is divided into bucket_count intervals
+// of consecutive numbers. Check that approximately equally many numbers
+// from each inteval are generated.
+void BucketTestSignedInterval(unsigned int bucket_count,
+ unsigned int samples,
+ int32_t low,
+ int32_t high,
+ int sigma_level,
+ Random* prng) {
+ std::vector<unsigned int> buckets(bucket_count, 0);
+
+ ASSERT_GE(high, low);
+ ASSERT_GE(bucket_count, 2u);
+ uint32_t interval = static_cast<uint32_t>(high - low + 1);
+ uint32_t numbers_per_bucket;
+ if (interval == 0) {
+ // The computation high - low + 1 should be 2^32 but overflowed
+ // Hence, bucket_count must be a power of 2
+ ASSERT_EQ(bucket_count & (bucket_count - 1), 0u);
+ numbers_per_bucket = (0x80000000u / bucket_count) * 2;
+ } else {
+ ASSERT_EQ(interval % bucket_count, 0u);
+ numbers_per_bucket = interval / bucket_count;
+ }
+
+ for (unsigned int i = 0; i < samples; i++) {
+ int32_t sample = prng->Rand(low, high);
+ EXPECT_LE(low, sample);
+ EXPECT_GE(high, sample);
+ buckets[static_cast<uint32_t>(sample - low) / numbers_per_bucket]++;
+ }
+
+ for (unsigned int i = 0; i < bucket_count; i++) {
+ // Expect the result to be within 3 standard deviations of the mean,
+ // or more generally, within sigma_level standard deviations of the mean.
+ double mean = static_cast<double>(samples) / bucket_count;
+ EXPECT_NEAR(buckets[i], mean, sigma_level * sqrt(mean));
+ }
+}
+
+// The range of the random numbers is divided into bucket_count intervals
+// of consecutive numbers. Check that approximately equally many numbers
+// from each inteval are generated.
+void BucketTestUnsignedInterval(unsigned int bucket_count,
+ unsigned int samples,
+ uint32_t low,
+ uint32_t high,
+ int sigma_level,
+ Random* prng) {
+ std::vector<unsigned int> buckets(bucket_count, 0);
+
+ ASSERT_GE(high, low);
+ ASSERT_GE(bucket_count, 2u);
+ uint32_t interval = static_cast<uint32_t>(high - low + 1);
+ uint32_t numbers_per_bucket;
+ if (interval == 0) {
+ // The computation high - low + 1 should be 2^32 but overflowed
+ // Hence, bucket_count must be a power of 2
+ ASSERT_EQ(bucket_count & (bucket_count - 1), 0u);
+ numbers_per_bucket = (0x80000000u / bucket_count) * 2;
+ } else {
+ ASSERT_EQ(interval % bucket_count, 0u);
+ numbers_per_bucket = interval / bucket_count;
+ }
+
+ for (unsigned int i = 0; i < samples; i++) {
+ uint32_t sample = prng->Rand(low, high);
+ EXPECT_LE(low, sample);
+ EXPECT_GE(high, sample);
+ buckets[static_cast<uint32_t>(sample - low) / numbers_per_bucket]++;
+ }
+
+ for (unsigned int i = 0; i < bucket_count; i++) {
+ // Expect the result to be within 3 standard deviations of the mean,
+ // or more generally, within sigma_level standard deviations of the mean.
+ double mean = static_cast<double>(samples) / bucket_count;
+ EXPECT_NEAR(buckets[i], mean, sigma_level * sqrt(mean));
+ }
+}
+
+TEST(RandomNumberGeneratorTest, UniformUnsignedInterval) {
+ Random prng(299792458ull);
+ BucketTestUnsignedInterval(2, 100000, 0, 1, 3, &prng);
+ BucketTestUnsignedInterval(7, 100000, 1, 14, 3, &prng);
+ BucketTestUnsignedInterval(11, 100000, 1000, 1010, 3, &prng);
+ BucketTestUnsignedInterval(100, 100000, 0, 99, 3, &prng);
+ BucketTestUnsignedInterval(2, 100000, 0, 4294967295, 3, &prng);
+ BucketTestUnsignedInterval(17, 100000, 455, 2147484110, 3, &prng);
+ // 99.7% of all samples will be within 3 standard deviations of the mean,
+ // but since we test 1000 buckets we allow an interval of 4 sigma.
+ BucketTestUnsignedInterval(1000, 1000000, 0, 2147483999, 4, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, UniformSignedInterval) {
+ Random prng(66260695729ull);
+ BucketTestSignedInterval(2, 100000, 0, 1, 3, &prng);
+ BucketTestSignedInterval(7, 100000, -2, 4, 3, &prng);
+ BucketTestSignedInterval(11, 100000, 1000, 1010, 3, &prng);
+ BucketTestSignedInterval(100, 100000, 0, 99, 3, &prng);
+ BucketTestSignedInterval(2, 100000, std::numeric_limits<int32_t>::min(),
+ std::numeric_limits<int32_t>::max(), 3, &prng);
+ BucketTestSignedInterval(17, 100000, -1073741826, 1073741829, 3, &prng);
+ // 99.7% of all samples will be within 3 standard deviations of the mean,
+ // but since we test 1000 buckets we allow an interval of 4 sigma.
+ BucketTestSignedInterval(1000, 1000000, -352, 2147483647, 4, &prng);
+}
+
+// The range of the random numbers is divided into bucket_count intervals
+// of consecutive numbers. Check that approximately equally many numbers
+// from each inteval are generated.
+void BucketTestFloat(unsigned int bucket_count,
+ unsigned int samples,
+ int sigma_level,
+ Random* prng) {
+ ASSERT_GE(bucket_count, 2u);
+ std::vector<unsigned int> buckets(bucket_count, 0);
+
+ for (unsigned int i = 0; i < samples; i++) {
+ uint32_t sample = bucket_count * prng->Rand<float>();
+ EXPECT_LE(0u, sample);
+ EXPECT_GE(bucket_count - 1, sample);
+ buckets[sample]++;
+ }
+
+ for (unsigned int i = 0; i < bucket_count; i++) {
+ // Expect the result to be within 3 standard deviations of the mean,
+ // or more generally, within sigma_level standard deviations of the mean.
+ double mean = static_cast<double>(samples) / bucket_count;
+ EXPECT_NEAR(buckets[i], mean, sigma_level * sqrt(mean));
+ }
+}
+
+TEST(RandomNumberGeneratorTest, UniformFloatInterval) {
+ Random prng(1380648813ull);
+ BucketTestFloat(100, 100000, 3, &prng);
+ // 99.7% of all samples will be within 3 standard deviations of the mean,
+ // but since we test 1000 buckets we allow an interval of 4 sigma.
+ // BucketTestSignedInterval(1000, 1000000, -352, 2147483647, 4, &prng);
+}
+
+TEST(RandomNumberGeneratorTest, SignedHasSameBitPattern) {
+ Random prng_signed(66738480ull), prng_unsigned(66738480ull);
+
+ for (int i = 0; i < 1000; i++) {
+ signed int s = prng_signed.Rand<signed int>();
+ unsigned int u = prng_unsigned.Rand<unsigned int>();
+ EXPECT_EQ(u, static_cast<unsigned int>(s));
+ }
+
+ for (int i = 0; i < 1000; i++) {
+ int16_t s = prng_signed.Rand<int16_t>();
+ uint16_t u = prng_unsigned.Rand<uint16_t>();
+ EXPECT_EQ(u, static_cast<uint16_t>(s));
+ }
+
+ for (int i = 0; i < 1000; i++) {
+ signed char s = prng_signed.Rand<signed char>();
+ unsigned char u = prng_unsigned.Rand<unsigned char>();
+ EXPECT_EQ(u, static_cast<unsigned char>(s));
+ }
+}
+
+TEST(RandomNumberGeneratorTest, Gaussian) {
+ const int kN = 100000;
+ const int kBuckets = 100;
+ const double kMean = 49;
+ const double kStddev = 10;
+
+ Random prng(1256637061);
+
+ std::vector<unsigned int> buckets(kBuckets, 0);
+ for (int i = 0; i < kN; i++) {
+ int index = prng.Gaussian(kMean, kStddev) + 0.5;
+ if (index >= 0 && index < kBuckets) {
+ buckets[index]++;
+ }
+ }
+
+ const double kPi = 3.14159265358979323846;
+ const double kScale = 1 / (kStddev * sqrt(2.0 * kPi));
+ const double kDiv = -2.0 * kStddev * kStddev;
+ for (int n = 0; n < kBuckets; ++n) {
+ // Use Simpsons rule to estimate the probability that a random gaussian
+ // sample is in the interval [n-0.5, n+0.5].
+ double f_left = kScale * exp((n - kMean - 0.5) * (n - kMean - 0.5) / kDiv);
+ double f_mid = kScale * exp((n - kMean) * (n - kMean) / kDiv);
+ double f_right = kScale * exp((n - kMean + 0.5) * (n - kMean + 0.5) / kDiv);
+ double normal_dist = (f_left + 4 * f_mid + f_right) / 6;
+ // Expect the number of samples to be within 3 standard deviations
+ // (rounded up) of the expected number of samples in the bucket.
+ EXPECT_NEAR(buckets[n], kN * normal_dist, 3 * sqrt(kN * normal_dist) + 1);
+ }
+}
+
+} // namespace webrtc
diff --git a/webrtc/base/ratetracker.cc b/webrtc/base/ratetracker.cc
index 5cb4490..35521a8 100644
--- a/webrtc/base/ratetracker.cc
+++ b/webrtc/base/ratetracker.cc
@@ -73,8 +73,9 @@
size_t start_bucket = NextBucketIndex(current_bucket_ + buckets_to_skip);
// Only count a portion of the first bucket according to how much of the
// first bucket is within the current interval.
- size_t total_samples = sample_buckets_[start_bucket] *
- (bucket_milliseconds_ - milliseconds_to_skip) /
+ size_t total_samples = ((sample_buckets_[start_bucket] *
+ (bucket_milliseconds_ - milliseconds_to_skip)) +
+ (bucket_milliseconds_ >> 1)) /
bucket_milliseconds_;
// All other buckets in the interval are counted in their entirety.
for (size_t i = NextBucketIndex(start_bucket);
diff --git a/webrtc/base/rtccertificate.cc b/webrtc/base/rtccertificate.cc
index a176d90..7b764bd 100644
--- a/webrtc/base/rtccertificate.cc
+++ b/webrtc/base/rtccertificate.cc
@@ -11,7 +11,6 @@
#include "webrtc/base/rtccertificate.h"
#include "webrtc/base/checks.h"
-#include "webrtc/base/timeutils.h"
namespace rtc {
@@ -28,13 +27,16 @@
RTCCertificate::~RTCCertificate() {
}
-uint64_t RTCCertificate::expires_timestamp_ns() const {
- // TODO(hbos): Update once SSLIdentity/SSLCertificate supports expires field.
- return 0;
+uint64_t RTCCertificate::Expires() const {
+ int64_t expires = ssl_certificate().CertificateExpirationTime();
+ if (expires != -1)
+ return static_cast<uint64_t>(expires) * kNumMillisecsPerSec;
+ // If the expiration time could not be retrieved return an expired timestamp.
+ return 0; // = 1970-01-01
}
-bool RTCCertificate::HasExpired() const {
- return expires_timestamp_ns() <= TimeNanos();
+bool RTCCertificate::HasExpired(uint64_t now) const {
+ return Expires() <= now;
}
const SSLCertificate& RTCCertificate::ssl_certificate() const {
diff --git a/webrtc/base/rtccertificate.h b/webrtc/base/rtccertificate.h
index d238938..600739b 100644
--- a/webrtc/base/rtccertificate.h
+++ b/webrtc/base/rtccertificate.h
@@ -27,8 +27,11 @@
// Takes ownership of |identity|.
static scoped_refptr<RTCCertificate> Create(scoped_ptr<SSLIdentity> identity);
- uint64_t expires_timestamp_ns() const;
- bool HasExpired() const;
+ // Returns the expiration time in ms relative to epoch, 1970-01-01T00:00:00Z.
+ uint64_t Expires() const;
+ // Checks if the certificate has expired, where |now| is expressed in ms
+ // relative to epoch, 1970-01-01T00:00:00Z.
+ bool HasExpired(uint64_t now) const;
const SSLCertificate& ssl_certificate() const;
// TODO(hbos): If possible, remove once RTCCertificate and its
diff --git a/webrtc/base/rtccertificate_unittests.cc b/webrtc/base/rtccertificate_unittests.cc
new file mode 100644
index 0000000..84c8544
--- /dev/null
+++ b/webrtc/base/rtccertificate_unittests.cc
@@ -0,0 +1,118 @@
+/*
+ * Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <utility>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/fakesslidentity.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/rtccertificate.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/sslidentity.h"
+#include "webrtc/base/thread.h"
+#include "webrtc/base/timeutils.h"
+
+namespace rtc {
+
+namespace {
+
+static const char* kTestCertCommonName = "RTCCertificateTest's certificate";
+
+} // namespace
+
+class RTCCertificateTest : public testing::Test {
+ public:
+ RTCCertificateTest() {}
+ ~RTCCertificateTest() {}
+
+ protected:
+ // Timestamp note:
+ // All timestamps in this unittest are expressed in number of seconds since
+ // epoch, 1970-01-01T00:00:00Z (UTC). The RTCCertificate interface uses ms,
+ // but only seconds-precision is supported by SSLCertificate. To make the
+ // tests clearer we convert everything to seconds since the precision matters
+ // when generating certificates or comparing timestamps.
+ // As a result, ExpiresSeconds and HasExpiredSeconds are used instead of
+ // RTCCertificate::Expires and ::HasExpired for ms -> s conversion.
+
+ uint64_t NowSeconds() const {
+ return TimeNanos() / kNumNanosecsPerSec;
+ }
+
+ uint64_t ExpiresSeconds(const scoped_refptr<RTCCertificate>& cert) const {
+ uint64_t exp_ms = cert->Expires();
+ uint64_t exp_s = exp_ms / kNumMillisecsPerSec;
+ // Make sure this did not result in loss of precision.
+ RTC_CHECK_EQ(exp_s * kNumMillisecsPerSec, exp_ms);
+ return exp_s;
+ }
+
+ bool HasExpiredSeconds(const scoped_refptr<RTCCertificate>& cert,
+ uint64_t now_s) const {
+ return cert->HasExpired(now_s * kNumMillisecsPerSec);
+ }
+
+ // An RTC_CHECK ensures that |expires_s| this is in valid range of time_t as
+ // is required by SSLIdentityParams. On some 32-bit systems time_t is limited
+ // to < 2^31. On such systems this will fail for expiration times of year 2038
+ // or later.
+ scoped_refptr<RTCCertificate> GenerateCertificateWithExpires(
+ uint64_t expires_s) const {
+ RTC_CHECK(IsValueInRangeForNumericType<time_t>(expires_s));
+
+ SSLIdentityParams params;
+ params.common_name = kTestCertCommonName;
+ params.not_before = 0;
+ params.not_after = static_cast<time_t>(expires_s);
+ // Certificate type does not matter for our purposes, using ECDSA because it
+ // is fast to generate.
+ params.key_params = KeyParams::ECDSA();
+
+ scoped_ptr<SSLIdentity> identity(SSLIdentity::GenerateForTest(params));
+ return RTCCertificate::Create(std::move(identity));
+ }
+};
+
+TEST_F(RTCCertificateTest, NewCertificateNotExpired) {
+ // Generate a real certificate without specifying the expiration time.
+ // Certificate type doesn't matter, using ECDSA because it's fast to generate.
+ scoped_ptr<SSLIdentity> identity(
+ SSLIdentity::Generate(kTestCertCommonName, KeyParams::ECDSA()));
+ scoped_refptr<RTCCertificate> certificate =
+ RTCCertificate::Create(std::move(identity));
+
+ uint64_t now = NowSeconds();
+ EXPECT_FALSE(HasExpiredSeconds(certificate, now));
+ // Even without specifying the expiration time we would expect it to be valid
+ // for at least half an hour.
+ EXPECT_FALSE(HasExpiredSeconds(certificate, now + 30*60));
+}
+
+TEST_F(RTCCertificateTest, UsesExpiresAskedFor) {
+ uint64_t now = NowSeconds();
+ scoped_refptr<RTCCertificate> certificate =
+ GenerateCertificateWithExpires(now);
+ EXPECT_EQ(now, ExpiresSeconds(certificate));
+}
+
+TEST_F(RTCCertificateTest, ExpiresInOneSecond) {
+ // Generate a certificate that expires in 1s.
+ uint64_t now = NowSeconds();
+ scoped_refptr<RTCCertificate> certificate =
+ GenerateCertificateWithExpires(now + 1);
+ // Now it should not have expired.
+ EXPECT_FALSE(HasExpiredSeconds(certificate, now));
+ // In 2s it should have expired.
+ EXPECT_TRUE(HasExpiredSeconds(certificate, now + 2));
+}
+
+} // namespace rtc
diff --git a/webrtc/base/scoped_ptr.h b/webrtc/base/scoped_ptr.h
index 4266d05..c4603c3 100644
--- a/webrtc/base/scoped_ptr.h
+++ b/webrtc/base/scoped_ptr.h
@@ -42,55 +42,39 @@
// }
//
// These scopers also implement part of the functionality of C++11 unique_ptr
-// in that they are "movable but not copyable." You can use the scopers in
-// the parameter and return types of functions to signify ownership transfer
-// in to and out of a function. When calling a function that has a scoper
-// as the argument type, it must be called with the result of an analogous
-// scoper's Pass() function or another function that generates a temporary;
-// passing by copy will NOT work. Here is an example using scoped_ptr:
+// in that they are "movable but not copyable." You can use the scopers in the
+// parameter and return types of functions to signify ownership transfer in to
+// and out of a function. When calling a function that has a scoper as the
+// argument type, it must be called with the result of calling std::move on an
+// analogous scoper, or another function that generates a temporary; passing by
+// copy will NOT work. Here is an example using scoped_ptr:
//
// void TakesOwnership(scoped_ptr<Foo> arg) {
// // Do something with arg
// }
// scoped_ptr<Foo> CreateFoo() {
-// // No need for calling Pass() because we are constructing a temporary
+// // No need for calling std::move because we are constructing a temporary
// // for the return value.
// return scoped_ptr<Foo>(new Foo("new"));
// }
// scoped_ptr<Foo> PassThru(scoped_ptr<Foo> arg) {
-// return arg.Pass();
+// return std::move(arg);
// }
//
// {
// scoped_ptr<Foo> ptr(new Foo("yay")); // ptr manages Foo("yay").
-// TakesOwnership(ptr.Pass()); // ptr no longer owns Foo("yay").
+// TakesOwnership(std::move(ptr)); // ptr no longer owns Foo("yay").
// scoped_ptr<Foo> ptr2 = CreateFoo(); // ptr2 owns the return Foo.
// scoped_ptr<Foo> ptr3 = // ptr3 now owns what was in ptr2.
-// PassThru(ptr2.Pass()); // ptr2 is correspondingly nullptr.
+// PassThru(std::move(ptr2)); // ptr2 is correspondingly nullptr.
// }
//
-// Notice that if you do not call Pass() when returning from PassThru(), or
+// Notice that if you do not call std::move when returning from PassThru(), or
// when invoking TakesOwnership(), the code will not compile because scopers
// are not copyable; they only implement move semantics which require calling
-// the Pass() function to signify a destructive transfer of state. CreateFoo()
-// is different though because we are constructing a temporary on the return
-// line and thus can avoid needing to call Pass().
-//
-// Pass() properly handles upcast in initialization, i.e. you can use a
-// scoped_ptr<Child> to initialize a scoped_ptr<Parent>:
-//
-// scoped_ptr<Foo> foo(new Foo());
-// scoped_ptr<FooParent> parent(foo.Pass());
-//
-// PassAs<>() should be used to upcast return value in return statement:
-//
-// scoped_ptr<Foo> CreateFoo() {
-// scoped_ptr<FooChild> result(new FooChild());
-// return result.PassAs<Foo>();
-// }
-//
-// Note that PassAs<>() is implemented only for scoped_ptr<T>, but not for
-// scoped_ptr<T[]>. This is because casting array pointers may not be safe.
+// std::move to signify a destructive transfer of state. CreateFoo() is
+// different though because we are constructing a temporary on the return line
+// and thus can avoid needing to call std::move.
#ifndef WEBRTC_BASE_SCOPED_PTR_H__
#define WEBRTC_BASE_SCOPED_PTR_H__
@@ -103,8 +87,10 @@
#include <stdlib.h>
#include <algorithm> // For std::swap().
+#include <cstddef>
#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/deprecation.h"
#include "webrtc/base/template_util.h"
#include "webrtc/typedefs.h"
@@ -342,7 +328,7 @@
scoped_ptr(element_type* p, const D& d) : impl_(p, d) {}
// Constructor. Allows construction from a nullptr.
- scoped_ptr(decltype(nullptr)) : impl_(nullptr) {}
+ scoped_ptr(std::nullptr_t) : impl_(nullptr) {}
// Constructor. Allows construction from a scoped_ptr rvalue for a
// convertible type and deleter.
@@ -379,7 +365,7 @@
// operator=. Allows assignment from a nullptr. Deletes the currently owned
// object, if any.
- scoped_ptr& operator=(decltype(nullptr)) {
+ scoped_ptr& operator=(std::nullptr_t) {
reset();
return *this;
}
@@ -389,7 +375,10 @@
scoped_ptr& operator=(const scoped_ptr& other) = delete;
// Get an rvalue reference. (sp.Pass() does the same thing as std::move(sp).)
- scoped_ptr&& Pass() { return static_cast<scoped_ptr&&>(*this); }
+ // Deprecated; remove in March 2016 (bug 5373).
+ RTC_DEPRECATED scoped_ptr&& Pass() {
+ return std::move(*this);
+ }
// Reset. Deletes the currently owned object, if any.
// Then takes ownership of a new object, if given.
@@ -499,7 +488,7 @@
explicit scoped_ptr(element_type* array) : impl_(array) {}
// Constructor. Allows construction from a nullptr.
- scoped_ptr(decltype(nullptr)) : impl_(nullptr) {}
+ scoped_ptr(std::nullptr_t) : impl_(nullptr) {}
// Constructor. Allows construction from a scoped_ptr rvalue.
scoped_ptr(scoped_ptr&& other) : impl_(&other.impl_) {}
@@ -512,7 +501,7 @@
// operator=. Allows assignment from a nullptr. Deletes the currently owned
// array, if any.
- scoped_ptr& operator=(decltype(nullptr)) {
+ scoped_ptr& operator=(std::nullptr_t) {
reset();
return *this;
}
@@ -522,7 +511,10 @@
scoped_ptr& operator=(const scoped_ptr& other) = delete;
// Get an rvalue reference. (sp.Pass() does the same thing as std::move(sp).)
- scoped_ptr&& Pass() { return static_cast<scoped_ptr&&>(*this); }
+ // Deprecated; remove in March 2016 (bug 5373).
+ RTC_DEPRECATED scoped_ptr&& Pass() {
+ return std::move(*this);
+ }
// Reset. Deletes the currently owned array, if any.
// Then takes ownership of a new object, if given.
diff --git a/webrtc/base/sec_buffer.h b/webrtc/base/sec_buffer.h
index d4cda00..e6ffea4 100644
--- a/webrtc/base/sec_buffer.h
+++ b/webrtc/base/sec_buffer.h
@@ -119,7 +119,7 @@
}
// Accessor for the descriptor
- const PSecBufferDesc desc() const {
+ PSecBufferDesc desc() const {
return &desc_;
}
diff --git a/webrtc/base/sharedexclusivelock_unittest.cc b/webrtc/base/sharedexclusivelock_unittest.cc
index 2857e00..9b64ed7 100644
--- a/webrtc/base/sharedexclusivelock_unittest.cc
+++ b/webrtc/base/sharedexclusivelock_unittest.cc
@@ -16,7 +16,6 @@
#include "webrtc/base/sharedexclusivelock.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/signalthread.cc b/webrtc/base/signalthread.cc
index d03f386..75f7b77 100644
--- a/webrtc/base/signalthread.cc
+++ b/webrtc/base/signalthread.cc
@@ -39,13 +39,6 @@
return worker_.SetName(name, obj);
}
-bool SignalThread::SetPriority(ThreadPriority priority) {
- EnterExit ee(this);
- ASSERT(main_->IsCurrent());
- ASSERT(kInit == state_);
- return worker_.SetPriority(priority);
-}
-
void SignalThread::Start() {
EnterExit ee(this);
ASSERT(main_->IsCurrent());
diff --git a/webrtc/base/signalthread.h b/webrtc/base/signalthread.h
index 4dda889..ec250c6 100644
--- a/webrtc/base/signalthread.h
+++ b/webrtc/base/signalthread.h
@@ -45,9 +45,6 @@
// Context: Main Thread. Call before Start to change the worker's name.
bool SetName(const std::string& name, const void* obj);
- // Context: Main Thread. Call before Start to change the worker's priority.
- bool SetPriority(ThreadPriority priority);
-
// Context: Main Thread. Call to begin the worker thread.
void Start();
diff --git a/webrtc/base/signalthread_unittest.cc b/webrtc/base/signalthread_unittest.cc
index fe6c602..a583aef 100644
--- a/webrtc/base/signalthread_unittest.cc
+++ b/webrtc/base/signalthread_unittest.cc
@@ -11,7 +11,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/signalthread.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using namespace rtc;
diff --git a/webrtc/base/sigslot.h b/webrtc/base/sigslot.h
index d9b12b0..a5fd5f7 100644
--- a/webrtc/base/sigslot.h
+++ b/webrtc/base/sigslot.h
@@ -532,7 +532,7 @@
m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool connected(has_slots_interface* pclass)
{
lock_block<mt_policy> lock(this);
@@ -686,7 +686,7 @@
m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool connected(has_slots_interface* pclass)
{
lock_block<mt_policy> lock(this);
@@ -825,7 +825,7 @@
m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool connected(has_slots_interface* pclass)
{
lock_block<mt_policy> lock(this);
@@ -963,7 +963,7 @@
m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool connected(has_slots_interface* pclass)
{
lock_block<mt_policy> lock(this);
@@ -1101,7 +1101,7 @@
m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool connected(has_slots_interface* pclass)
{
lock_block<mt_policy> lock(this);
@@ -1241,7 +1241,7 @@
m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool connected(has_slots_interface* pclass)
{
lock_block<mt_policy> lock(this);
@@ -1381,7 +1381,7 @@
m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool connected(has_slots_interface* pclass)
{
lock_block<mt_policy> lock(this);
@@ -1521,7 +1521,7 @@
m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool connected(has_slots_interface* pclass)
{
lock_block<mt_policy> lock(this);
@@ -1662,7 +1662,7 @@
m_connected_slots.erase(m_connected_slots.begin(), m_connected_slots.end());
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool connected(has_slots_interface* pclass)
{
lock_block<mt_policy> lock(this);
diff --git a/webrtc/base/socket_unittest.cc b/webrtc/base/socket_unittest.cc
index d078d7c..8143823 100644
--- a/webrtc/base/socket_unittest.cc
+++ b/webrtc/base/socket_unittest.cc
@@ -10,6 +10,7 @@
#include "webrtc/base/socket_unittest.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/nethelpers.h"
@@ -827,7 +828,7 @@
// Fill the socket buffer.
char buf[1024 * 16] = {0};
int sends = 0;
- while (++sends && accepted->Send(&buf, ARRAY_SIZE(buf)) != -1) {}
+ while (++sends && accepted->Send(&buf, arraysize(buf)) != -1) {}
EXPECT_TRUE(accepted->IsBlocking());
// Wait until data is available.
@@ -835,7 +836,7 @@
// Pull data.
for (int i = 0; i < sends; ++i) {
- client->Recv(buf, ARRAY_SIZE(buf));
+ client->Recv(buf, arraysize(buf));
}
// Expect at least one additional writable callback.
@@ -845,7 +846,7 @@
// callbacks.
int extras = 0;
for (int i = 0; i < 100; ++i) {
- accepted->Send(&buf, ARRAY_SIZE(buf));
+ accepted->Send(&buf, arraysize(buf));
rtc::Thread::Current()->ProcessMessages(1);
if (sink.Check(accepted.get(), testing::SSE_WRITE)) {
extras++;
diff --git a/webrtc/base/socket_unittest.h b/webrtc/base/socket_unittest.h
index d368afb..e4a6b32 100644
--- a/webrtc/base/socket_unittest.h
+++ b/webrtc/base/socket_unittest.h
@@ -21,8 +21,9 @@
// socketserver, and call the SocketTest test methods.
class SocketTest : public testing::Test {
protected:
- SocketTest() : ss_(NULL), kIPv4Loopback(INADDR_LOOPBACK),
- kIPv6Loopback(in6addr_loopback) {}
+ SocketTest() : kIPv4Loopback(INADDR_LOOPBACK),
+ kIPv6Loopback(in6addr_loopback),
+ ss_(nullptr) {}
virtual void SetUp() { ss_ = Thread::Current()->socketserver(); }
void TestConnectIPv4();
void TestConnectIPv6();
@@ -57,6 +58,10 @@
void TestGetSetOptionsIPv4();
void TestGetSetOptionsIPv6();
+ static const int kTimeout = 5000; // ms
+ const IPAddress kIPv4Loopback;
+ const IPAddress kIPv6Loopback;
+
private:
void ConnectInternal(const IPAddress& loopback);
void ConnectWithDnsLookupInternal(const IPAddress& loopback,
@@ -77,12 +82,13 @@
void UdpReadyToSend(const IPAddress& loopback);
void GetSetOptionsInternal(const IPAddress& loopback);
- static const int kTimeout = 5000; // ms
SocketServer* ss_;
- const IPAddress kIPv4Loopback;
- const IPAddress kIPv6Loopback;
};
+// For unbound sockets, GetLocalAddress / GetRemoteAddress return AF_UNSPEC
+// values on Windows, but an empty address of the same family on Linux/MacOS X.
+bool IsUnspecOrEmptyIP(const IPAddress& address);
+
} // namespace rtc
#endif // WEBRTC_BASE_SOCKET_UNITTEST_H_
diff --git a/webrtc/base/socketadapters.cc b/webrtc/base/socketadapters.cc
index af2efb8..2b513dc 100644
--- a/webrtc/base/socketadapters.cc
+++ b/webrtc/base/socketadapters.cc
@@ -688,7 +688,7 @@
request.WriteUInt8(5); // Socks Version
request.WriteUInt8(1); // CONNECT
request.WriteUInt8(0); // Reserved
- if (dest_.IsUnresolved()) {
+ if (dest_.IsUnresolvedIP()) {
std::string hostname = dest_.hostname();
request.WriteUInt8(3); // DOMAINNAME
request.WriteUInt8(static_cast<uint8_t>(hostname.size()));
diff --git a/webrtc/base/socketaddress.cc b/webrtc/base/socketaddress.cc
index 79ede80..c5fd798 100644
--- a/webrtc/base/socketaddress.cc
+++ b/webrtc/base/socketaddress.cc
@@ -307,39 +307,6 @@
return ToSockAddrStorageHelper(addr, ip_, port_, scope_id_);
}
-bool SocketAddress::StringToIP(const std::string& hostname, uint32_t* ip) {
- in_addr addr;
- if (rtc::inet_pton(AF_INET, hostname.c_str(), &addr) == 0)
- return false;
- *ip = NetworkToHost32(addr.s_addr);
- return true;
-}
-
-bool SocketAddress::StringToIP(const std::string& hostname, IPAddress* ip) {
- in_addr addr4;
- if (rtc::inet_pton(AF_INET, hostname.c_str(), &addr4) > 0) {
- if (ip) {
- *ip = IPAddress(addr4);
- }
- return true;
- }
-
- in6_addr addr6;
- if (rtc::inet_pton(AF_INET6, hostname.c_str(), &addr6) > 0) {
- if (ip) {
- *ip = IPAddress(addr6);
- }
- return true;
- }
- return false;
-}
-
-uint32_t SocketAddress::StringToIP(const std::string& hostname) {
- uint32_t ip = 0;
- StringToIP(hostname, &ip);
- return ip;
-}
-
bool SocketAddressFromSockAddrStorage(const sockaddr_storage& addr,
SocketAddress* out) {
if (!out) {
diff --git a/webrtc/base/socketaddress.h b/webrtc/base/socketaddress.h
index 1d975a1..175d7a9 100644
--- a/webrtc/base/socketaddress.h
+++ b/webrtc/base/socketaddress.h
@@ -141,7 +141,6 @@
// Determines whether the hostname has been resolved to an IP.
bool IsUnresolvedIP() const;
- inline bool IsUnresolved() const { return IsUnresolvedIP(); } // deprecated
// Determines whether this address is identical to the given one.
bool operator ==(const SocketAddress& addr) const;
@@ -177,16 +176,6 @@
size_t ToDualStackSockAddrStorage(sockaddr_storage* saddr) const;
size_t ToSockAddrStorage(sockaddr_storage* saddr) const;
- // Converts the IP address given in dotted form into compact form.
- // Only dotted names (A.B.C.D) are converted.
- // Output integer is returned in host byte order.
- // TODO: Deprecate, replace wth agnostic versions.
- static bool StringToIP(const std::string& str, uint32_t* ip);
- static uint32_t StringToIP(const std::string& str);
-
- // Converts the IP address given in printable form into an IPAddress.
- static bool StringToIP(const std::string& str, IPAddress* ip);
-
private:
std::string hostname_;
IPAddress ip_;
diff --git a/webrtc/base/socketaddress_unittest.cc b/webrtc/base/socketaddress_unittest.cc
index 6e9f089..e235447 100644
--- a/webrtc/base/socketaddress_unittest.cc
+++ b/webrtc/base/socketaddress_unittest.cc
@@ -27,10 +27,11 @@
0x00, 0x00, 0xFF, 0xFF,
0x01, 0x02, 0x03, 0x04} } };
const std::string kTestV6AddrString = "2001:db8:1020:3040:5060:7080:90a0:b0c0";
-const std::string kTestV6AddrAnonymizedString = "2001:db8:1020::";
+const std::string kTestV6AddrAnonymizedString = "2001:db8:1020:x:x:x:x:x";
const std::string kTestV6AddrFullString =
"[2001:db8:1020:3040:5060:7080:90a0:b0c0]:5678";
-const std::string kTestV6AddrFullAnonymizedString = "[2001:db8:1020::]:5678";
+const std::string kTestV6AddrFullAnonymizedString =
+ "[2001:db8:1020:x:x:x:x:x]:5678";
TEST(SocketAddressTest, TestDefaultCtor) {
SocketAddress addr;
@@ -325,23 +326,26 @@
SocketAddress addr_v4("1.2.3.4", 5678);
EXPECT_EQ("1.2.3.4", addr_v4.HostAsURIString());
EXPECT_EQ("1.2.3.4:5678", addr_v4.ToString());
- EXPECT_EQ("1.2.3.4", addr_v4.HostAsSensitiveURIString());
- EXPECT_EQ("1.2.3.4:5678", addr_v4.ToSensitiveString());
- IPAddress::set_strip_sensitive(true);
+
+#if defined(NDEBUG)
EXPECT_EQ("1.2.3.x", addr_v4.HostAsSensitiveURIString());
EXPECT_EQ("1.2.3.x:5678", addr_v4.ToSensitiveString());
- IPAddress::set_strip_sensitive(false);
+#else
+ EXPECT_EQ("1.2.3.4", addr_v4.HostAsSensitiveURIString());
+ EXPECT_EQ("1.2.3.4:5678", addr_v4.ToSensitiveString());
+#endif // defined(NDEBUG)
SocketAddress addr_v6(kTestV6AddrString, 5678);
EXPECT_EQ("[" + kTestV6AddrString + "]", addr_v6.HostAsURIString());
EXPECT_EQ(kTestV6AddrFullString, addr_v6.ToString());
- EXPECT_EQ("[" + kTestV6AddrString + "]", addr_v6.HostAsSensitiveURIString());
- EXPECT_EQ(kTestV6AddrFullString, addr_v6.ToSensitiveString());
- IPAddress::set_strip_sensitive(true);
+#if defined(NDEBUG)
EXPECT_EQ("[" + kTestV6AddrAnonymizedString + "]",
addr_v6.HostAsSensitiveURIString());
EXPECT_EQ(kTestV6AddrFullAnonymizedString, addr_v6.ToSensitiveString());
- IPAddress::set_strip_sensitive(false);
+#else
+ EXPECT_EQ("[" + kTestV6AddrString + "]", addr_v6.HostAsSensitiveURIString());
+ EXPECT_EQ(kTestV6AddrFullString, addr_v6.ToSensitiveString());
+#endif // defined(NDEBUG)
}
} // namespace rtc
diff --git a/webrtc/base/sslidentity.cc b/webrtc/base/sslidentity.cc
index 180e60c..5f6b686 100644
--- a/webrtc/base/sslidentity.cc
+++ b/webrtc/base/sslidentity.cc
@@ -15,6 +15,7 @@
#include "webrtc/base/sslidentity.h"
+#include <ctime>
#include <string>
#include "webrtc/base/base64.h"
@@ -177,4 +178,74 @@
#endif // SSL_USE_OPENSSL
+// Read |n| bytes from ASN1 number string at *|pp| and return the numeric value.
+// Update *|pp| and *|np| to reflect number of read bytes.
+static inline int ASN1ReadInt(const unsigned char** pp, size_t* np, size_t n) {
+ const unsigned char* p = *pp;
+ int x = 0;
+ for (size_t i = 0; i < n; i++)
+ x = 10 * x + p[i] - '0';
+ *pp = p + n;
+ *np = *np - n;
+ return x;
+}
+
+int64_t ASN1TimeToSec(const unsigned char* s, size_t length, bool long_format) {
+ size_t bytes_left = length;
+
+ // Make sure the string ends with Z. Doing it here protects the strspn call
+ // from running off the end of the string in Z's absense.
+ if (length == 0 || s[length - 1] != 'Z')
+ return -1;
+
+ // Make sure we only have ASCII digits so that we don't need to clutter the
+ // code below and ASN1ReadInt with error checking.
+ size_t n = strspn(reinterpret_cast<const char*>(s), "0123456789");
+ if (n + 1 != length)
+ return -1;
+
+ int year;
+
+ // Read out ASN1 year, in either 2-char "UTCTIME" or 4-char "GENERALIZEDTIME"
+ // format. Both format use UTC in this context.
+ if (long_format) {
+ // ASN1 format: yyyymmddhh[mm[ss[.fff]]]Z where the Z is literal, but
+ // RFC 5280 requires us to only support exactly yyyymmddhhmmssZ.
+
+ if (bytes_left < 11)
+ return -1;
+
+ year = ASN1ReadInt(&s, &bytes_left, 4);
+ year -= 1900;
+ } else {
+ // ASN1 format: yymmddhhmm[ss]Z where the Z is literal, but RFC 5280
+ // requires us to only support exactly yymmddhhmmssZ.
+
+ if (bytes_left < 9)
+ return -1;
+
+ year = ASN1ReadInt(&s, &bytes_left, 2);
+ if (year < 50) // Per RFC 5280 4.1.2.5.1
+ year += 100;
+ }
+
+ std::tm tm;
+ tm.tm_year = year;
+
+ // Read out remaining ASN1 time data and store it in |tm| in documented
+ // std::tm format.
+ tm.tm_mon = ASN1ReadInt(&s, &bytes_left, 2) - 1;
+ tm.tm_mday = ASN1ReadInt(&s, &bytes_left, 2);
+ tm.tm_hour = ASN1ReadInt(&s, &bytes_left, 2);
+ tm.tm_min = ASN1ReadInt(&s, &bytes_left, 2);
+ tm.tm_sec = ASN1ReadInt(&s, &bytes_left, 2);
+
+ if (bytes_left != 1) {
+ // Now just Z should remain. Its existence was asserted above.
+ return -1;
+ }
+
+ return TmToSeconds(tm);
+}
+
} // namespace rtc
diff --git a/webrtc/base/sslidentity.h b/webrtc/base/sslidentity.h
index cf99426..a143ee4 100644
--- a/webrtc/base/sslidentity.h
+++ b/webrtc/base/sslidentity.h
@@ -19,6 +19,7 @@
#include "webrtc/base/buffer.h"
#include "webrtc/base/messagedigest.h"
+#include "webrtc/base/timeutils.h"
namespace rtc {
@@ -68,6 +69,10 @@
unsigned char* digest,
size_t size,
size_t* length) const = 0;
+
+ // Returns the time in seconds relative to epoch, 1970-01-01T00:00:00Z (UTC),
+ // or -1 if an expiration time could not be retrieved.
+ virtual int64_t CertificateExpirationTime() const = 0;
};
// SSLCertChain is a simple wrapper for a vector of SSLCertificates. It serves
@@ -168,8 +173,8 @@
// random string will be used.
struct SSLIdentityParams {
std::string common_name;
- int not_before; // offset from current time in seconds.
- int not_after; // offset from current time in seconds.
+ time_t not_before; // Absolute time since epoch in seconds.
+ time_t not_after; // Absolute time since epoch in seconds.
KeyParams key_params;
};
@@ -217,6 +222,11 @@
size_t length);
};
+// Convert from ASN1 time as restricted by RFC 5280 to seconds from 1970-01-01
+// 00.00 ("epoch"). If the ASN1 time cannot be read, return -1. The data at
+// |s| is not 0-terminated; its char count is defined by |length|.
+int64_t ASN1TimeToSec(const unsigned char* s, size_t length, bool long_format);
+
extern const char kPemTypeCertificate[];
extern const char kPemTypeRsaPrivateKey[];
extern const char kPemTypeEcPrivateKey[];
diff --git a/webrtc/base/sslidentity_unittest.cc b/webrtc/base/sslidentity_unittest.cc
index e8df415..3582edb 100644
--- a/webrtc/base/sslidentity_unittest.cc
+++ b/webrtc/base/sslidentity_unittest.cc
@@ -11,6 +11,7 @@
#include <string>
#include "webrtc/base/gunit.h"
+#include "webrtc/base/helpers.h"
#include "webrtc/base/ssladapter.h"
#include "webrtc/base/sslidentity.h"
@@ -295,3 +296,119 @@
TEST_F(SSLIdentityTest, GetSignatureDigestAlgorithm) {
TestGetSignatureDigestAlgorithm();
}
+
+class SSLIdentityExpirationTest : public testing::Test {
+ public:
+ SSLIdentityExpirationTest() {
+ // Set use of the test RNG to get deterministic expiration timestamp.
+ rtc::SetRandomTestMode(true);
+ }
+ ~SSLIdentityExpirationTest() {
+ // Put it back for the next test.
+ rtc::SetRandomTestMode(false);
+ }
+
+ void TestASN1TimeToSec() {
+ struct asn_example {
+ const char* string;
+ bool long_format;
+ int64_t want;
+ } static const data[] = {
+ // Valid examples.
+ {"19700101000000Z", true, 0},
+ {"700101000000Z", false, 0},
+ {"19700101000001Z", true, 1},
+ {"700101000001Z", false, 1},
+ {"19700101000100Z", true, 60},
+ {"19700101000101Z", true, 61},
+ {"19700101010000Z", true, 3600},
+ {"19700101010001Z", true, 3601},
+ {"19700101010100Z", true, 3660},
+ {"19700101010101Z", true, 3661},
+ {"710911012345Z", false, 53400225},
+ {"20000101000000Z", true, 946684800},
+ {"20000101000000Z", true, 946684800},
+ {"20151130140156Z", true, 1448892116},
+ {"151130140156Z", false, 1448892116},
+ {"20491231235959Z", true, 2524607999},
+ {"491231235959Z", false, 2524607999},
+ {"20500101000000Z", true, 2524607999+1},
+ {"20700101000000Z", true, 3155760000},
+ {"21000101000000Z", true, 4102444800},
+ {"24000101000000Z", true, 13569465600},
+
+ // Invalid examples.
+ {"19700101000000", true, -1}, // missing Z long format
+ {"19700101000000X", true, -1}, // X instead of Z long format
+ {"197001010000000", true, -1}, // 0 instead of Z long format
+ {"1970010100000000Z", true, -1}, // excess digits long format
+ {"700101000000", false, -1}, // missing Z short format
+ {"700101000000X", false, -1}, // X instead of Z short format
+ {"7001010000000", false, -1}, // 0 instead of Z short format
+ {"70010100000000Z", false, -1}, // excess digits short format
+ {":9700101000000Z", true, -1}, // invalid character
+ {"1:700101000001Z", true, -1}, // invalid character
+ {"19:00101000100Z", true, -1}, // invalid character
+ {"197:0101000101Z", true, -1}, // invalid character
+ {"1970:101010000Z", true, -1}, // invalid character
+ {"19700:01010001Z", true, -1}, // invalid character
+ {"197001:1010100Z", true, -1}, // invalid character
+ {"1970010:010101Z", true, -1}, // invalid character
+ {"70010100:000Z", false, -1}, // invalid character
+ {"700101000:01Z", false, -1}, // invalid character
+ {"2000010100:000Z", true, -1}, // invalid character
+ {"21000101000:00Z", true, -1}, // invalid character
+ {"240001010000:0Z", true, -1}, // invalid character
+ {"500101000000Z", false, -1}, // but too old for epoch
+ {"691231235959Z", false, -1}, // too old for epoch
+ {"19611118043000Z", false, -1}, // way too old for epoch
+ };
+
+ unsigned char buf[20];
+
+ // Run all examples and check for the expected result.
+ for (const auto& entry : data) {
+ size_t length = strlen(entry.string);
+ memcpy(buf, entry.string, length); // Copy the ASN1 string...
+ buf[length] = rtc::CreateRandomId(); // ...and terminate it with junk.
+ int64_t res = rtc::ASN1TimeToSec(buf, length, entry.long_format);
+ LOG(LS_VERBOSE) << entry.string;
+ ASSERT_EQ(entry.want, res);
+ }
+ // Run all examples again, but with an invalid length.
+ for (const auto& entry : data) {
+ size_t length = strlen(entry.string);
+ memcpy(buf, entry.string, length); // Copy the ASN1 string...
+ buf[length] = rtc::CreateRandomId(); // ...and terminate it with junk.
+ int64_t res = rtc::ASN1TimeToSec(buf, length - 1, entry.long_format);
+ LOG(LS_VERBOSE) << entry.string;
+ ASSERT_EQ(-1, res);
+ }
+ }
+
+ void TestExpireTime(int times) {
+ for (int i = 0; i < times; i++) {
+ rtc::SSLIdentityParams params;
+ params.common_name = "";
+ params.not_before = 0;
+ // We limit the time to < 2^31 here, i.e., we stay before 2038, since else
+ // we hit time offset limitations in OpenSSL on some 32-bit systems.
+ params.not_after = rtc::CreateRandomId() % 0x80000000;
+ // We test just ECDSA here since what we're out to exercise here is the
+ // code for expiration setting and reading.
+ params.key_params = rtc::KeyParams::ECDSA(rtc::EC_NIST_P256);
+ SSLIdentity* identity = rtc::SSLIdentity::GenerateForTest(params);
+ EXPECT_EQ(params.not_after,
+ identity->certificate().CertificateExpirationTime());
+ delete identity;
+ }
+ }
+};
+
+TEST_F(SSLIdentityExpirationTest, TestASN1TimeToSec) {
+ TestASN1TimeToSec();
+}
+
+TEST_F(SSLIdentityExpirationTest, TestExpireTime) {
+ TestExpireTime(500);
+}
diff --git a/webrtc/base/sslroots.h b/webrtc/base/sslroots.h
index 31d601c..0464ac8 100644
--- a/webrtc/base/sslroots.h
+++ b/webrtc/base/sslroots.h
@@ -2,83 +2,813 @@
// Google.
// It was generated with the following command line:
-// > python //depot/googleclient/talk/tools/generate_sslroots.py
-// //depot/google3/security/cacerts/for_connecting_to_google/roots.pem
+// > python tools/sslroots/generate_sslroots.py
+// https://pki.google.com/roots.pem
-/* subject:/C=SE/O=AddTrust AB/OU=AddTrust External TTP Network/CN=AddTrust External CA Root */
-/* issuer :/C=SE/O=AddTrust AB/OU=AddTrust External TTP Network/CN=AddTrust External CA Root */
+/* subject:/C=BE/O=GlobalSign nv-sa/OU=Root CA/CN=GlobalSign Root CA */
+/* issuer :/C=BE/O=GlobalSign nv-sa/OU=Root CA/CN=GlobalSign Root CA */
-namespace rtc {
-const unsigned char AddTrust_External_Root_certificate[1082]={
-0x30,0x82,0x04,0x36,0x30,0x82,0x03,0x1E,0xA0,0x03,0x02,0x01,0x02,0x02,0x01,0x01,
+const unsigned char GlobalSign_Root_CA_certificate[889]={
+0x30,0x82,0x03,0x75,0x30,0x82,0x02,0x5D,0xA0,0x03,0x02,0x01,0x02,0x02,0x0B,0x04,
+0x00,0x00,0x00,0x00,0x01,0x15,0x4B,0x5A,0xC3,0x94,0x30,0x0D,0x06,0x09,0x2A,0x86,
+0x48,0x86,0xF7,0x0D,0x01,0x01,0x05,0x05,0x00,0x30,0x57,0x31,0x0B,0x30,0x09,0x06,
+0x03,0x55,0x04,0x06,0x13,0x02,0x42,0x45,0x31,0x19,0x30,0x17,0x06,0x03,0x55,0x04,
+0x0A,0x13,0x10,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,0x20,0x6E,0x76,
+0x2D,0x73,0x61,0x31,0x10,0x30,0x0E,0x06,0x03,0x55,0x04,0x0B,0x13,0x07,0x52,0x6F,
+0x6F,0x74,0x20,0x43,0x41,0x31,0x1B,0x30,0x19,0x06,0x03,0x55,0x04,0x03,0x13,0x12,
+0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,0x20,0x52,0x6F,0x6F,0x74,0x20,
+0x43,0x41,0x30,0x1E,0x17,0x0D,0x39,0x38,0x30,0x39,0x30,0x31,0x31,0x32,0x30,0x30,
+0x30,0x30,0x5A,0x17,0x0D,0x32,0x38,0x30,0x31,0x32,0x38,0x31,0x32,0x30,0x30,0x30,
+0x30,0x5A,0x30,0x57,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x42,
+0x45,0x31,0x19,0x30,0x17,0x06,0x03,0x55,0x04,0x0A,0x13,0x10,0x47,0x6C,0x6F,0x62,
+0x61,0x6C,0x53,0x69,0x67,0x6E,0x20,0x6E,0x76,0x2D,0x73,0x61,0x31,0x10,0x30,0x0E,
+0x06,0x03,0x55,0x04,0x0B,0x13,0x07,0x52,0x6F,0x6F,0x74,0x20,0x43,0x41,0x31,0x1B,
+0x30,0x19,0x06,0x03,0x55,0x04,0x03,0x13,0x12,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,
+0x69,0x67,0x6E,0x20,0x52,0x6F,0x6F,0x74,0x20,0x43,0x41,0x30,0x82,0x01,0x22,0x30,
+0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x01,0x05,0x00,0x03,0x82,
+0x01,0x0F,0x00,0x30,0x82,0x01,0x0A,0x02,0x82,0x01,0x01,0x00,0xDA,0x0E,0xE6,0x99,
+0x8D,0xCE,0xA3,0xE3,0x4F,0x8A,0x7E,0xFB,0xF1,0x8B,0x83,0x25,0x6B,0xEA,0x48,0x1F,
+0xF1,0x2A,0xB0,0xB9,0x95,0x11,0x04,0xBD,0xF0,0x63,0xD1,0xE2,0x67,0x66,0xCF,0x1C,
+0xDD,0xCF,0x1B,0x48,0x2B,0xEE,0x8D,0x89,0x8E,0x9A,0xAF,0x29,0x80,0x65,0xAB,0xE9,
+0xC7,0x2D,0x12,0xCB,0xAB,0x1C,0x4C,0x70,0x07,0xA1,0x3D,0x0A,0x30,0xCD,0x15,0x8D,
+0x4F,0xF8,0xDD,0xD4,0x8C,0x50,0x15,0x1C,0xEF,0x50,0xEE,0xC4,0x2E,0xF7,0xFC,0xE9,
+0x52,0xF2,0x91,0x7D,0xE0,0x6D,0xD5,0x35,0x30,0x8E,0x5E,0x43,0x73,0xF2,0x41,0xE9,
+0xD5,0x6A,0xE3,0xB2,0x89,0x3A,0x56,0x39,0x38,0x6F,0x06,0x3C,0x88,0x69,0x5B,0x2A,
+0x4D,0xC5,0xA7,0x54,0xB8,0x6C,0x89,0xCC,0x9B,0xF9,0x3C,0xCA,0xE5,0xFD,0x89,0xF5,
+0x12,0x3C,0x92,0x78,0x96,0xD6,0xDC,0x74,0x6E,0x93,0x44,0x61,0xD1,0x8D,0xC7,0x46,
+0xB2,0x75,0x0E,0x86,0xE8,0x19,0x8A,0xD5,0x6D,0x6C,0xD5,0x78,0x16,0x95,0xA2,0xE9,
+0xC8,0x0A,0x38,0xEB,0xF2,0x24,0x13,0x4F,0x73,0x54,0x93,0x13,0x85,0x3A,0x1B,0xBC,
+0x1E,0x34,0xB5,0x8B,0x05,0x8C,0xB9,0x77,0x8B,0xB1,0xDB,0x1F,0x20,0x91,0xAB,0x09,
+0x53,0x6E,0x90,0xCE,0x7B,0x37,0x74,0xB9,0x70,0x47,0x91,0x22,0x51,0x63,0x16,0x79,
+0xAE,0xB1,0xAE,0x41,0x26,0x08,0xC8,0x19,0x2B,0xD1,0x46,0xAA,0x48,0xD6,0x64,0x2A,
+0xD7,0x83,0x34,0xFF,0x2C,0x2A,0xC1,0x6C,0x19,0x43,0x4A,0x07,0x85,0xE7,0xD3,0x7C,
+0xF6,0x21,0x68,0xEF,0xEA,0xF2,0x52,0x9F,0x7F,0x93,0x90,0xCF,0x02,0x03,0x01,0x00,
+0x01,0xA3,0x42,0x30,0x40,0x30,0x0E,0x06,0x03,0x55,0x1D,0x0F,0x01,0x01,0xFF,0x04,
+0x04,0x03,0x02,0x01,0x06,0x30,0x0F,0x06,0x03,0x55,0x1D,0x13,0x01,0x01,0xFF,0x04,
+0x05,0x30,0x03,0x01,0x01,0xFF,0x30,0x1D,0x06,0x03,0x55,0x1D,0x0E,0x04,0x16,0x04,
+0x14,0x60,0x7B,0x66,0x1A,0x45,0x0D,0x97,0xCA,0x89,0x50,0x2F,0x7D,0x04,0xCD,0x34,
+0xA8,0xFF,0xFC,0xFD,0x4B,0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,
+0x01,0x05,0x05,0x00,0x03,0x82,0x01,0x01,0x00,0xD6,0x73,0xE7,0x7C,0x4F,0x76,0xD0,
+0x8D,0xBF,0xEC,0xBA,0xA2,0xBE,0x34,0xC5,0x28,0x32,0xB5,0x7C,0xFC,0x6C,0x9C,0x2C,
+0x2B,0xBD,0x09,0x9E,0x53,0xBF,0x6B,0x5E,0xAA,0x11,0x48,0xB6,0xE5,0x08,0xA3,0xB3,
+0xCA,0x3D,0x61,0x4D,0xD3,0x46,0x09,0xB3,0x3E,0xC3,0xA0,0xE3,0x63,0x55,0x1B,0xF2,
+0xBA,0xEF,0xAD,0x39,0xE1,0x43,0xB9,0x38,0xA3,0xE6,0x2F,0x8A,0x26,0x3B,0xEF,0xA0,
+0x50,0x56,0xF9,0xC6,0x0A,0xFD,0x38,0xCD,0xC4,0x0B,0x70,0x51,0x94,0x97,0x98,0x04,
+0xDF,0xC3,0x5F,0x94,0xD5,0x15,0xC9,0x14,0x41,0x9C,0xC4,0x5D,0x75,0x64,0x15,0x0D,
+0xFF,0x55,0x30,0xEC,0x86,0x8F,0xFF,0x0D,0xEF,0x2C,0xB9,0x63,0x46,0xF6,0xAA,0xFC,
+0xDF,0xBC,0x69,0xFD,0x2E,0x12,0x48,0x64,0x9A,0xE0,0x95,0xF0,0xA6,0xEF,0x29,0x8F,
+0x01,0xB1,0x15,0xB5,0x0C,0x1D,0xA5,0xFE,0x69,0x2C,0x69,0x24,0x78,0x1E,0xB3,0xA7,
+0x1C,0x71,0x62,0xEE,0xCA,0xC8,0x97,0xAC,0x17,0x5D,0x8A,0xC2,0xF8,0x47,0x86,0x6E,
+0x2A,0xC4,0x56,0x31,0x95,0xD0,0x67,0x89,0x85,0x2B,0xF9,0x6C,0xA6,0x5D,0x46,0x9D,
+0x0C,0xAA,0x82,0xE4,0x99,0x51,0xDD,0x70,0xB7,0xDB,0x56,0x3D,0x61,0xE4,0x6A,0xE1,
+0x5C,0xD6,0xF6,0xFE,0x3D,0xDE,0x41,0xCC,0x07,0xAE,0x63,0x52,0xBF,0x53,0x53,0xF4,
+0x2B,0xE9,0xC7,0xFD,0xB6,0xF7,0x82,0x5F,0x85,0xD2,0x41,0x18,0xDB,0x81,0xB3,0x04,
+0x1C,0xC5,0x1F,0xA4,0x80,0x6F,0x15,0x20,0xC9,0xDE,0x0C,0x88,0x0A,0x1D,0xD6,0x66,
+0x55,0xE2,0xFC,0x48,0xC9,0x29,0x26,0x69,0xE0,
+};
+
+
+/* subject:/C=US/ST=New Jersey/L=Jersey City/O=The USERTRUST Network/CN=USERTrust RSA Certification Authority */
+/* issuer :/C=US/ST=New Jersey/L=Jersey City/O=The USERTRUST Network/CN=USERTrust RSA Certification Authority */
+
+
+const unsigned char USERTrust_RSA_Certification_Authority_certificate[1506]={
+0x30,0x82,0x05,0xDE,0x30,0x82,0x03,0xC6,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x01,
+0xFD,0x6D,0x30,0xFC,0xA3,0xCA,0x51,0xA8,0x1B,0xBC,0x64,0x0E,0x35,0x03,0x2D,0x30,
+0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x0C,0x05,0x00,0x30,0x81,
+0x88,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x55,0x53,0x31,0x13,
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+0xA8,0xFD,
+};
+
+
+/* subject:/C=US/O=Starfield Technologies, Inc./OU=Starfield Class 2 Certification Authority */
+/* issuer :/C=US/O=Starfield Technologies, Inc./OU=Starfield Class 2 Certification Authority */
+
+
+const unsigned char Starfield_Class_2_CA_certificate[1043]={
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+};
+
+
+/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G3 */
+/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G3 */
+
+
+const unsigned char Verisign_Class_3_Public_Primary_Certification_Authority___G3_certificate[1054]={
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+0xEA,0xFD,0xAD,0xB2,0xC2,0xDA,0xEF,0x02,0x47,0x13,0x8A,0xC0,0xF1,0xB3,0x31,0xAD,
+0x4F,0x1C,0xE1,0x4F,0x9C,0xAF,0x0F,0x0C,0x9D,0xF7,0x78,0x0D,0xD8,0xF4,0x35,0x56,
+0x80,0xDA,0xB7,0x6D,0x17,0x8F,0x9D,0x1E,0x81,0x64,0xE1,0xFE,0xC5,0x45,0xBA,0xAD,
+0x6B,0xB9,0x0A,0x7A,0x4E,0x4F,0x4B,0x84,0xEE,0x4B,0xF1,0x7D,0xDD,0x11,
+};
+
+
+/* subject:/C=US/ST=New Jersey/L=Jersey City/O=The USERTRUST Network/CN=USERTrust ECC Certification Authority */
+/* issuer :/C=US/ST=New Jersey/L=Jersey City/O=The USERTRUST Network/CN=USERTrust ECC Certification Authority */
+
+
+const unsigned char USERTrust_ECC_Certification_Authority_certificate[659]={
+0x30,0x82,0x02,0x8F,0x30,0x82,0x02,0x15,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x5C,
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+0x02,0x31,0x00,0xE9,0x2A,0x61,0x47,0x8C,0x52,0x4A,0x4B,0x4E,0x18,0x70,0xF6,0xD6,
+0x44,0xD6,0x6E,0xF5,0x83,0xBA,0x6D,0x58,0xBD,0x24,0xD9,0x56,0x48,0xEA,0xEF,0xC4,
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+0x57,0x6A,0x18,
+};
+
+
+/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA */
+/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA */
+
+
+const unsigned char GeoTrust_Global_CA_certificate[856]={
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+0x4E,0x9B,0x4B,0xEE,0x4A,0x6F,0x55,0x52,0xB3,0x24,0xA1,0x37,0x0A,0xEB,0x64,0x76,
+0x2A,0x2E,0x2C,0xF3,0xFD,0x3B,0x75,0x90,0xBF,0xFA,0x71,0xD8,0xC7,0x3D,0x37,0xD2,
+0xB5,0x05,0x95,0x62,0xB9,0xA6,0xDE,0x89,0x3D,0x36,0x7B,0x38,0x77,0x48,0x97,0xAC,
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+0xB8,0x9C,0xE4,0x1D,0xB6,0xAB,0xE6,0x94,0xA5,0xC1,0xC7,0x83,0xAD,0xDB,0xF5,0x27,
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+0x33,0x5F,0x92,0x09,0x2F,0x88,0x66,0x5D,0x77,0x97,0xC7,0x1D,0x76,0x13,0xA9,0xD5,
+0xE5,0xF1,0x16,0x09,0x11,0x35,0xD5,0xAC,0xDB,0x24,0x71,0x70,0x2C,0x98,0x56,0x0B,
+0xD9,0x17,0xB4,0xD1,0xE3,0x51,0x2B,0x5E,0x75,0xE8,0xD5,0xD0,0xDC,0x4F,0x34,0xED,
+0xC2,0x05,0x66,0x80,0xA1,0xCB,0xE6,0x33,
+};
+
+
+/* subject:/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Root Certificate Authority - G2 */
+/* issuer :/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Root Certificate Authority - G2 */
+
+
+const unsigned char Starfield_Root_Certificate_Authority___G2_certificate[993]={
+0x30,0x82,0x03,0xDD,0x30,0x82,0x02,0xC5,0xA0,0x03,0x02,0x01,0x02,0x02,0x01,0x00,
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+0x79,0x51,0x4E,0xCD,0x90,0xF4,0x67,0x07,0x19,0xC8,0x3C,0x46,0x7A,0x0D,0x01,0x7D,
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+0x77,0x38,0xD3,0x32,0x3E,0x38,0x81,0x75,0x96,0x73,0xFA,0x68,0x8F,0xB1,0xCB,0xCE,
+0x1F,0xC5,0xEC,0xFA,0x9C,0x7E,0xCF,0x7E,0xB1,0xF1,0x07,0x2D,0xB6,0xFC,0xBF,0xCA,
+0xA4,0xBF,0xD0,0x97,0x05,0x4A,0xBC,0xEA,0x18,0x28,0x02,0x90,0xBD,0x54,0x78,0x09,
+0x21,0x71,0xD3,0xD1,0x7D,0x1D,0xD9,0x16,0xB0,0xA9,0x61,0x3D,0xD0,0x0A,0x00,0x22,
+0xFC,0xC7,0x7B,0xCB,0x09,0x64,0x45,0x0B,0x3B,0x40,0x81,0xF7,0x7D,0x7C,0x32,0xF5,
+0x98,0xCA,0x58,0x8E,0x7D,0x2A,0xEE,0x90,0x59,0x73,0x64,0xF9,0x36,0x74,0x5E,0x25,
+0xA1,0xF5,0x66,0x05,0x2E,0x7F,0x39,0x15,0xA9,0x2A,0xFB,0x50,0x8B,0x8E,0x85,0x69,
+0xF4,
+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root G3 */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root G3 */
+
+
+const unsigned char DigiCert_Global_Root_G3_certificate[579]={
+0x30,0x82,0x02,0x3F,0x30,0x82,0x01,0xC5,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x05,
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+0x2B,0xA8,0x9A,0xA9,0x8A,0xC5,0xD1,0x00,0xBD,0xF8,0x54,0xE2,0x9A,0xE5,0x5B,0x7C,
+0xB3,0x27,0x17,
+};
+
+
+/* subject:/C=US/O=thawte, Inc./OU=(c) 2007 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA - G2 */
+/* issuer :/C=US/O=thawte, Inc./OU=(c) 2007 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA - G2 */
+
+
+const unsigned char thawte_Primary_Root_CA___G2_certificate[652]={
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+0xF7,0x58,0x3F,0x2E,0x72,0x02,0x57,0xA3,0x8F,0xA1,0x14,0x2E,
+};
+
+
+/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2008 VeriSign, Inc. - For authorized use only/CN=VeriSign Universal Root Certification Authority */
+/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2008 VeriSign, Inc. - For authorized use only/CN=VeriSign Universal Root Certification Authority */
+
+
+const unsigned char VeriSign_Universal_Root_Certification_Authority_certificate[1213]={
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+0x0D,0xDC,0xC8,0xB9,0x35,0xB9,0x70,0x4F,0x63,0xFA,0x24,0xFA,0x6C,0x83,0x8C,0x47,
+0x9D,0x3B,0x63,0xF3,0x9A,0xF9,0x76,0x32,0x95,0x91,0xB1,0x77,0xBC,0xAC,0x9A,0xBE,
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+0xAC,0xF1,0x63,0xCB,0xB8,0x4C,0x1D,0x59,0x90,0x4A,0xEF,0x90,0x16,0x28,0x1F,0x5A,
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+0xE3,0x21,0xC9,0x24,0x39,0xE9,0xFD,0x15,0x66,0x46,0xF4,0x1B,0x11,0xD0,0x4D,0x73,
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+0x07,0x11,0xF0,0xD5,0xDB,0xDD,0xE5,0x8C,0xF0,0xD5,0x32,0xB0,0x83,0xE6,0x57,0xE2,
+0x8F,0xBF,0xBE,0xA1,0xAA,0xBF,0x3D,0x1D,0xB5,0xD4,0x38,0xEA,0xD7,0xB0,0x5C,0x3A,
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+0x95,0x14,0x0E,0x7D,0xCD,0x95,0x34,0xD9,0xD2,0x8F,0x70,0x73,0x81,0x7B,0x9C,0x7E,
+0xBD,0x98,0x61,0xD8,0x45,0x87,0x98,0x90,0xC5,0xEB,0x86,0x30,0xC6,0x35,0xBF,0xF0,
+0xFF,0xC3,0x55,0x88,0x83,0x4B,0xEF,0x05,0x92,0x06,0x71,0xF2,0xB8,0x98,0x93,0xB7,
+0xEC,0xCD,0x82,0x61,0xF1,0x38,0xE6,0x4F,0x97,0x98,0x2A,0x5A,0x8D,
+};
+
+
+/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2007 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G4 */
+/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2007 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G4 */
+
+
+const unsigned char VeriSign_Class_3_Public_Primary_Certification_Authority___G4_certificate[904]={
+0x30,0x82,0x03,0x84,0x30,0x82,0x03,0x0A,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x2F,
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+0x2D,0xA7,0xD8,0x86,0x2A,0xDD,0x2E,0x10,
+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root G2 */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root G2 */
+
+
+const unsigned char DigiCert_Global_Root_G2_certificate[914]={
+0x30,0x82,0x03,0x8E,0x30,0x82,0x02,0x76,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x03,
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+0x32,0xB6,
};
@@ -156,6 +886,196 @@
};
+/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Premium ECC */
+/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Premium ECC */
+
+
+const unsigned char AffirmTrust_Premium_ECC_certificate[514]={
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+0x02,0x30,0x42,0x59,0x87,0x43,0xD4,0x51,0xDF,0xBA,0xD3,0x09,0x32,0x5A,0xCE,0x88,
+0x7E,0x57,0x3D,0x9C,0x5F,0x42,0x6B,0xF5,0x07,0x2D,0xB5,0xF0,0x82,0x93,0xF9,0x59,
+0x6F,0xAE,0x64,0xFA,0x58,0xE5,0x8B,0x1E,0xE3,0x63,0xBE,0xB5,0x81,0xCD,0x6F,0x02,
+0x8C,0x79,
+};
+
+
+/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 4 Public Primary Certification Authority - G3 */
+/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 4 Public Primary Certification Authority - G3 */
+
+
+const unsigned char Verisign_Class_4_Public_Primary_Certification_Authority___G3_certificate[1054]={
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+0xF7,0x66,0x43,0xF3,0x9E,0x83,0x3E,0x20,0xAA,0xC3,0x35,0x60,0x91,0xCE,
+};
+
+
+/* subject:/C=US/O=thawte, Inc./OU=Certification Services Division/OU=(c) 2006 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA */
+/* issuer :/C=US/O=thawte, Inc./OU=Certification Services Division/OU=(c) 2006 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA */
+
+
+const unsigned char thawte_Primary_Root_CA_certificate[1060]={
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+0x8D,0x56,0x8C,0x68,
+};
+
+
/* subject:/C=SE/O=AddTrust AB/OU=AddTrust TTP Network/CN=AddTrust Public CA Root */
/* issuer :/C=SE/O=AddTrust AB/OU=AddTrust TTP Network/CN=AddTrust Public CA Root */
@@ -305,1822 +1225,6 @@
};
-/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Commercial */
-/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Commercial */
-
-
-const unsigned char AffirmTrust_Commercial_certificate[848]={
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-};
-
-
-/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Networking */
-/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Networking */
-
-
-const unsigned char AffirmTrust_Networking_certificate[848]={
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-};
-
-
-/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Premium */
-/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Premium */
-
-
-const unsigned char AffirmTrust_Premium_certificate[1354]={
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-};
-
-
-/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Premium ECC */
-/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Premium ECC */
-
-
-const unsigned char AffirmTrust_Premium_ECC_certificate[514]={
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-};
-
-
-/* subject:/C=US/O=America Online Inc./CN=America Online Root Certification Authority 1 */
-/* issuer :/C=US/O=America Online Inc./CN=America Online Root Certification Authority 1 */
-
-
-const unsigned char America_Online_Root_Certification_Authority_1_certificate[936]={
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-};
-
-
-/* subject:/C=US/O=America Online Inc./CN=America Online Root Certification Authority 2 */
-/* issuer :/C=US/O=America Online Inc./CN=America Online Root Certification Authority 2 */
-
-
-const unsigned char America_Online_Root_Certification_Authority_2_certificate[1448]={
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-};
-
-
-/* subject:/C=IE/O=Baltimore/OU=CyberTrust/CN=Baltimore CyberTrust Root */
-/* issuer :/C=IE/O=Baltimore/OU=CyberTrust/CN=Baltimore CyberTrust Root */
-
-
-const unsigned char Baltimore_CyberTrust_Root_certificate[891]={
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-};
-
-
-/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=AAA Certificate Services */
-/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=AAA Certificate Services */
-
-
-const unsigned char Comodo_AAA_Services_root_certificate[1078]={
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-};
-
-
-/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO Certification Authority */
-/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO Certification Authority */
-
-
-const unsigned char COMODO_Certification_Authority_certificate[1057]={
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-};
-
-
-/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO ECC Certification Authority */
-/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO ECC Certification Authority */
-
-
-const unsigned char COMODO_ECC_Certification_Authority_certificate[653]={
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-};
-
-
-/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Secure Certificate Services */
-/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Secure Certificate Services */
-
-
-const unsigned char Comodo_Secure_Services_root_certificate[1091]={
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-};
-
-
-/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Trusted Certificate Services */
-/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Trusted Certificate Services */
-
-
-const unsigned char Comodo_Trusted_Services_root_certificate[1095]={
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-};
-
-
-/* subject:/O=Cybertrust, Inc/CN=Cybertrust Global Root */
-/* issuer :/O=Cybertrust, Inc/CN=Cybertrust Global Root */
-
-
-const unsigned char Cybertrust_Global_Root_certificate[933]={
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-};
-
-
-/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root CA */
-/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root CA */
-
-
-const unsigned char DigiCert_Assured_ID_Root_CA_certificate[955]={
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-0xE6,0x50,0xB2,0xA7,0xFA,0x0A,0x45,0x2F,0xA2,0xF0,0xF2,
-};
-
-
-/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root CA */
-/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root CA */
-
-
-const unsigned char DigiCert_Global_Root_CA_certificate[947]={
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-0x01,0x12,0xAD,0xC8,0x88,0xC6,0x98,0x34,0x5F,0x8D,0x0A,0x3C,0xC6,0xE9,0xD5,0x95,
-0x95,0x6D,0xDE,
-};
-
-
-/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert High Assurance EV Root CA */
-/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert High Assurance EV Root CA */
-
-
-const unsigned char DigiCert_High_Assurance_EV_Root_CA_certificate[969]={
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-};
-
-
-/* subject:/O=Entrust.net/OU=www.entrust.net/CPS_2048 incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Certification Authority (2048) */
-/* issuer :/O=Entrust.net/OU=www.entrust.net/CPS_2048 incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Certification Authority (2048) */
-
-
-const unsigned char Entrust_net_Premium_2048_Secure_Server_CA_certificate[1120]={
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-};
-
-
-/* subject:/C=US/O=Entrust.net/OU=www.entrust.net/CPS incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Secure Server Certification Authority */
-/* issuer :/C=US/O=Entrust.net/OU=www.entrust.net/CPS incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Secure Server Certification Authority */
-
-
-const unsigned char Entrust_net_Secure_Server_CA_certificate[1244]={
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-};
-
-
-/* subject:/C=US/O=Entrust, Inc./OU=www.entrust.net/CPS is incorporated by reference/OU=(c) 2006 Entrust, Inc./CN=Entrust Root Certification Authority */
-/* issuer :/C=US/O=Entrust, Inc./OU=www.entrust.net/CPS is incorporated by reference/OU=(c) 2006 Entrust, Inc./CN=Entrust Root Certification Authority */
-
-
-const unsigned char Entrust_Root_Certification_Authority_certificate[1173]={
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-};
-
-
-/* subject:/C=US/O=Equifax/OU=Equifax Secure Certificate Authority */
-/* issuer :/C=US/O=Equifax/OU=Equifax Secure Certificate Authority */
-
-
-const unsigned char Equifax_Secure_CA_certificate[804]={
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-0xAC,0x07,0x77,0x38,
-};
-
-
-/* subject:/C=US/O=Equifax Secure Inc./CN=Equifax Secure eBusiness CA-1 */
-/* issuer :/C=US/O=Equifax Secure Inc./CN=Equifax Secure eBusiness CA-1 */
-
-
-const unsigned char Equifax_Secure_eBusiness_CA_1_certificate[646]={
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-0x5A,0x2A,0x82,0xB2,0x37,0x79,
-};
-
-
-/* subject:/C=US/O=Equifax Secure/OU=Equifax Secure eBusiness CA-2 */
-/* issuer :/C=US/O=Equifax Secure/OU=Equifax Secure eBusiness CA-2 */
-
-
-const unsigned char Equifax_Secure_eBusiness_CA_2_certificate[804]={
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-0x91,0x30,0xEA,0xCD,
-};
-
-
-/* subject:/C=US/O=Equifax Secure Inc./CN=Equifax Secure Global eBusiness CA-1 */
-/* issuer :/C=US/O=Equifax Secure Inc./CN=Equifax Secure Global eBusiness CA-1 */
-
-
-const unsigned char Equifax_Secure_Global_eBusiness_CA_certificate[660]={
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-0x56,0x94,0xA9,0x55,
-};
-
-
-/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA */
-/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA */
-
-
-const unsigned char GeoTrust_Global_CA_certificate[856]={
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-};
-
-
-/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA 2 */
-/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA 2 */
-
-
-const unsigned char GeoTrust_Global_CA_2_certificate[874]={
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-};
-
-
-/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Primary Certification Authority */
-/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Primary Certification Authority */
-
-
-const unsigned char GeoTrust_Primary_Certification_Authority_certificate[896]={
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-};
-
-
-/* subject:/C=US/O=GeoTrust Inc./OU=(c) 2007 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G2 */
-/* issuer :/C=US/O=GeoTrust Inc./OU=(c) 2007 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G2 */
-
-
-const unsigned char GeoTrust_Primary_Certification_Authority___G2_certificate[690]={
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-};
-
-
/* subject:/C=US/O=GeoTrust Inc./OU=(c) 2008 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G3 */
/* issuer :/C=US/O=GeoTrust Inc./OU=(c) 2008 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G3 */
@@ -2194,101 +1298,6 @@
};
-/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA */
-/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA */
-
-
-const unsigned char GeoTrust_Universal_CA_certificate[1388]={
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-};
-
-
/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA 2 */
/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA 2 */
@@ -2384,67 +1393,67 @@
};
-/* subject:/C=BE/O=GlobalSign nv-sa/OU=Root CA/CN=GlobalSign Root CA */
-/* issuer :/C=BE/O=GlobalSign nv-sa/OU=Root CA/CN=GlobalSign Root CA */
+/* subject:/C=IE/O=Baltimore/OU=CyberTrust/CN=Baltimore CyberTrust Root */
+/* issuer :/C=IE/O=Baltimore/OU=CyberTrust/CN=Baltimore CyberTrust Root */
-const unsigned char GlobalSign_Root_CA_certificate[889]={
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-0x05,0x30,0x03,0x01,0x01,0xFF,0x30,0x1D,0x06,0x03,0x55,0x1D,0x0E,0x04,0x16,0x04,
-0x14,0x60,0x7B,0x66,0x1A,0x45,0x0D,0x97,0xCA,0x89,0x50,0x2F,0x7D,0x04,0xCD,0x34,
-0xA8,0xFF,0xFC,0xFD,0x4B,0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,
-0x01,0x05,0x05,0x00,0x03,0x82,0x01,0x01,0x00,0xD6,0x73,0xE7,0x7C,0x4F,0x76,0xD0,
-0x8D,0xBF,0xEC,0xBA,0xA2,0xBE,0x34,0xC5,0x28,0x32,0xB5,0x7C,0xFC,0x6C,0x9C,0x2C,
-0x2B,0xBD,0x09,0x9E,0x53,0xBF,0x6B,0x5E,0xAA,0x11,0x48,0xB6,0xE5,0x08,0xA3,0xB3,
-0xCA,0x3D,0x61,0x4D,0xD3,0x46,0x09,0xB3,0x3E,0xC3,0xA0,0xE3,0x63,0x55,0x1B,0xF2,
-0xBA,0xEF,0xAD,0x39,0xE1,0x43,0xB9,0x38,0xA3,0xE6,0x2F,0x8A,0x26,0x3B,0xEF,0xA0,
-0x50,0x56,0xF9,0xC6,0x0A,0xFD,0x38,0xCD,0xC4,0x0B,0x70,0x51,0x94,0x97,0x98,0x04,
-0xDF,0xC3,0x5F,0x94,0xD5,0x15,0xC9,0x14,0x41,0x9C,0xC4,0x5D,0x75,0x64,0x15,0x0D,
-0xFF,0x55,0x30,0xEC,0x86,0x8F,0xFF,0x0D,0xEF,0x2C,0xB9,0x63,0x46,0xF6,0xAA,0xFC,
-0xDF,0xBC,0x69,0xFD,0x2E,0x12,0x48,0x64,0x9A,0xE0,0x95,0xF0,0xA6,0xEF,0x29,0x8F,
-0x01,0xB1,0x15,0xB5,0x0C,0x1D,0xA5,0xFE,0x69,0x2C,0x69,0x24,0x78,0x1E,0xB3,0xA7,
-0x1C,0x71,0x62,0xEE,0xCA,0xC8,0x97,0xAC,0x17,0x5D,0x8A,0xC2,0xF8,0x47,0x86,0x6E,
-0x2A,0xC4,0x56,0x31,0x95,0xD0,0x67,0x89,0x85,0x2B,0xF9,0x6C,0xA6,0x5D,0x46,0x9D,
-0x0C,0xAA,0x82,0xE4,0x99,0x51,0xDD,0x70,0xB7,0xDB,0x56,0x3D,0x61,0xE4,0x6A,0xE1,
-0x5C,0xD6,0xF6,0xFE,0x3D,0xDE,0x41,0xCC,0x07,0xAE,0x63,0x52,0xBF,0x53,0x53,0xF4,
-0x2B,0xE9,0xC7,0xFD,0xB6,0xF7,0x82,0x5F,0x85,0xD2,0x41,0x18,0xDB,0x81,0xB3,0x04,
-0x1C,0xC5,0x1F,0xA4,0x80,0x6F,0x15,0x20,0xC9,0xDE,0x0C,0x88,0x0A,0x1D,0xD6,0x66,
-0x55,0xE2,0xFC,0x48,0xC9,0x29,0x26,0x69,0xE0,
+const unsigned char Baltimore_CyberTrust_Root_certificate[891]={
+0x30,0x82,0x03,0x77,0x30,0x82,0x02,0x5F,0xA0,0x03,0x02,0x01,0x02,0x02,0x04,0x02,
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+0x0D,0x32,0x35,0x30,0x35,0x31,0x32,0x32,0x33,0x35,0x39,0x30,0x30,0x5A,0x30,0x5A,
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+0x98,0x3D,0x57,0xE8,0x26,0x72,0x9A,0xB5,0x79,0xD4,0x29,0xE2,0xE1,0xE8,0x95,0x80,
+0xB1,0xB0,0xE3,0x5B,0x8E,0x2B,0x29,0x9A,0x64,0xDF,0xA1,0x5D,0xED,0xB0,0x09,0x05,
+0x6D,0xDB,0x28,0x2E,0xCE,0x62,0xA2,0x62,0xFE,0xB4,0x88,0xDA,0x12,0xEB,0x38,0xEB,
+0x21,0x9D,0xC0,0x41,0x2B,0x01,0x52,0x7B,0x88,0x77,0xD3,0x1C,0x8F,0xC7,0xBA,0xB9,
+0x88,0xB5,0x6A,0x09,0xE7,0x73,0xE8,0x11,0x40,0xA7,0xD1,0xCC,0xCA,0x62,0x8D,0x2D,
+0xE5,0x8F,0x0B,0xA6,0x50,0xD2,0xA8,0x50,0xC3,0x28,0xEA,0xF5,0xAB,0x25,0x87,0x8A,
+0x9A,0x96,0x1C,0xA9,0x67,0xB8,0x3F,0x0C,0xD5,0xF7,0xF9,0x52,0x13,0x2F,0xC2,0x1B,
+0xD5,0x70,0x70,0xF0,0x8F,0xC0,0x12,0xCA,0x06,0xCB,0x9A,0xE1,0xD9,0xCA,0x33,0x7A,
+0x77,0xD6,0xF8,0xEC,0xB9,0xF1,0x68,0x44,0x42,0x48,0x13,0xD2,0xC0,0xC2,0xA4,0xAE,
+0x5E,0x60,0xFE,0xB6,0xA6,0x05,0xFC,0xB4,0xDD,0x07,0x59,0x02,0xD4,0x59,0x18,0x98,
+0x63,0xF5,0xA5,0x63,0xE0,0x90,0x0C,0x7D,0x5D,0xB2,0x06,0x7A,0xF3,0x85,0xEA,0xEB,
+0xD4,0x03,0xAE,0x5E,0x84,0x3E,0x5F,0xFF,0x15,0xED,0x69,0xBC,0xF9,0x39,0x36,0x72,
+0x75,0xCF,0x77,0x52,0x4D,0xF3,0xC9,0x90,0x2C,0xB9,0x3D,0xE5,0xC9,0x23,0x53,0x3F,
+0x1F,0x24,0x98,0x21,0x5C,0x07,0x99,0x29,0xBD,0xC6,0x3A,0xEC,0xE7,0x6E,0x86,0x3A,
+0x6B,0x97,0x74,0x63,0x33,0xBD,0x68,0x18,0x31,0xF0,0x78,0x8D,0x76,0xBF,0xFC,0x9E,
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+0xA3,0x45,0x30,0x43,0x30,0x1D,0x06,0x03,0x55,0x1D,0x0E,0x04,0x16,0x04,0x14,0xE5,
+0x9D,0x59,0x30,0x82,0x47,0x58,0xCC,0xAC,0xFA,0x08,0x54,0x36,0x86,0x7B,0x3A,0xB5,
+0x04,0x4D,0xF0,0x30,0x12,0x06,0x03,0x55,0x1D,0x13,0x01,0x01,0xFF,0x04,0x08,0x30,
+0x06,0x01,0x01,0xFF,0x02,0x01,0x03,0x30,0x0E,0x06,0x03,0x55,0x1D,0x0F,0x01,0x01,
+0xFF,0x04,0x04,0x03,0x02,0x01,0x06,0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,
+0x0D,0x01,0x01,0x05,0x05,0x00,0x03,0x82,0x01,0x01,0x00,0x85,0x0C,0x5D,0x8E,0xE4,
+0x6F,0x51,0x68,0x42,0x05,0xA0,0xDD,0xBB,0x4F,0x27,0x25,0x84,0x03,0xBD,0xF7,0x64,
+0xFD,0x2D,0xD7,0x30,0xE3,0xA4,0x10,0x17,0xEB,0xDA,0x29,0x29,0xB6,0x79,0x3F,0x76,
+0xF6,0x19,0x13,0x23,0xB8,0x10,0x0A,0xF9,0x58,0xA4,0xD4,0x61,0x70,0xBD,0x04,0x61,
+0x6A,0x12,0x8A,0x17,0xD5,0x0A,0xBD,0xC5,0xBC,0x30,0x7C,0xD6,0xE9,0x0C,0x25,0x8D,
+0x86,0x40,0x4F,0xEC,0xCC,0xA3,0x7E,0x38,0xC6,0x37,0x11,0x4F,0xED,0xDD,0x68,0x31,
+0x8E,0x4C,0xD2,0xB3,0x01,0x74,0xEE,0xBE,0x75,0x5E,0x07,0x48,0x1A,0x7F,0x70,0xFF,
+0x16,0x5C,0x84,0xC0,0x79,0x85,0xB8,0x05,0xFD,0x7F,0xBE,0x65,0x11,0xA3,0x0F,0xC0,
+0x02,0xB4,0xF8,0x52,0x37,0x39,0x04,0xD5,0xA9,0x31,0x7A,0x18,0xBF,0xA0,0x2A,0xF4,
+0x12,0x99,0xF7,0xA3,0x45,0x82,0xE3,0x3C,0x5E,0xF5,0x9D,0x9E,0xB5,0xC8,0x9E,0x7C,
+0x2E,0xC8,0xA4,0x9E,0x4E,0x08,0x14,0x4B,0x6D,0xFD,0x70,0x6D,0x6B,0x1A,0x63,0xBD,
+0x64,0xE6,0x1F,0xB7,0xCE,0xF0,0xF2,0x9F,0x2E,0xBB,0x1B,0xB7,0xF2,0x50,0x88,0x73,
+0x92,0xC2,0xE2,0xE3,0x16,0x8D,0x9A,0x32,0x02,0xAB,0x8E,0x18,0xDD,0xE9,0x10,0x11,
+0xEE,0x7E,0x35,0xAB,0x90,0xAF,0x3E,0x30,0x94,0x7A,0xD0,0x33,0x3D,0xA7,0x65,0x0F,
+0xF5,0xFC,0x8E,0x9E,0x62,0xCF,0x47,0x44,0x2C,0x01,0x5D,0xBB,0x1D,0xB5,0x32,0xD2,
+0x47,0xD2,0x38,0x2E,0xD0,0xFE,0x81,0xDC,0x32,0x6A,0x1E,0xB5,0xEE,0x3C,0xD5,0xFC,
+0xE7,0x81,0x1D,0x19,0xC3,0x24,0x42,0xEA,0x63,0x39,0xA9,
};
@@ -2579,1249 +1588,140 @@
};
-/* subject:/C=US/O=The Go Daddy Group, Inc./OU=Go Daddy Class 2 Certification Authority */
-/* issuer :/C=US/O=The Go Daddy Group, Inc./OU=Go Daddy Class 2 Certification Authority */
+/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Networking */
+/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Networking */
-const unsigned char Go_Daddy_Class_2_CA_certificate[1028]={
-0x30,0x82,0x04,0x00,0x30,0x82,0x02,0xE8,0xA0,0x03,0x02,0x01,0x02,0x02,0x01,0x00,
-0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x05,0x05,0x00,0x30,
-0x63,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x55,0x53,0x31,0x21,
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-};
-
-
-/* subject:/C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./CN=Go Daddy Root Certificate Authority - G2 */
-/* issuer :/C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./CN=Go Daddy Root Certificate Authority - G2 */
-
-
-const unsigned char Go_Daddy_Root_Certificate_Authority___G2_certificate[969]={
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-};
-
-
-/* subject:/C=US/O=GTE Corporation/OU=GTE CyberTrust Solutions, Inc./CN=GTE CyberTrust Global Root */
-/* issuer :/C=US/O=GTE Corporation/OU=GTE CyberTrust Solutions, Inc./CN=GTE CyberTrust Global Root */
-
-
-const unsigned char GTE_CyberTrust_Global_Root_certificate[606]={
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-};
-
-
-/* subject:/C=US/O=Network Solutions L.L.C./CN=Network Solutions Certificate Authority */
-/* issuer :/C=US/O=Network Solutions L.L.C./CN=Network Solutions Certificate Authority */
-
-
-const unsigned char Network_Solutions_Certificate_Authority_certificate[1002]={
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-};
-
-
-/* subject:/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 3 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-/* issuer :/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 3 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-
-
-const unsigned char RSA_Root_Certificate_1_certificate[747]={
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-};
-
-
-/* subject:/C=US/O=Starfield Technologies, Inc./OU=Starfield Class 2 Certification Authority */
-/* issuer :/C=US/O=Starfield Technologies, Inc./OU=Starfield Class 2 Certification Authority */
-
-
-const unsigned char Starfield_Class_2_CA_certificate[1043]={
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-};
-
-
-/* subject:/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Root Certificate Authority - G2 */
-/* issuer :/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Root Certificate Authority - G2 */
-
-
-const unsigned char Starfield_Root_Certificate_Authority___G2_certificate[993]={
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-};
-
-
-/* subject:/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Services Root Certificate Authority - G2 */
-/* issuer :/C=US/ST=Arizona/L=Scottsdale/O=Starfield Technologies, Inc./CN=Starfield Services Root Certificate Authority - G2 */
-
-
-const unsigned char Starfield_Services_Root_Certificate_Authority___G2_certificate[1011]={
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-};
-
-
-/* subject:/C=IL/O=StartCom Ltd./OU=Secure Digital Certificate Signing/CN=StartCom Certification Authority */
-/* issuer :/C=IL/O=StartCom Ltd./OU=Secure Digital Certificate Signing/CN=StartCom Certification Authority */
-
-
-const unsigned char StartCom_Certification_Authority_certificate[1931]={
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-};
-
-
-/* subject:/C=IL/O=StartCom Ltd./CN=StartCom Certification Authority G2 */
-/* issuer :/C=IL/O=StartCom Ltd./CN=StartCom Certification Authority G2 */
-
-
-const unsigned char StartCom_Certification_Authority_G2_certificate[1383]={
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-};
-
-
-/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 2 CA/CN=TC TrustCenter Class 2 CA II */
-/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 2 CA/CN=TC TrustCenter Class 2 CA II */
-
-
-const unsigned char TC_TrustCenter_Class_2_CA_II_certificate[1198]={
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-
-
-/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 3 CA/CN=TC TrustCenter Class 3 CA II */
-/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 3 CA/CN=TC TrustCenter Class 3 CA II */
-
-
-const unsigned char TC_TrustCenter_Class_3_CA_II_certificate[1198]={
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-};
-
-
-/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA I */
-/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA I */
-
-
-const unsigned char TC_TrustCenter_Universal_CA_I_certificate[993]={
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+const unsigned char AffirmTrust_Networking_certificate[848]={
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};
-/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA III */
-/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA III */
+/* subject:/C=SE/O=AddTrust AB/OU=AddTrust External TTP Network/CN=AddTrust External CA Root */
+/* issuer :/C=SE/O=AddTrust AB/OU=AddTrust External TTP Network/CN=AddTrust External CA Root */
-const unsigned char TC_TrustCenter_Universal_CA_III_certificate[997]={
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-};
-
-
-/* subject:/C=ZA/ST=Western Cape/L=Cape Town/O=Thawte Consulting cc/OU=Certification Services Division/CN=Thawte Premium Server CA/emailAddress=premium-server@thawte.com */
-/* issuer :/C=ZA/ST=Western Cape/L=Cape Town/O=Thawte Consulting cc/OU=Certification Services Division/CN=Thawte Premium Server CA/emailAddress=premium-server@thawte.com */
-
-
-const unsigned char Thawte_Premium_Server_CA_certificate[811]={
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-};
-
-
-/* subject:/C=US/O=thawte, Inc./OU=Certification Services Division/OU=(c) 2006 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA */
-/* issuer :/C=US/O=thawte, Inc./OU=Certification Services Division/OU=(c) 2006 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA */
-
-
-const unsigned char thawte_Primary_Root_CA_certificate[1060]={
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-};
-
-
-/* subject:/C=US/O=thawte, Inc./OU=(c) 2007 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA - G2 */
-/* issuer :/C=US/O=thawte, Inc./OU=(c) 2007 thawte, Inc. - For authorized use only/CN=thawte Primary Root CA - G2 */
-
-
-const unsigned char thawte_Primary_Root_CA___G2_certificate[652]={
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+const unsigned char AddTrust_External_Root_certificate[1082]={
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};
@@ -3900,570 +1800,208 @@
};
-/* subject:/C=ZA/ST=Western Cape/L=Cape Town/O=Thawte Consulting cc/OU=Certification Services Division/CN=Thawte Server CA/emailAddress=server-certs@thawte.com */
-/* issuer :/C=ZA/ST=Western Cape/L=Cape Town/O=Thawte Consulting cc/OU=Certification Services Division/CN=Thawte Server CA/emailAddress=server-certs@thawte.com */
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root CA */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root CA */
-const unsigned char Thawte_Server_CA_certificate[791]={
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+const unsigned char DigiCert_Assured_ID_Root_CA_certificate[955]={
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};
-/* subject:/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN - DATACorp SGC */
-/* issuer :/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN - DATACorp SGC */
+/* subject:/C=US/O=The Go Daddy Group, Inc./OU=Go Daddy Class 2 Certification Authority */
+/* issuer :/C=US/O=The Go Daddy Group, Inc./OU=Go Daddy Class 2 Certification Authority */
-const unsigned char UTN_DATACorp_SGC_Root_CA_certificate[1122]={
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-0xF4,0x08,
+const unsigned char Go_Daddy_Class_2_CA_certificate[1028]={
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+0x7F,0xDB,0xBD,0x9F,
};
-/* subject:/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN-USERFirst-Hardware */
-/* issuer :/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN-USERFirst-Hardware */
+/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Primary Certification Authority */
+/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Primary Certification Authority */
-const unsigned char UTN_USERFirst_Hardware_Root_CA_certificate[1144]={
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-};
-
-
-/* subject:/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 1 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-/* issuer :/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 1 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-
-
-const unsigned char ValiCert_Class_1_VA_certificate[747]={
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-};
-
-
-/* subject:/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 2 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-/* issuer :/L=ValiCert Validation Network/O=ValiCert, Inc./OU=ValiCert Class 2 Policy Validation Authority/CN=http://www.valicert.com//emailAddress=info@valicert.com */
-
-
-const unsigned char ValiCert_Class_2_VA_certificate[747]={
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-
-
-/* subject:/C=US/O=VeriSign, Inc./OU=Class 3 Public Primary Certification Authority */
-/* issuer :/C=US/O=VeriSign, Inc./OU=Class 3 Public Primary Certification Authority */
-
-
-const unsigned char Verisign_Class_3_Public_Primary_Certification_Authority_certificate[576]={
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-
-
-/* subject:/C=US/O=VeriSign, Inc./OU=Class 3 Public Primary Certification Authority - G2/OU=(c) 1998 VeriSign, Inc. - For authorized use only/OU=VeriSign Trust Network */
-/* issuer :/C=US/O=VeriSign, Inc./OU=Class 3 Public Primary Certification Authority - G2/OU=(c) 1998 VeriSign, Inc. - For authorized use only/OU=VeriSign Trust Network */
-
-
-const unsigned char Verisign_Class_3_Public_Primary_Certification_Authority___G2_certificate[774]={
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-};
-
-
-/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G3 */
-/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G3 */
-
-
-const unsigned char Verisign_Class_3_Public_Primary_Certification_Authority___G3_certificate[1054]={
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+const unsigned char GeoTrust_Primary_Certification_Authority_certificate[896]={
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-};
-
-
-/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2007 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G4 */
-/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2007 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 3 Public Primary Certification Authority - G4 */
-
-
-const unsigned char VeriSign_Class_3_Public_Primary_Certification_Authority___G4_certificate[904]={
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};
@@ -4553,380 +2091,2176 @@
};
-/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 4 Public Primary Certification Authority - G3 */
-/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 1999 VeriSign, Inc. - For authorized use only/CN=VeriSign Class 4 Public Primary Certification Authority - G3 */
+/* subject:/C=US/O=Equifax/OU=Equifax Secure Certificate Authority */
+/* issuer :/C=US/O=Equifax/OU=Equifax Secure Certificate Authority */
-const unsigned char Verisign_Class_4_Public_Primary_Certification_Authority___G3_certificate[1054]={
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+const unsigned char Equifax_Secure_CA_certificate[804]={
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+};
+
+
+/* subject:/O=Entrust.net/OU=www.entrust.net/CPS_2048 incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Certification Authority (2048) */
+/* issuer :/O=Entrust.net/OU=www.entrust.net/CPS_2048 incorp. by ref. (limits liab.)/OU=(c) 1999 Entrust.net Limited/CN=Entrust.net Certification Authority (2048) */
+
+
+const unsigned char Entrust_net_Premium_2048_Secure_Server_CA_certificate[1120]={
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};
-/* subject:/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2008 VeriSign, Inc. - For authorized use only/CN=VeriSign Universal Root Certification Authority */
-/* issuer :/C=US/O=VeriSign, Inc./OU=VeriSign Trust Network/OU=(c) 2008 VeriSign, Inc. - For authorized use only/CN=VeriSign Universal Root Certification Authority */
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root G3 */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root G3 */
-const unsigned char VeriSign_Universal_Root_Certification_Authority_certificate[1213]={
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-0x95,0x14,0x0E,0x7D,0xCD,0x95,0x34,0xD9,0xD2,0x8F,0x70,0x73,0x81,0x7B,0x9C,0x7E,
-0xBD,0x98,0x61,0xD8,0x45,0x87,0x98,0x90,0xC5,0xEB,0x86,0x30,0xC6,0x35,0xBF,0xF0,
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-0xEC,0xCD,0x82,0x61,0xF1,0x38,0xE6,0x4F,0x97,0x98,0x2A,0x5A,0x8D,
+const unsigned char DigiCert_Assured_ID_Root_G3_certificate[586]={
+0x30,0x82,0x02,0x46,0x30,0x82,0x01,0xCD,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x0B,
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+0xEA,0x96,0x63,0x6A,0x65,0x45,0x92,0x95,0x01,0xB4,
};
-/* subject:/C=US/OU=www.xrampsecurity.com/O=XRamp Security Services Inc/CN=XRamp Global Certification Authority */
-/* issuer :/C=US/OU=www.xrampsecurity.com/O=XRamp Security Services Inc/CN=XRamp Global Certification Authority */
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO Certification Authority */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO Certification Authority */
-const unsigned char XRamp_Global_CA_Root_certificate[1076]={
-0x30,0x82,0x04,0x30,0x30,0x82,0x03,0x18,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x50,
-0x94,0x6C,0xEC,0x18,0xEA,0xD5,0x9C,0x4D,0xD5,0x97,0xEF,0x75,0x8F,0xA0,0xAD,0x30,
+const unsigned char COMODO_Certification_Authority_certificate[1057]={
+0x30,0x82,0x04,0x1D,0x30,0x82,0x03,0x05,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x4E,
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+0x04,0x0E,0xBE,0x0B,0x83,0xA6,0x17,0x83,0xB9,0x30,0x12,0xA0,0xC5,0x33,0x15,0x05,
+0xB9,0x0D,0xFB,0xC7,0x05,0x76,0xE3,0xD8,0x4A,0x8D,0xFC,0x34,0x17,0xA3,0xC6,0x21,
+0x28,0xBE,0x30,0x45,0x31,0x1E,0xC7,0x78,0xBE,0x58,0x61,0x38,0xAC,0x3B,0xE2,0x01,
+0x65,
+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root CA */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Global Root CA */
+
+
+const unsigned char DigiCert_Global_Root_CA_certificate[947]={
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+0x95,0x6D,0xDE,
+};
+
+
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=AAA Certificate Services */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=AAA Certificate Services */
+
+
+const unsigned char Comodo_AAA_Services_root_certificate[1078]={
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+0xE3,0x2B,0x38,0x3A,0x09,0x81,0xB5,0x6B,0x85,0xD7,0xBE,0x2D,0xED,0x3F,0x1A,0xB7,
+0xB2,0x63,0xE2,0xF5,0x62,0x2C,0x82,0xD4,0x6A,0x00,0x41,0x50,0xF1,0x39,0x83,0x9F,
+0x95,0xE9,0x36,0x96,0x98,0x6E,
+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert High Assurance EV Root CA */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert High Assurance EV Root CA */
+
+
+const unsigned char DigiCert_High_Assurance_EV_Root_CA_certificate[969]={
+0x30,0x82,0x03,0xC5,0x30,0x82,0x02,0xAD,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x02,
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+0xF7,0xEA,0x21,0xB9,0x68,0x12,0x9C,0x3C,0x9C,0xBF,0x9E,0xFC,0x80,0x5C,0x9B,0x63,
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+0xF8,0xE9,0x2E,0x13,0xA3,0x77,0xE8,0x1F,0x4A,
+};
+
+
+/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA */
+/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Universal CA */
+
+
+const unsigned char GeoTrust_Universal_CA_certificate[1388]={
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+0x6C,0x9C,0x4F,0x81,0x6A,0x7A,0x64,0xAC,0xBB,0x3E,0xD5,0xE7,0xCB,0x76,0x2E,0xC5,
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+0xA4,0xE6,0x8E,0xD8,0xF9,0x29,0x48,0x8A,0xCE,0x73,0xFE,0x2C,
+};
+
+
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO ECC Certification Authority */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO ECC Certification Authority */
+
+
+const unsigned char COMODO_ECC_Certification_Authority_certificate[653]={
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+0x59,0x5B,0xA5,0x5B,0x0B,0x33,0x99,0xE2,0xC3,0x3D,0x89,0xF9,0x6A,0x2C,0xEF,0xB2,
+0xD3,0x06,0xE9,0xA3,0x42,0x30,0x40,0x30,0x1D,0x06,0x03,0x55,0x1D,0x0E,0x04,0x16,
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+0xC4,0x48,0x77,0x99,0xD3,0x79,0x30,0x0E,0x06,0x03,0x55,0x1D,0x0F,0x01,0x01,0xFF,
+0x04,0x04,0x03,0x02,0x01,0x06,0x30,0x0F,0x06,0x03,0x55,0x1D,0x13,0x01,0x01,0xFF,
+0x04,0x05,0x30,0x03,0x01,0x01,0xFF,0x30,0x0A,0x06,0x08,0x2A,0x86,0x48,0xCE,0x3D,
+0x04,0x03,0x03,0x03,0x68,0x00,0x30,0x65,0x02,0x31,0x00,0xEF,0x03,0x5B,0x7A,0xAC,
+0xB7,0x78,0x0A,0x72,0xB7,0x88,0xDF,0xFF,0xB5,0x46,0x14,0x09,0x0A,0xFA,0xA0,0xE6,
+0x7D,0x08,0xC6,0x1A,0x87,0xBD,0x18,0xA8,0x73,0xBD,0x26,0xCA,0x60,0x0C,0x9D,0xCE,
+0x99,0x9F,0xCF,0x5C,0x0F,0x30,0xE1,0xBE,0x14,0x31,0xEA,0x02,0x30,0x14,0xF4,0x93,
+0x3C,0x49,0xA7,0x33,0x7A,0x90,0x46,0x47,0xB3,0x63,0x7D,0x13,0x9B,0x4E,0xB7,0x6F,
+0x18,0x37,0x80,0x53,0xFE,0xDD,0x20,0xE0,0x35,0x9A,0x36,0xD1,0xC7,0x01,0xB9,0xE6,
+0xDC,0xDD,0xF3,0xFF,0x1D,0x2C,0x3A,0x16,0x57,0xD9,0x92,0x39,0xD6,
+};
+
+
+/* subject:/C=US/O=Entrust, Inc./OU=See www.entrust.net/legal-terms/OU=(c) 2009 Entrust, Inc. - for authorized use only/CN=Entrust Root Certification Authority - G2 */
+/* issuer :/C=US/O=Entrust, Inc./OU=See www.entrust.net/legal-terms/OU=(c) 2009 Entrust, Inc. - for authorized use only/CN=Entrust Root Certification Authority - G2 */
+
+
+const unsigned char Entrust_Root_Certification_Authority___G2_certificate[1090]={
+0x30,0x82,0x04,0x3E,0x30,0x82,0x03,0x26,0xA0,0x03,0x02,0x01,0x02,0x02,0x04,0x4A,
+0x53,0x8C,0x28,0x30,0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x0B,
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+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root G2 */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Assured ID Root G2 */
+
+
+const unsigned char DigiCert_Assured_ID_Root_G2_certificate[922]={
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+0x96,0x6C,0x0E,0x44,0x66,0x10,0x44,0x8F,0xAD,0x05,0xEB,0xF8,0x79,0xAC,0xA6,0x1B,
+0xE8,0x37,0x34,0x9D,0x53,0xC9,0x61,0xAA,0xA2,0x52,0xAF,0x4A,0x70,0x16,0x86,0xC2,
+0x3A,0xC8,0xB1,0x13,0x70,0x36,0xD8,0xCF,0xEE,0xF4,0x0A,0x34,0xD5,0x5B,0x4C,0xFD,
+0x07,0x9C,0xA2,0xBA,0xD9,0x01,0x72,0x5C,0xF3,0x4D,0xC1,0xDD,0x0E,0xB1,0x1C,0x0D,
+0xC4,0x63,0xBE,0xAD,0xF4,0x14,0xFB,0x89,0xEC,0xA2,0x41,0x0E,0x4C,0xCC,0xC8,0x57,
+0x40,0xD0,0x6E,0x03,0xAA,0xCD,0x0C,0x8E,0x89,0x99,0x99,0x6C,0xF0,0x3C,0x30,0xAF,
+0x38,0xDF,0x6F,0xBC,0xA3,0xBE,0x29,0x20,0x27,0xAB,0x74,0xFF,0x13,0x22,0x78,0xDE,
+0x97,0x52,0x55,0x1E,0x83,0xB5,0x54,0x20,0x03,0xEE,0xAE,0xC0,0x4F,0x56,0xDE,0x37,
+0xCC,0xC3,0x7F,0xAA,0x04,0x27,0xBB,0xD3,0x77,0xB8,0x62,0xDB,0x17,0x7C,0x9C,0x28,
+0x22,0x13,0x73,0x6C,0xCF,0x26,0xF5,0x8A,0x29,0xE7,
+};
+
+
+/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Commercial */
+/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Commercial */
+
+
+const unsigned char AffirmTrust_Commercial_certificate[848]={
+0x30,0x82,0x03,0x4C,0x30,0x82,0x02,0x34,0xA0,0x03,0x02,0x01,0x02,0x02,0x08,0x77,
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+0x0D,0x01,0x01,0x0B,0x05,0x00,0x30,0x44,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,
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+0xB7,0xAB,0x2C,0x4F,0xC6,0x41,0x32,0x94,0x57,0xFA,0x12,0xA7,0x5B,0x0E,0xE2,0x8F,
+0x1F,0x1E,0x86,0x19,0xA7,0xAA,0xB5,0x2D,0xB9,0x5F,0x0D,0x8A,0xC2,0xAF,0x85,0x35,
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+0x9A,0xAB,0x21,0x97,0x14,0xB1,0xDE,0x8C,0xD3,0xAB,0x89,0x15,0xD8,0x6B,0x24,0xD4,
+0xF1,0x16,0xAE,0xD8,0xA4,0x5C,0xD4,0x7F,0x51,0x8E,0xED,0x18,0x01,0xB1,0x93,0x63,
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+0x7D,0x27,0xA1,0xFE,0x6F,0xEC,0xB3,0x1E,0x24,0xDA,0xE3,0x4B,0x55,0x1A,0x00,0x3B,
+0x35,0xB4,0x3B,0xD9,0xD7,0x5D,0x30,0xFD,0x81,0x13,0x89,0xF2,0xC2,0x06,0x2B,0xED,
+0x67,0xC4,0x8E,0xC9,0x43,0xB2,0x5C,0x6B,0x15,0x89,0x02,0xBC,0x62,0xFC,0x4E,0xF2,
+0xB5,0x33,0xAA,0xB2,0x6F,0xD3,0x0A,0xA2,0x50,0xE3,0xF6,0x3B,0xE8,0x2E,0x44,0xC2,
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+0x47,0x45,0x47,0xB4,0xEF,0x6A,0x34,0x11,0xC9,0x4D,0x9A,0xCC,0x59,0xB7,0xD6,0x02,
+0x9E,0x5A,0x4E,0x65,0xB5,0x94,0xAE,0x1B,0xDF,0x29,0xB0,0x16,0xF1,0xBF,0x00,0x9E,
+0x07,0x3A,0x17,0x64,0xB5,0x04,0xB5,0x23,0x21,0x99,0x0A,0x95,0x3B,0x97,0x7C,0xEF,
+};
+
+
+/* subject:/C=US/O=AffirmTrust/CN=AffirmTrust Premium */
+/* issuer :/C=US/O=AffirmTrust/CN=AffirmTrust Premium */
+
+
+const unsigned char AffirmTrust_Premium_certificate[1354]={
+0x30,0x82,0x05,0x46,0x30,0x82,0x03,0x2E,0xA0,0x03,0x02,0x01,0x02,0x02,0x08,0x6D,
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+0x0D,0x01,0x01,0x0C,0x05,0x00,0x30,0x41,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,
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+0x29,0xE0,0xB6,0xB8,0x09,0x68,0x19,0x1C,0x18,0x43,
+};
+
+
+/* subject:/C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./CN=Go Daddy Root Certificate Authority - G2 */
+/* issuer :/C=US/ST=Arizona/L=Scottsdale/O=GoDaddy.com, Inc./CN=Go Daddy Root Certificate Authority - G2 */
+
+
+const unsigned char Go_Daddy_Root_Certificate_Authority___G2_certificate[969]={
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+0xE2,0xE2,0x44,0xBE,0x5C,0xF7,0xEA,0x1C,0xF5,
+};
+
+
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Secure Certificate Services */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Secure Certificate Services */
+
+
+const unsigned char Comodo_Secure_Services_root_certificate[1091]={
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+};
+
+
+/* subject:/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Trusted Root G4 */
+/* issuer :/C=US/O=DigiCert Inc/OU=www.digicert.com/CN=DigiCert Trusted Root G4 */
+
+
+const unsigned char DigiCert_Trusted_Root_G4_certificate[1428]={
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+0x49,0x5D,0x44,0x63,0xE0,0xAD,0x9C,0x96,0x18,0x66,0x1B,0x23,0x0D,0x3D,0x79,0xE9,
+0x6D,0x6B,0xD6,0x54,0xF8,0xD2,0x3C,0xC1,0x43,0x40,0xAE,0x1D,0x50,0xF5,0x52,0xFC,
+0x90,0x3B,0xBB,0x98,0x99,0x69,0x6B,0xC7,0xC1,0xA7,0xA8,0x68,0xA4,0x27,0xDC,0x9D,
+0xF9,0x27,0xAE,0x30,0x85,0xB9,0xF6,0x67,0x4D,0x3A,0x3E,0x8F,0x59,0x39,0x22,0x53,
+0x44,0xEB,0xC8,0x5D,0x03,0xCA,0xED,0x50,0x7A,0x7D,0x62,0x21,0x0A,0x80,0xC8,0x73,
+0x66,0xD1,0xA0,0x05,0x60,0x5F,0xE8,0xA5,0xB4,0xA7,0xAF,0xA8,0xF7,0x6D,0x35,0x9C,
+0x7C,0x5A,0x8A,0xD6,0xA2,0x38,0x99,0xF3,0x78,0x8B,0xF4,0x4D,0xD2,0x20,0x0B,0xDE,
+0x04,0xEE,0x8C,0x9B,0x47,0x81,0x72,0x0D,0xC0,0x14,0x32,0xEF,0x30,0x59,0x2E,0xAE,
+0xE0,0x71,0xF2,0x56,0xE4,0x6A,0x97,0x6F,0x92,0x50,0x6D,0x96,0x8D,0x68,0x7A,0x9A,
+0xB2,0x36,0x14,0x7A,0x06,0xF2,0x24,0xB9,0x09,0x11,0x50,0xD7,0x08,0xB1,0xB8,0x89,
+0x7A,0x84,0x23,0x61,0x42,0x29,0xE5,0xA3,0xCD,0xA2,0x20,0x41,0xD7,0xD1,0x9C,0x64,
+0xD9,0xEA,0x26,0xA1,0x8B,0x14,0xD7,0x4C,0x19,0xB2,0x50,0x41,0x71,0x3D,0x3F,0x4D,
+0x70,0x23,0x86,0x0C,0x4A,0xDC,0x81,0xD2,0xCC,0x32,0x94,0x84,0x0D,0x08,0x09,0x97,
+0x1C,0x4F,0xC0,0xEE,0x6B,0x20,0x74,0x30,0xD2,0xE0,0x39,0x34,0x10,0x85,0x21,0x15,
+0x01,0x08,0xE8,0x55,0x32,0xDE,0x71,0x49,0xD9,0x28,0x17,0x50,0x4D,0xE6,0xBE,0x4D,
+0xD1,0x75,0xAC,0xD0,0xCA,0xFB,0x41,0xB8,0x43,0xA5,0xAA,0xD3,0xC3,0x05,0x44,0x4F,
+0x2C,0x36,0x9B,0xE2,0xFA,0xE2,0x45,0xB8,0x23,0x53,0x6C,0x06,0x6F,0x67,0x55,0x7F,
+0x46,0xB5,0x4C,0x3F,0x6E,0x28,0x5A,0x79,0x26,0xD2,0xA4,0xA8,0x62,0x97,0xD2,0x1E,
+0xE2,0xED,0x4A,0x8B,0xBC,0x1B,0xFD,0x47,0x4A,0x0D,0xDF,0x67,0x66,0x7E,0xB2,0x5B,
+0x41,0xD0,0x3B,0xE4,0xF4,0x3B,0xF4,0x04,0x63,0xE9,0xEF,0xC2,0x54,0x00,0x51,0xA0,
+0x8A,0x2A,0xC9,0xCE,0x78,0xCC,0xD5,0xEA,0x87,0x04,0x18,0xB3,0xCE,0xAF,0x49,0x88,
+0xAF,0xF3,0x92,0x99,0xB6,0xB3,0xE6,0x61,0x0F,0xD2,0x85,0x00,0xE7,0x50,0x1A,0xE4,
+0x1B,0x95,0x9D,0x19,0xA1,0xB9,0x9C,0xB1,0x9B,0xB1,0x00,0x1E,0xEF,0xD0,0x0F,0x4F,
+0x42,0x6C,0xC9,0x0A,0xBC,0xEE,0x43,0xFA,0x3A,0x71,0xA5,0xC8,0x4D,0x26,0xA5,0x35,
+0xFD,0x89,0x5D,0xBC,0x85,0x62,0x1D,0x32,0xD2,0xA0,0x2B,0x54,0xED,0x9A,0x57,0xC1,
+0xDB,0xFA,0x10,0xCF,0x19,0xB7,0x8B,0x4A,0x1B,0x8F,0x01,0xB6,0x27,0x95,0x53,0xE8,
+0xB6,0x89,0x6D,0x5B,0xBC,0x68,0xD4,0x23,0xE8,0x8B,0x51,0xA2,0x56,0xF9,0xF0,0xA6,
+0x80,0xA0,0xD6,0x1E,0xB3,0xBC,0x0F,0x0F,0x53,0x75,0x29,0xAA,0xEA,0x13,0x77,0xE4,
+0xDE,0x8C,0x81,0x21,0xAD,0x07,0x10,0x47,0x11,0xAD,0x87,0x3D,0x07,0xD1,0x75,0xBC,
+0xCF,0xF3,0x66,0x7E,
+};
+
+
+/* subject:/OU=GlobalSign ECC Root CA - R5/O=GlobalSign/CN=GlobalSign */
+/* issuer :/OU=GlobalSign ECC Root CA - R5/O=GlobalSign/CN=GlobalSign */
+
+
+const unsigned char GlobalSign_ECC_Root_CA___R5_certificate[546]={
+0x30,0x82,0x02,0x1E,0x30,0x82,0x01,0xA4,0xA0,0x03,0x02,0x01,0x02,0x02,0x11,0x60,
+0x59,0x49,0xE0,0x26,0x2E,0xBB,0x55,0xF9,0x0A,0x77,0x8A,0x71,0xF9,0x4A,0xD8,0x6C,
+0x30,0x0A,0x06,0x08,0x2A,0x86,0x48,0xCE,0x3D,0x04,0x03,0x03,0x30,0x50,0x31,0x24,
+0x30,0x22,0x06,0x03,0x55,0x04,0x0B,0x13,0x1B,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,
+0x69,0x67,0x6E,0x20,0x45,0x43,0x43,0x20,0x52,0x6F,0x6F,0x74,0x20,0x43,0x41,0x20,
+0x2D,0x20,0x52,0x35,0x31,0x13,0x30,0x11,0x06,0x03,0x55,0x04,0x0A,0x13,0x0A,0x47,
+0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,0x31,0x13,0x30,0x11,0x06,0x03,0x55,
+0x04,0x03,0x13,0x0A,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,0x30,0x1E,
+0x17,0x0D,0x31,0x32,0x31,0x31,0x31,0x33,0x30,0x30,0x30,0x30,0x30,0x30,0x5A,0x17,
+0x0D,0x33,0x38,0x30,0x31,0x31,0x39,0x30,0x33,0x31,0x34,0x30,0x37,0x5A,0x30,0x50,
+0x31,0x24,0x30,0x22,0x06,0x03,0x55,0x04,0x0B,0x13,0x1B,0x47,0x6C,0x6F,0x62,0x61,
+0x6C,0x53,0x69,0x67,0x6E,0x20,0x45,0x43,0x43,0x20,0x52,0x6F,0x6F,0x74,0x20,0x43,
+0x41,0x20,0x2D,0x20,0x52,0x35,0x31,0x13,0x30,0x11,0x06,0x03,0x55,0x04,0x0A,0x13,
+0x0A,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,0x31,0x13,0x30,0x11,0x06,
+0x03,0x55,0x04,0x03,0x13,0x0A,0x47,0x6C,0x6F,0x62,0x61,0x6C,0x53,0x69,0x67,0x6E,
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+0x81,0x04,0x00,0x22,0x03,0x62,0x00,0x04,0x47,0x45,0x0E,0x96,0xFB,0x7D,0x5D,0xBF,
+0xE9,0x39,0xD1,0x21,0xF8,0x9F,0x0B,0xB6,0xD5,0x7B,0x1E,0x92,0x3A,0x48,0x59,0x1C,
+0xF0,0x62,0x31,0x2D,0xC0,0x7A,0x28,0xFE,0x1A,0xA7,0x5C,0xB3,0xB6,0xCC,0x97,0xE7,
+0x45,0xD4,0x58,0xFA,0xD1,0x77,0x6D,0x43,0xA2,0xC0,0x87,0x65,0x34,0x0A,0x1F,0x7A,
+0xDD,0xEB,0x3C,0x33,0xA1,0xC5,0x9D,0x4D,0xA4,0x6F,0x41,0x95,0x38,0x7F,0xC9,0x1E,
+0x84,0xEB,0xD1,0x9E,0x49,0x92,0x87,0x94,0x87,0x0C,0x3A,0x85,0x4A,0x66,0x9F,0x9D,
+0x59,0x93,0x4D,0x97,0x61,0x06,0x86,0x4A,0xA3,0x42,0x30,0x40,0x30,0x0E,0x06,0x03,
+0x55,0x1D,0x0F,0x01,0x01,0xFF,0x04,0x04,0x03,0x02,0x01,0x06,0x30,0x0F,0x06,0x03,
+0x55,0x1D,0x13,0x01,0x01,0xFF,0x04,0x05,0x30,0x03,0x01,0x01,0xFF,0x30,0x1D,0x06,
+0x03,0x55,0x1D,0x0E,0x04,0x16,0x04,0x14,0x3D,0xE6,0x29,0x48,0x9B,0xEA,0x07,0xCA,
+0x21,0x44,0x4A,0x26,0xDE,0x6E,0xDE,0xD2,0x83,0xD0,0x9F,0x59,0x30,0x0A,0x06,0x08,
+0x2A,0x86,0x48,0xCE,0x3D,0x04,0x03,0x03,0x03,0x68,0x00,0x30,0x65,0x02,0x31,0x00,
+0xE5,0x69,0x12,0xC9,0x6E,0xDB,0xC6,0x31,0xBA,0x09,0x41,0xE1,0x97,0xF8,0xFB,0xFD,
+0x9A,0xE2,0x7D,0x12,0xC9,0xED,0x7C,0x64,0xD3,0xCB,0x05,0x25,0x8B,0x56,0xD9,0xA0,
+0xE7,0x5E,0x5D,0x4E,0x0B,0x83,0x9C,0x5B,0x76,0x29,0xA0,0x09,0x26,0x21,0x6A,0x62,
+0x02,0x30,0x71,0xD2,0xB5,0x8F,0x5C,0xEA,0x3B,0xE1,0x78,0x09,0x85,0xA8,0x75,0x92,
+0x3B,0xC8,0x5C,0xFD,0x48,0xEF,0x0D,0x74,0x22,0xA8,0x08,0xE2,0x6E,0xC5,0x49,0xCE,
+0xC7,0x0C,0xBC,0xA7,0x61,0x69,0xF1,0xF7,0x3B,0xE1,0x2A,0xCB,0xF9,0x2B,0xF3,0x66,
+0x90,0x37,
+};
+
+
+/* subject:/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN-USERFirst-Hardware */
+/* issuer :/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN-USERFirst-Hardware */
+
+
+const unsigned char UTN_USERFirst_Hardware_Root_CA_certificate[1144]={
+0x30,0x82,0x04,0x74,0x30,0x82,0x03,0x5C,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x44,
+0xBE,0x0C,0x8B,0x50,0x00,0x24,0xB4,0x11,0xD3,0x36,0x2A,0xFE,0x65,0x0A,0xFD,0x30,
+0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x05,0x05,0x00,0x30,0x81,
+0x97,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x55,0x53,0x31,0x0B,
+0x30,0x09,0x06,0x03,0x55,0x04,0x08,0x13,0x02,0x55,0x54,0x31,0x17,0x30,0x15,0x06,
+0x03,0x55,0x04,0x07,0x13,0x0E,0x53,0x61,0x6C,0x74,0x20,0x4C,0x61,0x6B,0x65,0x20,
+0x43,0x69,0x74,0x79,0x31,0x1E,0x30,0x1C,0x06,0x03,0x55,0x04,0x0A,0x13,0x15,0x54,
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+0x32,0x9C,0x1E,0xBB,0x9D,0xF8,0x66,0xA8,
+};
+
+
+/* subject:/OU=GlobalSign ECC Root CA - R4/O=GlobalSign/CN=GlobalSign */
+/* issuer :/OU=GlobalSign ECC Root CA - R4/O=GlobalSign/CN=GlobalSign */
+
+
+const unsigned char GlobalSign_ECC_Root_CA___R4_certificate[485]={
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+0x7B,0x0B,0xF8,0x9F,0x84,
+};
+
+
+/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA I */
+/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Universal CA/CN=TC TrustCenter Universal CA I */
+
+
+const unsigned char TC_TrustCenter_Universal_CA_I_certificate[993]={
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+0x98,
+};
+
+
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Trusted Certificate Services */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=Comodo CA Limited/CN=Trusted Certificate Services */
+
+
+const unsigned char Comodo_Trusted_Services_root_certificate[1095]={
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+};
+
+
+/* subject:/C=US/O=Entrust, Inc./OU=www.entrust.net/CPS is incorporated by reference/OU=(c) 2006 Entrust, Inc./CN=Entrust Root Certification Authority */
+/* issuer :/C=US/O=Entrust, Inc./OU=www.entrust.net/CPS is incorporated by reference/OU=(c) 2006 Entrust, Inc./CN=Entrust Root Certification Authority */
+
+
+const unsigned char Entrust_Root_Certification_Authority_certificate[1173]={
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+};
+
+
+/* subject:/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 2 CA/CN=TC TrustCenter Class 2 CA II */
+/* issuer :/C=DE/O=TC TrustCenter GmbH/OU=TC TrustCenter Class 2 CA/CN=TC TrustCenter Class 2 CA II */
+
+
+const unsigned char TC_TrustCenter_Class_2_CA_II_certificate[1198]={
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+};
+
+
+/* subject:/O=Cybertrust, Inc/CN=Cybertrust Global Root */
+/* issuer :/O=Cybertrust, Inc/CN=Cybertrust Global Root */
+
+
+const unsigned char Cybertrust_Global_Root_certificate[933]={
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+};
+
+
+/* subject:/C=US/O=Entrust, Inc./OU=See www.entrust.net/legal-terms/OU=(c) 2012 Entrust, Inc. - for authorized use only/CN=Entrust Root Certification Authority - EC1 */
+/* issuer :/C=US/O=Entrust, Inc./OU=See www.entrust.net/legal-terms/OU=(c) 2012 Entrust, Inc. - for authorized use only/CN=Entrust Root Certification Authority - EC1 */
+
+
+const unsigned char Entrust_Root_Certification_Authority___EC1_certificate[765]={
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+};
+
+
+/* subject:/C=US/O=GeoTrust Inc./OU=(c) 2007 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G2 */
+/* issuer :/C=US/O=GeoTrust Inc./OU=(c) 2007 GeoTrust Inc. - For authorized use only/CN=GeoTrust Primary Certification Authority - G2 */
+
+
+const unsigned char GeoTrust_Primary_Certification_Authority___G2_certificate[690]={
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+};
+
+
+/* subject:/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA 2 */
+/* issuer :/C=US/O=GeoTrust Inc./CN=GeoTrust Global CA 2 */
+
+
+const unsigned char GeoTrust_Global_CA_2_certificate[874]={
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+};
+
+
+/* subject:/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO RSA Certification Authority */
+/* issuer :/C=GB/ST=Greater Manchester/L=Salford/O=COMODO CA Limited/CN=COMODO RSA Certification Authority */
+
+
+const unsigned char COMODO_RSA_Certification_Authority_certificate[1500]={
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+0x4B,0x46,0x63,0xFE,0x00,0x34,0x40,0x70,0xC1,0xC3,0xB9,0xA1,0xDD,0xA6,0x70,0xE2,
+0x04,0xB3,0x41,0xBC,0xE9,0x80,0x91,0xEA,0x64,0x9C,0x7A,0xE1,0x22,0x03,0xA9,0x9C,
+0x6E,0x6F,0x0E,0x65,0x4F,0x6C,0x87,0x87,0x5E,0xF3,0x6E,0xA0,0xF9,0x75,0xA5,0x9B,
+0x40,0xE8,0x53,0xB2,0x27,0x9D,0x4A,0xB9,0xC0,0x77,0x21,0x8D,0xFF,0x87,0xF2,0xDE,
+0xBC,0x8C,0xEF,0x17,0xDF,0xB7,0x49,0x0B,0xD1,0xF2,0x6E,0x30,0x0B,0x1A,0x0E,0x4E,
+0x76,0xED,0x11,0xFC,0xF5,0xE9,0x56,0xB2,0x7D,0xBF,0xC7,0x6D,0x0A,0x93,0x8C,0xA5,
+0xD0,0xC0,0xB6,0x1D,0xBE,0x3A,0x4E,0x94,0xA2,0xD7,0x6E,0x6C,0x0B,0xC2,0x8A,0x7C,
+0xFA,0x20,0xF3,0xC4,0xE4,0xE5,0xCD,0x0D,0xA8,0xCB,0x91,0x92,0xB1,0x7C,0x85,0xEC,
+0xB5,0x14,0x69,0x66,0x0E,0x82,0xE7,0xCD,0xCE,0xC8,0x2D,0xA6,0x51,0x7F,0x21,0xC1,
+0x35,0x53,0x85,0x06,0x4A,0x5D,0x9F,0xAD,0xBB,0x1B,0x5F,0x74,
+};
+
+
+/* subject:/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN - DATACorp SGC */
+/* issuer :/C=US/ST=UT/L=Salt Lake City/O=The USERTRUST Network/OU=http://www.usertrust.com/CN=UTN - DATACorp SGC */
+
+
+const unsigned char UTN_DATACorp_SGC_Root_CA_certificate[1122]={
+0x30,0x82,0x04,0x5E,0x30,0x82,0x03,0x46,0xA0,0x03,0x02,0x01,0x02,0x02,0x10,0x44,
+0xBE,0x0C,0x8B,0x50,0x00,0x21,0xB4,0x11,0xD3,0x2A,0x68,0x06,0xA9,0xAD,0x69,0x30,
+0x0D,0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x05,0x05,0x00,0x30,0x81,
+0x93,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,0x13,0x02,0x55,0x53,0x31,0x0B,
+0x30,0x09,0x06,0x03,0x55,0x04,0x08,0x13,0x02,0x55,0x54,0x31,0x17,0x30,0x15,0x06,
+0x03,0x55,0x04,0x07,0x13,0x0E,0x53,0x61,0x6C,0x74,0x20,0x4C,0x61,0x6B,0x65,0x20,
+0x43,0x69,0x74,0x79,0x31,0x1E,0x30,0x1C,0x06,0x03,0x55,0x04,0x0A,0x13,0x15,0x54,
+0x68,0x65,0x20,0x55,0x53,0x45,0x52,0x54,0x52,0x55,0x53,0x54,0x20,0x4E,0x65,0x74,
+0x77,0x6F,0x72,0x6B,0x31,0x21,0x30,0x1F,0x06,0x03,0x55,0x04,0x0B,0x13,0x18,0x68,
+0x74,0x74,0x70,0x3A,0x2F,0x2F,0x77,0x77,0x77,0x2E,0x75,0x73,0x65,0x72,0x74,0x72,
+0x75,0x73,0x74,0x2E,0x63,0x6F,0x6D,0x31,0x1B,0x30,0x19,0x06,0x03,0x55,0x04,0x03,
+0x13,0x12,0x55,0x54,0x4E,0x20,0x2D,0x20,0x44,0x41,0x54,0x41,0x43,0x6F,0x72,0x70,
+0x20,0x53,0x47,0x43,0x30,0x1E,0x17,0x0D,0x39,0x39,0x30,0x36,0x32,0x34,0x31,0x38,
+0x35,0x37,0x32,0x31,0x5A,0x17,0x0D,0x31,0x39,0x30,0x36,0x32,0x34,0x31,0x39,0x30,
+0x36,0x33,0x30,0x5A,0x30,0x81,0x93,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x06,
+0x13,0x02,0x55,0x53,0x31,0x0B,0x30,0x09,0x06,0x03,0x55,0x04,0x08,0x13,0x02,0x55,
+0x54,0x31,0x17,0x30,0x15,0x06,0x03,0x55,0x04,0x07,0x13,0x0E,0x53,0x61,0x6C,0x74,
+0x20,0x4C,0x61,0x6B,0x65,0x20,0x43,0x69,0x74,0x79,0x31,0x1E,0x30,0x1C,0x06,0x03,
+0x55,0x04,0x0A,0x13,0x15,0x54,0x68,0x65,0x20,0x55,0x53,0x45,0x52,0x54,0x52,0x55,
+0x53,0x54,0x20,0x4E,0x65,0x74,0x77,0x6F,0x72,0x6B,0x31,0x21,0x30,0x1F,0x06,0x03,
+0x55,0x04,0x0B,0x13,0x18,0x68,0x74,0x74,0x70,0x3A,0x2F,0x2F,0x77,0x77,0x77,0x2E,
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+0x19,0x06,0x03,0x55,0x04,0x03,0x13,0x12,0x55,0x54,0x4E,0x20,0x2D,0x20,0x44,0x41,
+0x54,0x41,0x43,0x6F,0x72,0x70,0x20,0x53,0x47,0x43,0x30,0x82,0x01,0x22,0x30,0x0D,
+0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x01,0x05,0x00,0x03,0x82,0x01,
+0x0F,0x00,0x30,0x82,0x01,0x0A,0x02,0x82,0x01,0x01,0x00,0xDF,0xEE,0x58,0x10,0xA2,
+0x2B,0x6E,0x55,0xC4,0x8E,0xBF,0x2E,0x46,0x09,0xE7,0xE0,0x08,0x0F,0x2E,0x2B,0x7A,
+0x13,0x94,0x1B,0xBD,0xF6,0xB6,0x80,0x8E,0x65,0x05,0x93,0x00,0x1E,0xBC,0xAF,0xE2,
+0x0F,0x8E,0x19,0x0D,0x12,0x47,0xEC,0xAC,0xAD,0xA3,0xFA,0x2E,0x70,0xF8,0xDE,0x6E,
+0xFB,0x56,0x42,0x15,0x9E,0x2E,0x5C,0xEF,0x23,0xDE,0x21,0xB9,0x05,0x76,0x27,0x19,
+0x0F,0x4F,0xD6,0xC3,0x9C,0xB4,0xBE,0x94,0x19,0x63,0xF2,0xA6,0x11,0x0A,0xEB,0x53,
+0x48,0x9C,0xBE,0xF2,0x29,0x3B,0x16,0xE8,0x1A,0xA0,0x4C,0xA6,0xC9,0xF4,0x18,0x59,
+0x68,0xC0,0x70,0xF2,0x53,0x00,0xC0,0x5E,0x50,0x82,0xA5,0x56,0x6F,0x36,0xF9,0x4A,
+0xE0,0x44,0x86,0xA0,0x4D,0x4E,0xD6,0x47,0x6E,0x49,0x4A,0xCB,0x67,0xD7,0xA6,0xC4,
+0x05,0xB9,0x8E,0x1E,0xF4,0xFC,0xFF,0xCD,0xE7,0x36,0xE0,0x9C,0x05,0x6C,0xB2,0x33,
+0x22,0x15,0xD0,0xB4,0xE0,0xCC,0x17,0xC0,0xB2,0xC0,0xF4,0xFE,0x32,0x3F,0x29,0x2A,
+0x95,0x7B,0xD8,0xF2,0xA7,0x4E,0x0F,0x54,0x7C,0xA1,0x0D,0x80,0xB3,0x09,0x03,0xC1,
+0xFF,0x5C,0xDD,0x5E,0x9A,0x3E,0xBC,0xAE,0xBC,0x47,0x8A,0x6A,0xAE,0x71,0xCA,0x1F,
+0xB1,0x2A,0xB8,0x5F,0x42,0x05,0x0B,0xEC,0x46,0x30,0xD1,0x72,0x0B,0xCA,0xE9,0x56,
+0x6D,0xF5,0xEF,0xDF,0x78,0xBE,0x61,0xBA,0xB2,0xA5,0xAE,0x04,0x4C,0xBC,0xA8,0xAC,
+0x69,0x15,0x97,0xBD,0xEF,0xEB,0xB4,0x8C,0xBF,0x35,0xF8,0xD4,0xC3,0xD1,0x28,0x0E,
+0x5C,0x3A,0x9F,0x70,0x18,0x33,0x20,0x77,0xC4,0xA2,0xAF,0x02,0x03,0x01,0x00,0x01,
+0xA3,0x81,0xAB,0x30,0x81,0xA8,0x30,0x0B,0x06,0x03,0x55,0x1D,0x0F,0x04,0x04,0x03,
+0x02,0x01,0xC6,0x30,0x0F,0x06,0x03,0x55,0x1D,0x13,0x01,0x01,0xFF,0x04,0x05,0x30,
+0x03,0x01,0x01,0xFF,0x30,0x1D,0x06,0x03,0x55,0x1D,0x0E,0x04,0x16,0x04,0x14,0x53,
+0x32,0xD1,0xB3,0xCF,0x7F,0xFA,0xE0,0xF1,0xA0,0x5D,0x85,0x4E,0x92,0xD2,0x9E,0x45,
+0x1D,0xB4,0x4F,0x30,0x3D,0x06,0x03,0x55,0x1D,0x1F,0x04,0x36,0x30,0x34,0x30,0x32,
+0xA0,0x30,0xA0,0x2E,0x86,0x2C,0x68,0x74,0x74,0x70,0x3A,0x2F,0x2F,0x63,0x72,0x6C,
+0x2E,0x75,0x73,0x65,0x72,0x74,0x72,0x75,0x73,0x74,0x2E,0x63,0x6F,0x6D,0x2F,0x55,
+0x54,0x4E,0x2D,0x44,0x41,0x54,0x41,0x43,0x6F,0x72,0x70,0x53,0x47,0x43,0x2E,0x63,
+0x72,0x6C,0x30,0x2A,0x06,0x03,0x55,0x1D,0x25,0x04,0x23,0x30,0x21,0x06,0x08,0x2B,
+0x06,0x01,0x05,0x05,0x07,0x03,0x01,0x06,0x0A,0x2B,0x06,0x01,0x04,0x01,0x82,0x37,
+0x0A,0x03,0x03,0x06,0x09,0x60,0x86,0x48,0x01,0x86,0xF8,0x42,0x04,0x01,0x30,0x0D,
+0x06,0x09,0x2A,0x86,0x48,0x86,0xF7,0x0D,0x01,0x01,0x05,0x05,0x00,0x03,0x82,0x01,
+0x01,0x00,0x27,0x35,0x97,0x00,0x8A,0x8B,0x28,0xBD,0xC6,0x33,0x30,0x1E,0x29,0xFC,
+0xE2,0xF7,0xD5,0x98,0xD4,0x40,0xBB,0x60,0xCA,0xBF,0xAB,0x17,0x2C,0x09,0x36,0x7F,
+0x50,0xFA,0x41,0xDC,0xAE,0x96,0x3A,0x0A,0x23,0x3E,0x89,0x59,0xC9,0xA3,0x07,0xED,
+0x1B,0x37,0xAD,0xFC,0x7C,0xBE,0x51,0x49,0x5A,0xDE,0x3A,0x0A,0x54,0x08,0x16,0x45,
+0xC2,0x99,0xB1,0x87,0xCD,0x8C,0x68,0xE0,0x69,0x03,0xE9,0xC4,0x4E,0x98,0xB2,0x3B,
+0x8C,0x16,0xB3,0x0E,0xA0,0x0C,0x98,0x50,0x9B,0x93,0xA9,0x70,0x09,0xC8,0x2C,0xA3,
+0x8F,0xDF,0x02,0xE4,0xE0,0x71,0x3A,0xF1,0xB4,0x23,0x72,0xA0,0xAA,0x01,0xDF,0xDF,
+0x98,0x3E,0x14,0x50,0xA0,0x31,0x26,0xBD,0x28,0xE9,0x5A,0x30,0x26,0x75,0xF9,0x7B,
+0x60,0x1C,0x8D,0xF3,0xCD,0x50,0x26,0x6D,0x04,0x27,0x9A,0xDF,0xD5,0x0D,0x45,0x47,
+0x29,0x6B,0x2C,0xE6,0x76,0xD9,0xA9,0x29,0x7D,0x32,0xDD,0xC9,0x36,0x3C,0xBD,0xAE,
+0x35,0xF1,0x11,0x9E,0x1D,0xBB,0x90,0x3F,0x12,0x47,0x4E,0x8E,0xD7,0x7E,0x0F,0x62,
+0x73,0x1D,0x52,0x26,0x38,0x1C,0x18,0x49,0xFD,0x30,0x74,0x9A,0xC4,0xE5,0x22,0x2F,
+0xD8,0xC0,0x8D,0xED,0x91,0x7A,0x4C,0x00,0x8F,0x72,0x7F,0x5D,0xDA,0xDD,0x1B,0x8B,
+0x45,0x6B,0xE7,0xDD,0x69,0x97,0xA8,0xC5,0x56,0x4C,0x0F,0x0C,0xF6,0x9F,0x7A,0x91,
+0x37,0xF6,0x97,0x82,0xE0,0xDD,0x71,0x69,0xFF,0x76,0x3F,0x60,0x4D,0x3C,0xCF,0xF7,
+0x99,0xF9,0xC6,0x57,0xF4,0xC9,0x55,0x39,0x78,0xBA,0x2C,0x79,0xC9,0xA6,0x88,0x2B,
+0xF4,0x08,
};
const unsigned char* kSSLCertCertificateList[] = {
- AddTrust_External_Root_certificate,
+ GlobalSign_Root_CA_certificate,
+ USERTrust_RSA_Certification_Authority_certificate,
+ Starfield_Class_2_CA_certificate,
+ Verisign_Class_3_Public_Primary_Certification_Authority___G3_certificate,
+ USERTrust_ECC_Certification_Authority_certificate,
+ GeoTrust_Global_CA_certificate,
+ Starfield_Root_Certificate_Authority___G2_certificate,
+ DigiCert_Global_Root_G3_certificate,
+ thawte_Primary_Root_CA___G2_certificate,
+ VeriSign_Universal_Root_Certification_Authority_certificate,
+ VeriSign_Class_3_Public_Primary_Certification_Authority___G4_certificate,
+ DigiCert_Global_Root_G2_certificate,
AddTrust_Low_Value_Services_Root_certificate,
+ AffirmTrust_Premium_ECC_certificate,
+ Verisign_Class_4_Public_Primary_Certification_Authority___G3_certificate,
+ thawte_Primary_Root_CA_certificate,
AddTrust_Public_Services_Root_certificate,
AddTrust_Qualified_Certificates_Root_certificate,
- AffirmTrust_Commercial_certificate,
- AffirmTrust_Networking_certificate,
- AffirmTrust_Premium_certificate,
- AffirmTrust_Premium_ECC_certificate,
- America_Online_Root_Certification_Authority_1_certificate,
- America_Online_Root_Certification_Authority_2_certificate,
- Baltimore_CyberTrust_Root_certificate,
- Comodo_AAA_Services_root_certificate,
- COMODO_Certification_Authority_certificate,
- COMODO_ECC_Certification_Authority_certificate,
- Comodo_Secure_Services_root_certificate,
- Comodo_Trusted_Services_root_certificate,
- Cybertrust_Global_Root_certificate,
- DigiCert_Assured_ID_Root_CA_certificate,
- DigiCert_Global_Root_CA_certificate,
- DigiCert_High_Assurance_EV_Root_CA_certificate,
- Entrust_net_Premium_2048_Secure_Server_CA_certificate,
- Entrust_net_Secure_Server_CA_certificate,
- Entrust_Root_Certification_Authority_certificate,
- Equifax_Secure_CA_certificate,
- Equifax_Secure_eBusiness_CA_1_certificate,
- Equifax_Secure_eBusiness_CA_2_certificate,
- Equifax_Secure_Global_eBusiness_CA_certificate,
- GeoTrust_Global_CA_certificate,
- GeoTrust_Global_CA_2_certificate,
- GeoTrust_Primary_Certification_Authority_certificate,
- GeoTrust_Primary_Certification_Authority___G2_certificate,
GeoTrust_Primary_Certification_Authority___G3_certificate,
- GeoTrust_Universal_CA_certificate,
GeoTrust_Universal_CA_2_certificate,
- GlobalSign_Root_CA_certificate,
+ Baltimore_CyberTrust_Root_certificate,
GlobalSign_Root_CA___R2_certificate,
GlobalSign_Root_CA___R3_certificate,
- Go_Daddy_Class_2_CA_certificate,
- Go_Daddy_Root_Certificate_Authority___G2_certificate,
- GTE_CyberTrust_Global_Root_certificate,
- Network_Solutions_Certificate_Authority_certificate,
- RSA_Root_Certificate_1_certificate,
- Starfield_Class_2_CA_certificate,
- Starfield_Root_Certificate_Authority___G2_certificate,
- Starfield_Services_Root_Certificate_Authority___G2_certificate,
- StartCom_Certification_Authority_certificate,
- StartCom_Certification_Authority_G2_certificate,
- TC_TrustCenter_Class_2_CA_II_certificate,
- TC_TrustCenter_Class_3_CA_II_certificate,
- TC_TrustCenter_Universal_CA_I_certificate,
- TC_TrustCenter_Universal_CA_III_certificate,
- Thawte_Premium_Server_CA_certificate,
- thawte_Primary_Root_CA_certificate,
- thawte_Primary_Root_CA___G2_certificate,
+ AffirmTrust_Networking_certificate,
+ AddTrust_External_Root_certificate,
thawte_Primary_Root_CA___G3_certificate,
- Thawte_Server_CA_certificate,
- UTN_DATACorp_SGC_Root_CA_certificate,
- UTN_USERFirst_Hardware_Root_CA_certificate,
- ValiCert_Class_1_VA_certificate,
- ValiCert_Class_2_VA_certificate,
- Verisign_Class_3_Public_Primary_Certification_Authority_certificate,
- Verisign_Class_3_Public_Primary_Certification_Authority___G2_certificate,
- Verisign_Class_3_Public_Primary_Certification_Authority___G3_certificate,
- VeriSign_Class_3_Public_Primary_Certification_Authority___G4_certificate,
+ DigiCert_Assured_ID_Root_CA_certificate,
+ Go_Daddy_Class_2_CA_certificate,
+ GeoTrust_Primary_Certification_Authority_certificate,
VeriSign_Class_3_Public_Primary_Certification_Authority___G5_certificate,
- Verisign_Class_4_Public_Primary_Certification_Authority___G3_certificate,
- VeriSign_Universal_Root_Certification_Authority_certificate,
- XRamp_Global_CA_Root_certificate,
+ Equifax_Secure_CA_certificate,
+ Entrust_net_Premium_2048_Secure_Server_CA_certificate,
+ DigiCert_Assured_ID_Root_G3_certificate,
+ COMODO_Certification_Authority_certificate,
+ DigiCert_Global_Root_CA_certificate,
+ Comodo_AAA_Services_root_certificate,
+ DigiCert_High_Assurance_EV_Root_CA_certificate,
+ GeoTrust_Universal_CA_certificate,
+ COMODO_ECC_Certification_Authority_certificate,
+ Entrust_Root_Certification_Authority___G2_certificate,
+ DigiCert_Assured_ID_Root_G2_certificate,
+ AffirmTrust_Commercial_certificate,
+ AffirmTrust_Premium_certificate,
+ Go_Daddy_Root_Certificate_Authority___G2_certificate,
+ Comodo_Secure_Services_root_certificate,
+ DigiCert_Trusted_Root_G4_certificate,
+ GlobalSign_ECC_Root_CA___R5_certificate,
+ UTN_USERFirst_Hardware_Root_CA_certificate,
+ GlobalSign_ECC_Root_CA___R4_certificate,
+ TC_TrustCenter_Universal_CA_I_certificate,
+ Comodo_Trusted_Services_root_certificate,
+ Entrust_Root_Certification_Authority_certificate,
+ TC_TrustCenter_Class_2_CA_II_certificate,
+ Cybertrust_Global_Root_certificate,
+ Entrust_Root_Certification_Authority___EC1_certificate,
+ GeoTrust_Primary_Certification_Authority___G2_certificate,
+ GeoTrust_Global_CA_2_certificate,
+ COMODO_RSA_Certification_Authority_certificate,
+ UTN_DATACorp_SGC_Root_CA_certificate,
};
const size_t kSSLCertCertificateSizeList[] = {
- 1082,
+ 889,
+ 1506,
+ 1043,
+ 1054,
+ 659,
+ 856,
+ 993,
+ 579,
+ 652,
+ 1213,
+ 904,
+ 914,
1052,
+ 514,
+ 1054,
+ 1060,
1049,
1058,
- 848,
- 848,
- 1354,
- 514,
- 936,
- 1448,
- 891,
- 1078,
- 1057,
- 653,
- 1091,
- 1095,
- 933,
- 955,
- 947,
- 969,
- 1120,
- 1244,
- 1173,
- 804,
- 646,
- 804,
- 660,
- 856,
- 874,
- 896,
- 690,
1026,
- 1388,
1392,
- 889,
+ 891,
958,
867,
- 1028,
- 969,
- 606,
- 1002,
- 747,
- 1043,
- 993,
- 1011,
- 1931,
- 1383,
- 1198,
- 1198,
- 993,
- 997,
- 811,
- 1060,
- 652,
+ 848,
+ 1082,
1070,
- 791,
- 1122,
- 1144,
- 747,
- 747,
- 576,
- 774,
- 1054,
- 904,
+ 955,
+ 1028,
+ 896,
1239,
- 1054,
- 1213,
- 1076,
+ 804,
+ 1120,
+ 586,
+ 1057,
+ 947,
+ 1078,
+ 969,
+ 1388,
+ 653,
+ 1090,
+ 922,
+ 848,
+ 1354,
+ 969,
+ 1091,
+ 1428,
+ 546,
+ 1144,
+ 485,
+ 993,
+ 1095,
+ 1173,
+ 1198,
+ 933,
+ 765,
+ 690,
+ 874,
+ 1500,
+ 1122,
};
-} // namspace rtc
diff --git a/webrtc/base/sslstreamadapter.cc b/webrtc/base/sslstreamadapter.cc
index b9ba9d3..a2cff3e 100644
--- a/webrtc/base/sslstreamadapter.cc
+++ b/webrtc/base/sslstreamadapter.cc
@@ -30,12 +30,20 @@
const char CS_AES_CM_128_HMAC_SHA1_80[] = "AES_CM_128_HMAC_SHA1_80";
const char CS_AES_CM_128_HMAC_SHA1_32[] = "AES_CM_128_HMAC_SHA1_32";
-int GetSrtpCryptoSuiteFromName(const std::string& cipher) {
- if (cipher == CS_AES_CM_128_HMAC_SHA1_32)
+std::string SrtpCryptoSuiteToName(int crypto_suite) {
+ if (crypto_suite == SRTP_AES128_CM_SHA1_32)
+ return CS_AES_CM_128_HMAC_SHA1_32;
+ if (crypto_suite == SRTP_AES128_CM_SHA1_80)
+ return CS_AES_CM_128_HMAC_SHA1_80;
+ return std::string();
+}
+
+int SrtpCryptoSuiteFromName(const std::string& crypto_suite) {
+ if (crypto_suite == CS_AES_CM_128_HMAC_SHA1_32)
return SRTP_AES128_CM_SHA1_32;
- if (cipher == CS_AES_CM_128_HMAC_SHA1_80)
+ if (crypto_suite == CS_AES_CM_128_HMAC_SHA1_80)
return SRTP_AES128_CM_SHA1_80;
- return 0;
+ return SRTP_INVALID_CRYPTO_SUITE;
}
SSLStreamAdapter* SSLStreamAdapter::Create(StreamInterface* stream) {
@@ -46,7 +54,7 @@
#endif // SSL_USE_OPENSSL
}
-bool SSLStreamAdapter::GetSslCipherSuite(int* cipher) {
+bool SSLStreamAdapter::GetSslCipherSuite(int* cipher_suite) {
return false;
}
@@ -59,12 +67,12 @@
return false; // Default is unsupported
}
-bool SSLStreamAdapter::SetDtlsSrtpCiphers(
- const std::vector<std::string>& ciphers) {
+bool SSLStreamAdapter::SetDtlsSrtpCryptoSuites(
+ const std::vector<int>& crypto_suites) {
return false;
}
-bool SSLStreamAdapter::GetDtlsSrtpCipher(std::string* cipher) {
+bool SSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) {
return false;
}
@@ -83,8 +91,8 @@
return OpenSSLStreamAdapter::GetDefaultSslCipherForTest(version, key_type);
}
-std::string SSLStreamAdapter::GetSslCipherSuiteName(int cipher) {
- return OpenSSLStreamAdapter::GetSslCipherSuiteName(cipher);
+std::string SSLStreamAdapter::SslCipherSuiteToName(int cipher_suite) {
+ return OpenSSLStreamAdapter::SslCipherSuiteToName(cipher_suite);
}
#endif // SSL_USE_OPENSSL
diff --git a/webrtc/base/sslstreamadapter.h b/webrtc/base/sslstreamadapter.h
index 65a7729..c57056b 100644
--- a/webrtc/base/sslstreamadapter.h
+++ b/webrtc/base/sslstreamadapter.h
@@ -19,7 +19,11 @@
namespace rtc {
+// Constants for SSL profile.
+const int TLS_NULL_WITH_NULL_NULL = 0;
+
// Constants for SRTP profiles.
+const int SRTP_INVALID_CRYPTO_SUITE = 0;
const int SRTP_AES128_CM_SHA1_80 = 0x0001;
const int SRTP_AES128_CM_SHA1_32 = 0x0002;
@@ -31,10 +35,13 @@
// 128-bit AES with 32-bit SHA-1 HMAC.
extern const char CS_AES_CM_128_HMAC_SHA1_32[];
-// Returns the DTLS-SRTP protection profile ID, as defined in
-// https://tools.ietf.org/html/rfc5764#section-4.1.2, for the given SRTP
-// Crypto-suite, as defined in https://tools.ietf.org/html/rfc4568#section-6.2
-int GetSrtpCryptoSuiteFromName(const std::string& cipher_rfc_name);
+// Given the DTLS-SRTP protection profile ID, as defined in
+// https://tools.ietf.org/html/rfc4568#section-6.2 , return the SRTP profile
+// name, as defined in https://tools.ietf.org/html/rfc5764#section-4.1.2.
+std::string SrtpCryptoSuiteToName(int crypto_suite);
+
+// The reverse of above conversion.
+int SrtpCryptoSuiteFromName(const std::string& crypto_suite);
// SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS.
// After SSL has been started, the stream will only open on successful
@@ -152,7 +159,7 @@
// Retrieves the IANA registration id of the cipher suite used for the
// connection (e.g. 0x2F for "TLS_RSA_WITH_AES_128_CBC_SHA").
- virtual bool GetSslCipherSuite(int* cipher);
+ virtual bool GetSslCipherSuite(int* cipher_suite);
// Key Exporter interface from RFC 5705
// Arguments are:
@@ -174,8 +181,8 @@
size_t result_len);
// DTLS-SRTP interface
- virtual bool SetDtlsSrtpCiphers(const std::vector<std::string>& ciphers);
- virtual bool GetDtlsSrtpCipher(std::string* cipher);
+ virtual bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites);
+ virtual bool GetDtlsSrtpCryptoSuite(int* crypto_suite);
// Capabilities testing
static bool HaveDtls();
@@ -191,7 +198,7 @@
// TODO(guoweis): Move this away from a static class method. Currently this is
// introduced such that any caller could depend on sslstreamadapter.h without
// depending on specific SSL implementation.
- static std::string GetSslCipherSuiteName(int cipher);
+ static std::string SslCipherSuiteToName(int cipher_suite);
private:
// If true, the server certificate need not match the configured
diff --git a/webrtc/base/sslstreamadapter_unittest.cc b/webrtc/base/sslstreamadapter_unittest.cc
index a3e8d9c..1ed06c3 100644
--- a/webrtc/base/sslstreamadapter_unittest.cc
+++ b/webrtc/base/sslstreamadapter_unittest.cc
@@ -13,6 +13,7 @@
#include <set>
#include <string>
+#include "webrtc/base/bufferqueue.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/scoped_ptr.h"
@@ -21,7 +22,6 @@
#include "webrtc/base/sslidentity.h"
#include "webrtc/base/sslstreamadapter.h"
#include "webrtc/base/stream.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using ::testing::WithParamInterface;
using ::testing::Values;
@@ -29,8 +29,6 @@
using ::testing::tuple;
static const int kBlockSize = 4096;
-static const char kAES_CM_HMAC_SHA1_80[] = "AES_CM_128_HMAC_SHA1_80";
-static const char kAES_CM_HMAC_SHA1_32[] = "AES_CM_128_HMAC_SHA1_32";
static const char kExporterLabel[] = "label";
static const unsigned char kExporterContext[] = "context";
static int kExporterContextLen = sizeof(kExporterContext);
@@ -74,26 +72,26 @@
class SSLStreamAdapterTestBase;
-class SSLDummyStream : public rtc::StreamInterface,
- public sigslot::has_slots<> {
+class SSLDummyStreamBase : public rtc::StreamInterface,
+ public sigslot::has_slots<> {
public:
- explicit SSLDummyStream(SSLStreamAdapterTestBase *test,
- const std::string &side,
- rtc::FifoBuffer *in,
- rtc::FifoBuffer *out) :
- test_(test),
+ SSLDummyStreamBase(SSLStreamAdapterTestBase* test,
+ const std::string &side,
+ rtc::StreamInterface* in,
+ rtc::StreamInterface* out) :
+ test_base_(test),
side_(side),
in_(in),
out_(out),
first_packet_(true) {
- in_->SignalEvent.connect(this, &SSLDummyStream::OnEventIn);
- out_->SignalEvent.connect(this, &SSLDummyStream::OnEventOut);
+ in_->SignalEvent.connect(this, &SSLDummyStreamBase::OnEventIn);
+ out_->SignalEvent.connect(this, &SSLDummyStreamBase::OnEventOut);
}
- virtual rtc::StreamState GetState() const { return rtc::SS_OPEN; }
+ rtc::StreamState GetState() const override { return rtc::SS_OPEN; }
- virtual rtc::StreamResult Read(void* buffer, size_t buffer_len,
- size_t* read, int* error) {
+ rtc::StreamResult Read(void* buffer, size_t buffer_len,
+ size_t* read, int* error) override {
rtc::StreamResult r;
r = in_->Read(buffer, buffer_len, read, error);
@@ -111,22 +109,20 @@
}
// Catch readability events on in and pass them up.
- virtual void OnEventIn(rtc::StreamInterface *stream, int sig,
- int err) {
+ void OnEventIn(rtc::StreamInterface* stream, int sig, int err) {
int mask = (rtc::SE_READ | rtc::SE_CLOSE);
if (sig & mask) {
- LOG(LS_INFO) << "SSLDummyStream::OnEvent side=" << side_ << " sig="
+ LOG(LS_INFO) << "SSLDummyStreamBase::OnEvent side=" << side_ << " sig="
<< sig << " forwarding upward";
PostEvent(sig & mask, 0);
}
}
// Catch writeability events on out and pass them up.
- virtual void OnEventOut(rtc::StreamInterface *stream, int sig,
- int err) {
+ void OnEventOut(rtc::StreamInterface* stream, int sig, int err) {
if (sig & rtc::SE_WRITE) {
- LOG(LS_INFO) << "SSLDummyStream::OnEvent side=" << side_ << " sig="
+ LOG(LS_INFO) << "SSLDummyStreamBase::OnEvent side=" << side_ << " sig="
<< sig << " forwarding upward";
PostEvent(sig & rtc::SE_WRITE, 0);
@@ -135,28 +131,92 @@
// Write to the outgoing FifoBuffer
rtc::StreamResult WriteData(const void* data, size_t data_len,
- size_t* written, int* error) {
+ size_t* written, int* error) {
return out_->Write(data, data_len, written, error);
}
- // Defined later
- virtual rtc::StreamResult Write(const void* data, size_t data_len,
- size_t* written, int* error);
+ rtc::StreamResult Write(const void* data, size_t data_len,
+ size_t* written, int* error) override;
- virtual void Close() {
+ void Close() override {
LOG(LS_INFO) << "Closing outbound stream";
out_->Close();
}
- private:
- SSLStreamAdapterTestBase *test_;
+ protected:
+ SSLStreamAdapterTestBase* test_base_;
const std::string side_;
- rtc::FifoBuffer *in_;
- rtc::FifoBuffer *out_;
+ rtc::StreamInterface* in_;
+ rtc::StreamInterface* out_;
bool first_packet_;
};
+class SSLDummyStreamTLS : public SSLDummyStreamBase {
+ public:
+ SSLDummyStreamTLS(SSLStreamAdapterTestBase* test,
+ const std::string& side,
+ rtc::FifoBuffer* in,
+ rtc::FifoBuffer* out) :
+ SSLDummyStreamBase(test, side, in, out) {
+ }
+};
+
+class BufferQueueStream : public rtc::BufferQueue,
+ public rtc::StreamInterface {
+ public:
+ BufferQueueStream(size_t capacity, size_t default_size)
+ : rtc::BufferQueue(capacity, default_size) {
+ }
+
+ // Implementation of abstract StreamInterface methods.
+
+ // A buffer queue stream is always "open".
+ rtc::StreamState GetState() const override { return rtc::SS_OPEN; }
+
+ // Reading a buffer queue stream will either succeed or block.
+ rtc::StreamResult Read(void* buffer, size_t buffer_len,
+ size_t* read, int* error) override {
+ if (!ReadFront(buffer, buffer_len, read)) {
+ return rtc::SR_BLOCK;
+ }
+ return rtc::SR_SUCCESS;
+ }
+
+ // Writing to a buffer queue stream will either succeed or block.
+ rtc::StreamResult Write(const void* data, size_t data_len,
+ size_t* written, int* error) override {
+ if (!WriteBack(data, data_len, written)) {
+ return rtc::SR_BLOCK;
+ }
+ return rtc::SR_SUCCESS;
+ }
+
+ // A buffer queue stream can not be closed.
+ void Close() override {}
+
+ protected:
+ void NotifyReadableForTest() override {
+ PostEvent(rtc::SE_READ, 0);
+ }
+
+ void NotifyWritableForTest() override {
+ PostEvent(rtc::SE_WRITE, 0);
+ }
+};
+
+class SSLDummyStreamDTLS : public SSLDummyStreamBase {
+ public:
+ SSLDummyStreamDTLS(SSLStreamAdapterTestBase* test,
+ const std::string& side,
+ BufferQueueStream* in,
+ BufferQueueStream* out) :
+ SSLDummyStreamBase(test, side, in, out) {
+ }
+};
+
static const int kFifoBufferSize = 4096;
+static const int kBufferCapacity = 1;
+static const size_t kDefaultBufferSize = 2048;
class SSLStreamAdapterTestBase : public testing::Test,
public sigslot::has_slots<> {
@@ -167,14 +227,12 @@
bool dtls,
rtc::KeyParams client_key_type = rtc::KeyParams(rtc::KT_DEFAULT),
rtc::KeyParams server_key_type = rtc::KeyParams(rtc::KT_DEFAULT))
- : client_buffer_(kFifoBufferSize),
- server_buffer_(kFifoBufferSize),
- client_stream_(
- new SSLDummyStream(this, "c2s", &client_buffer_, &server_buffer_)),
- server_stream_(
- new SSLDummyStream(this, "s2c", &server_buffer_, &client_buffer_)),
- client_ssl_(rtc::SSLStreamAdapter::Create(client_stream_)),
- server_ssl_(rtc::SSLStreamAdapter::Create(server_stream_)),
+ : client_cert_pem_(client_cert_pem),
+ client_private_key_pem_(client_private_key_pem),
+ client_key_type_(client_key_type),
+ server_key_type_(server_key_type),
+ client_stream_(NULL),
+ server_stream_(NULL),
client_identity_(NULL),
server_identity_(NULL),
delay_(0),
@@ -187,21 +245,6 @@
identities_set_(false) {
// Set use of the test RNG to get predictable loss patterns.
rtc::SetRandomTestMode(true);
-
- // Set up the slots
- client_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
- server_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
-
- if (!client_cert_pem.empty() && !client_private_key_pem.empty()) {
- client_identity_ = rtc::SSLIdentity::FromPEMStrings(
- client_private_key_pem, client_cert_pem);
- } else {
- client_identity_ = rtc::SSLIdentity::Generate("client", client_key_type);
- }
- server_identity_ = rtc::SSLIdentity::Generate("server", server_key_type);
-
- client_ssl_->SetIdentity(client_identity_);
- server_ssl_->SetIdentity(server_identity_);
}
~SSLStreamAdapterTestBase() {
@@ -209,14 +252,40 @@
rtc::SetRandomTestMode(false);
}
+ void SetUp() override {
+ CreateStreams();
+
+ client_ssl_.reset(rtc::SSLStreamAdapter::Create(client_stream_));
+ server_ssl_.reset(rtc::SSLStreamAdapter::Create(server_stream_));
+
+ // Set up the slots
+ client_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
+ server_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
+
+ if (!client_cert_pem_.empty() && !client_private_key_pem_.empty()) {
+ client_identity_ = rtc::SSLIdentity::FromPEMStrings(
+ client_private_key_pem_, client_cert_pem_);
+ } else {
+ client_identity_ = rtc::SSLIdentity::Generate("client", client_key_type_);
+ }
+ server_identity_ = rtc::SSLIdentity::Generate("server", server_key_type_);
+
+ client_ssl_->SetIdentity(client_identity_);
+ server_ssl_->SetIdentity(server_identity_);
+ }
+
+ void TearDown() override {
+ client_ssl_.reset(nullptr);
+ server_ssl_.reset(nullptr);
+ }
+
+ virtual void CreateStreams() = 0;
+
// Recreate the client/server identities with the specified validity period.
// |not_before| and |not_after| are offsets from the current time in number
// of seconds.
void ResetIdentitiesWithValidity(int not_before, int not_after) {
- client_stream_ =
- new SSLDummyStream(this, "c2s", &client_buffer_, &server_buffer_);
- server_stream_ =
- new SSLDummyStream(this, "s2c", &server_buffer_, &client_buffer_);
+ CreateStreams();
client_ssl_.reset(rtc::SSLStreamAdapter::Create(client_stream_));
server_ssl_.reset(rtc::SSLStreamAdapter::Create(server_stream_));
@@ -224,18 +293,20 @@
client_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
server_ssl_->SignalEvent.connect(this, &SSLStreamAdapterTestBase::OnEvent);
+ time_t now = time(nullptr);
+
rtc::SSLIdentityParams client_params;
client_params.key_params = rtc::KeyParams(rtc::KT_DEFAULT);
client_params.common_name = "client";
- client_params.not_before = not_before;
- client_params.not_after = not_after;
+ client_params.not_before = now + not_before;
+ client_params.not_after = now + not_after;
client_identity_ = rtc::SSLIdentity::GenerateForTest(client_params);
rtc::SSLIdentityParams server_params;
server_params.key_params = rtc::KeyParams(rtc::KT_DEFAULT);
server_params.common_name = "server";
- server_params.not_before = not_before;
- server_params.not_after = not_after;
+ server_params.not_before = now + not_before;
+ server_params.not_after = now + not_after;
server_identity_ = rtc::SSLIdentity::GenerateForTest(server_params);
client_ssl_->SetIdentity(client_identity_);
@@ -331,9 +402,9 @@
}
}
- rtc::StreamResult DataWritten(SSLDummyStream *from, const void *data,
- size_t data_len, size_t *written,
- int *error) {
+ rtc::StreamResult DataWritten(SSLDummyStreamBase *from, const void *data,
+ size_t data_len, size_t *written,
+ int *error) {
// Randomly drop loss_ percent of packets
if (rtc::CreateRandomId() % 100 < static_cast<uint32_t>(loss_)) {
LOG(LS_INFO) << "Randomly dropping packet, size=" << data_len;
@@ -389,19 +460,18 @@
handshake_wait_ = wait;
}
- void SetDtlsSrtpCiphers(const std::vector<std::string> &ciphers,
- bool client) {
+ void SetDtlsSrtpCryptoSuites(const std::vector<int>& ciphers, bool client) {
if (client)
- client_ssl_->SetDtlsSrtpCiphers(ciphers);
+ client_ssl_->SetDtlsSrtpCryptoSuites(ciphers);
else
- server_ssl_->SetDtlsSrtpCiphers(ciphers);
+ server_ssl_->SetDtlsSrtpCryptoSuites(ciphers);
}
- bool GetDtlsSrtpCipher(bool client, std::string *retval) {
+ bool GetDtlsSrtpCryptoSuite(bool client, int* retval) {
if (client)
- return client_ssl_->GetDtlsSrtpCipher(retval);
+ return client_ssl_->GetDtlsSrtpCryptoSuite(retval);
else
- return server_ssl_->GetDtlsSrtpCipher(retval);
+ return server_ssl_->GetDtlsSrtpCryptoSuite(retval);
}
bool GetPeerCertificate(bool client, rtc::SSLCertificate** cert) {
@@ -443,10 +513,12 @@
virtual void TestTransfer(int size) = 0;
protected:
- rtc::FifoBuffer client_buffer_;
- rtc::FifoBuffer server_buffer_;
- SSLDummyStream *client_stream_; // freed by client_ssl_ destructor
- SSLDummyStream *server_stream_; // freed by server_ssl_ destructor
+ std::string client_cert_pem_;
+ std::string client_private_key_pem_;
+ rtc::KeyParams client_key_type_;
+ rtc::KeyParams server_key_type_;
+ SSLDummyStreamBase *client_stream_; // freed by client_ssl_ destructor
+ SSLDummyStreamBase *server_stream_; // freed by server_ssl_ destructor
rtc::scoped_ptr<rtc::SSLStreamAdapter> client_ssl_;
rtc::scoped_ptr<rtc::SSLStreamAdapter> server_ssl_;
rtc::SSLIdentity *client_identity_; // freed by client_ssl_ destructor
@@ -470,7 +542,17 @@
"",
false,
::testing::get<0>(GetParam()),
- ::testing::get<1>(GetParam())){};
+ ::testing::get<1>(GetParam())),
+ client_buffer_(kFifoBufferSize),
+ server_buffer_(kFifoBufferSize) {
+ }
+
+ void CreateStreams() override {
+ client_stream_ =
+ new SSLDummyStreamTLS(this, "c2s", &client_buffer_, &server_buffer_);
+ server_stream_ =
+ new SSLDummyStreamTLS(this, "s2c", &server_buffer_, &client_buffer_);
+ }
// Test data transfer for TLS
virtual void TestTransfer(int size) {
@@ -549,7 +631,7 @@
if (r == rtc::SR_ERROR || r == rtc::SR_EOS) {
// Unfortunately, errors are the way that the stream adapter
- // signals close in OpenSSL
+ // signals close in OpenSSL.
stream->Close();
return;
}
@@ -565,6 +647,8 @@
}
private:
+ rtc::FifoBuffer client_buffer_;
+ rtc::FifoBuffer server_buffer_;
rtc::MemoryStream send_stream_;
rtc::MemoryStream recv_stream_;
};
@@ -579,6 +663,8 @@
true,
::testing::get<0>(GetParam()),
::testing::get<1>(GetParam())),
+ client_buffer_(kBufferCapacity, kDefaultBufferSize),
+ server_buffer_(kBufferCapacity, kDefaultBufferSize),
packet_size_(1000),
count_(0),
sent_(0) {}
@@ -586,18 +672,32 @@
SSLStreamAdapterTestDTLS(const std::string& cert_pem,
const std::string& private_key_pem) :
SSLStreamAdapterTestBase(cert_pem, private_key_pem, true),
+ client_buffer_(kBufferCapacity, kDefaultBufferSize),
+ server_buffer_(kBufferCapacity, kDefaultBufferSize),
packet_size_(1000), count_(0), sent_(0) {
}
+ void CreateStreams() override {
+ client_stream_ =
+ new SSLDummyStreamDTLS(this, "c2s", &client_buffer_, &server_buffer_);
+ server_stream_ =
+ new SSLDummyStreamDTLS(this, "s2c", &server_buffer_, &client_buffer_);
+ }
+
virtual void WriteData() {
unsigned char *packet = new unsigned char[1600];
- do {
- memset(packet, sent_ & 0xff, packet_size_);
- *(reinterpret_cast<uint32_t *>(packet)) = sent_;
+ while (sent_ < count_) {
+ unsigned int rand_state = sent_;
+ packet[0] = sent_;
+ for (size_t i = 1; i < packet_size_; i++) {
+ // This is a simple LC PRNG. Keep in synch with identical code below.
+ rand_state = (rand_state * 251 + 19937) >> 7;
+ packet[i] = rand_state & 0xff;
+ }
size_t sent;
- int rv = client_ssl_->Write(packet, packet_size_, &sent, 0);
+ rtc::StreamResult rv = client_ssl_->Write(packet, packet_size_, &sent, 0);
if (rv == rtc::SR_SUCCESS) {
LOG(LS_VERBOSE) << "Sent: " << sent_;
sent_++;
@@ -608,7 +708,7 @@
ADD_FAILURE();
break;
}
- } while (sent_ < count_);
+ }
delete [] packet;
}
@@ -637,11 +737,13 @@
// Now parse the datagram
ASSERT_EQ(packet_size_, bread);
- unsigned char* ptr_to_buffer = buffer;
- uint32_t packet_num = *(reinterpret_cast<uint32_t *>(ptr_to_buffer));
+ unsigned char packet_num = buffer[0];
- for (size_t i = 4; i < packet_size_; i++) {
- ASSERT_EQ((packet_num & 0xff), buffer[i]);
+ unsigned int rand_state = packet_num;
+ for (size_t i = 1; i < packet_size_; i++) {
+ // This is a simple LC PRNG. Keep in synch with identical code above.
+ rand_state = (rand_state * 251 + 19937) >> 7;
+ ASSERT_EQ(rand_state & 0xff, buffer[i]);
}
received_.insert(packet_num);
}
@@ -667,6 +769,8 @@
};
private:
+ BufferQueueStream client_buffer_;
+ BufferQueueStream server_buffer_;
size_t packet_size_;
int count_;
int sent_;
@@ -674,23 +778,20 @@
};
-rtc::StreamResult SSLDummyStream::Write(const void* data, size_t data_len,
+rtc::StreamResult SSLDummyStreamBase::Write(const void* data, size_t data_len,
size_t* written, int* error) {
- *written = data_len;
-
LOG(LS_INFO) << "Writing to loopback " << data_len;
if (first_packet_) {
first_packet_ = false;
- if (test_->GetLoseFirstPacket()) {
+ if (test_base_->GetLoseFirstPacket()) {
LOG(LS_INFO) << "Losing initial packet of length " << data_len;
+ *written = data_len; // Fake successful writing also to writer.
return rtc::SR_SUCCESS;
}
}
- return test_->DataWritten(this, data, data_len, written, error);
-
- return rtc::SR_SUCCESS;
+ return test_base_->DataWritten(this, data, data_len, written, error);
};
class SSLStreamAdapterTestDTLSFromPEMStrings : public SSLStreamAdapterTestDTLS {
@@ -782,23 +883,20 @@
};
// Test transfer -- trivial
-// Disabled due to https://code.google.com/p/webrtc/issues/detail?id=5005
-TEST_P(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSTransfer) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransfer) {
MAYBE_SKIP_TEST(HaveDtls);
TestHandshake();
TestTransfer(100);
};
-// Disabled due to https://code.google.com/p/webrtc/issues/detail?id=5005
-TEST_P(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSTransferWithLoss) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithLoss) {
MAYBE_SKIP_TEST(HaveDtls);
TestHandshake();
SetLoss(10);
TestTransfer(100);
};
-// Disabled due to https://code.google.com/p/webrtc/issues/detail?id=5005
-TEST_P(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSTransferWithDamage) {
+TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithDamage) {
MAYBE_SKIP_TEST(HaveDtls);
SetDamage(); // Must be called first because first packet
// write happens at end of handshake.
@@ -809,74 +907,74 @@
// Test DTLS-SRTP with all high ciphers
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHigh) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
- std::vector<std::string> high;
- high.push_back(kAES_CM_HMAC_SHA1_80);
- SetDtlsSrtpCiphers(high, true);
- SetDtlsSrtpCiphers(high, false);
+ std::vector<int> high;
+ high.push_back(rtc::SRTP_AES128_CM_SHA1_80);
+ SetDtlsSrtpCryptoSuites(high, true);
+ SetDtlsSrtpCryptoSuites(high, false);
TestHandshake();
- std::string client_cipher;
- ASSERT_TRUE(GetDtlsSrtpCipher(true, &client_cipher));
- std::string server_cipher;
- ASSERT_TRUE(GetDtlsSrtpCipher(false, &server_cipher));
+ int client_cipher;
+ ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher));
+ int server_cipher;
+ ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher));
ASSERT_EQ(client_cipher, server_cipher);
- ASSERT_EQ(client_cipher, kAES_CM_HMAC_SHA1_80);
+ ASSERT_EQ(client_cipher, rtc::SRTP_AES128_CM_SHA1_80);
};
// Test DTLS-SRTP with all low ciphers
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpLow) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
- std::vector<std::string> low;
- low.push_back(kAES_CM_HMAC_SHA1_32);
- SetDtlsSrtpCiphers(low, true);
- SetDtlsSrtpCiphers(low, false);
+ std::vector<int> low;
+ low.push_back(rtc::SRTP_AES128_CM_SHA1_32);
+ SetDtlsSrtpCryptoSuites(low, true);
+ SetDtlsSrtpCryptoSuites(low, false);
TestHandshake();
- std::string client_cipher;
- ASSERT_TRUE(GetDtlsSrtpCipher(true, &client_cipher));
- std::string server_cipher;
- ASSERT_TRUE(GetDtlsSrtpCipher(false, &server_cipher));
+ int client_cipher;
+ ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher));
+ int server_cipher;
+ ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher));
ASSERT_EQ(client_cipher, server_cipher);
- ASSERT_EQ(client_cipher, kAES_CM_HMAC_SHA1_32);
+ ASSERT_EQ(client_cipher, rtc::SRTP_AES128_CM_SHA1_32);
};
// Test DTLS-SRTP with a mismatch -- should not converge
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHighLow) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
- std::vector<std::string> high;
- high.push_back(kAES_CM_HMAC_SHA1_80);
- std::vector<std::string> low;
- low.push_back(kAES_CM_HMAC_SHA1_32);
- SetDtlsSrtpCiphers(high, true);
- SetDtlsSrtpCiphers(low, false);
+ std::vector<int> high;
+ high.push_back(rtc::SRTP_AES128_CM_SHA1_80);
+ std::vector<int> low;
+ low.push_back(rtc::SRTP_AES128_CM_SHA1_32);
+ SetDtlsSrtpCryptoSuites(high, true);
+ SetDtlsSrtpCryptoSuites(low, false);
TestHandshake();
- std::string client_cipher;
- ASSERT_FALSE(GetDtlsSrtpCipher(true, &client_cipher));
- std::string server_cipher;
- ASSERT_FALSE(GetDtlsSrtpCipher(false, &server_cipher));
+ int client_cipher;
+ ASSERT_FALSE(GetDtlsSrtpCryptoSuite(true, &client_cipher));
+ int server_cipher;
+ ASSERT_FALSE(GetDtlsSrtpCryptoSuite(false, &server_cipher));
};
// Test DTLS-SRTP with each side being mixed -- should select high
TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpMixed) {
MAYBE_SKIP_TEST(HaveDtlsSrtp);
- std::vector<std::string> mixed;
- mixed.push_back(kAES_CM_HMAC_SHA1_80);
- mixed.push_back(kAES_CM_HMAC_SHA1_32);
- SetDtlsSrtpCiphers(mixed, true);
- SetDtlsSrtpCiphers(mixed, false);
+ std::vector<int> mixed;
+ mixed.push_back(rtc::SRTP_AES128_CM_SHA1_80);
+ mixed.push_back(rtc::SRTP_AES128_CM_SHA1_32);
+ SetDtlsSrtpCryptoSuites(mixed, true);
+ SetDtlsSrtpCryptoSuites(mixed, false);
TestHandshake();
- std::string client_cipher;
- ASSERT_TRUE(GetDtlsSrtpCipher(true, &client_cipher));
- std::string server_cipher;
- ASSERT_TRUE(GetDtlsSrtpCipher(false, &server_cipher));
+ int client_cipher;
+ ASSERT_TRUE(GetDtlsSrtpCryptoSuite(true, &client_cipher));
+ int server_cipher;
+ ASSERT_TRUE(GetDtlsSrtpCryptoSuite(false, &server_cipher));
ASSERT_EQ(client_cipher, server_cipher);
- ASSERT_EQ(client_cipher, kAES_CM_HMAC_SHA1_80);
+ ASSERT_EQ(client_cipher, rtc::SRTP_AES128_CM_SHA1_80);
};
// Test an exporter
@@ -921,14 +1019,14 @@
}
// Test data transfer using certs created from strings.
-TEST_P(SSLStreamAdapterTestDTLSFromPEMStrings, TestTransfer) {
+TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestTransfer) {
MAYBE_SKIP_TEST(HaveDtls);
TestHandshake();
TestTransfer(100);
}
// Test getting the remote certificate.
-TEST_P(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) {
+TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) {
MAYBE_SKIP_TEST(HaveDtls);
// Peer certificates haven't been received yet.
@@ -1052,6 +1150,10 @@
Values(rtc::KeyParams::RSA(1024, 65537),
rtc::KeyParams::RSA(1152, 65537),
rtc::KeyParams::ECDSA(rtc::EC_NIST_P256))));
+
+#if !defined(MEMORY_SANITIZER)
+// Fails under MemorySanitizer:
+// See https://code.google.com/p/webrtc/issues/detail?id=5381.
INSTANTIATE_TEST_CASE_P(
SSLStreamAdapterTestsDTLS,
SSLStreamAdapterTestDTLS,
@@ -1061,3 +1163,4 @@
Values(rtc::KeyParams::RSA(1024, 65537),
rtc::KeyParams::RSA(1152, 65537),
rtc::KeyParams::ECDSA(rtc::EC_NIST_P256))));
+#endif
diff --git a/webrtc/base/sslstreamadapterhelper.cc b/webrtc/base/sslstreamadapterhelper.cc
index c3be4ea..61c0e43 100644
--- a/webrtc/base/sslstreamadapterhelper.cc
+++ b/webrtc/base/sslstreamadapterhelper.cc
@@ -29,8 +29,7 @@
role_(SSL_CLIENT),
ssl_error_code_(0), // Not meaningful yet
ssl_mode_(SSL_MODE_TLS),
- ssl_max_version_(SSL_PROTOCOL_TLS_11) {
-}
+ ssl_max_version_(SSL_PROTOCOL_TLS_12) {}
SSLStreamAdapterHelper::~SSLStreamAdapterHelper() = default;
diff --git a/webrtc/base/stream_unittest.cc b/webrtc/base/stream_unittest.cc
index 4172a97..8cfd052 100644
--- a/webrtc/base/stream_unittest.cc
+++ b/webrtc/base/stream_unittest.cc
@@ -12,7 +12,6 @@
#include "webrtc/base/gunit.h"
#include "webrtc/base/pathutils.h"
#include "webrtc/base/stream.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
diff --git a/webrtc/base/stringencode_unittest.cc b/webrtc/base/stringencode_unittest.cc
index 406d9c7..588e9d8 100644
--- a/webrtc/base/stringencode_unittest.cc
+++ b/webrtc/base/stringencode_unittest.cc
@@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/stringencode.h"
@@ -48,7 +49,7 @@
}
char buffer[5];
- memset(buffer, 0x01, ARRAY_SIZE(buffer));
+ memset(buffer, 0x01, arraysize(buffer));
ASSERT_EQ(kTests[i].enclen, utf8_encode(buffer,
kTests[i].encsize,
kTests[i].decoded));
@@ -56,7 +57,7 @@
// Make sure remainder of buffer is unchanged
ASSERT_TRUE(memory_check(buffer + kTests[i].enclen,
0x1,
- ARRAY_SIZE(buffer) - kTests[i].enclen));
+ arraysize(buffer) - kTests[i].enclen));
}
}
diff --git a/webrtc/base/stringutils.cc b/webrtc/base/stringutils.cc
index 868e475..9580253 100644
--- a/webrtc/base/stringutils.cc
+++ b/webrtc/base/stringutils.cc
@@ -77,11 +77,11 @@
} else if (srclen >= buflen) {
srclen = buflen - 1;
}
-#if _DEBUG
+#if !defined(NDEBUG)
// Double check that characters are not UTF-8
for (size_t pos = 0; pos < srclen; ++pos)
RTC_DCHECK_LT(static_cast<unsigned char>(source[pos]), 128);
-#endif // _DEBUG
+#endif
std::copy(source, source + srclen, buffer);
buffer[srclen] = 0;
return srclen;
diff --git a/webrtc/base/systeminfo.cc b/webrtc/base/systeminfo.cc
index bfc96b3..b400aa0 100644
--- a/webrtc/base/systeminfo.cc
+++ b/webrtc/base/systeminfo.cc
@@ -110,7 +110,7 @@
DWORD_PTR process_mask = 0;
DWORD_PTR system_mask = 0;
::GetProcessAffinityMask(::GetCurrentProcess(), &process_mask, &system_mask);
- for (int i = 0; i < sizeof(DWORD_PTR) * 8; ++i) {
+ for (size_t i = 0; i < sizeof(DWORD_PTR) * 8; ++i) {
if (process_mask & 1)
++cur_cpus;
process_mask >>= 1;
diff --git a/webrtc/base/task.cc b/webrtc/base/task.cc
index bdf8f1d..b09ced1 100644
--- a/webrtc/base/task.cc
+++ b/webrtc/base/task.cc
@@ -68,7 +68,7 @@
void Task::Step() {
if (done_) {
-#ifdef _DEBUG
+#if !defined(NDEBUG)
// we do not know how !blocked_ happens when done_ - should be impossible.
// But it causes problems, so in retail build, we force blocked_, and
// under debug we assert.
@@ -88,7 +88,7 @@
// SignalDone();
Stop();
-#ifdef _DEBUG
+#if !defined(NDEBUG)
// verify that stop removed this from its parent
ASSERT(!parent()->IsChildTask(this));
#endif
@@ -125,7 +125,7 @@
// SignalDone();
Stop();
-#if _DEBUG
+#if !defined(NDEBUG)
// verify that stop removed this from its parent
ASSERT(!parent()->IsChildTask(this));
#endif
@@ -150,7 +150,7 @@
// "done_" is set before calling "Stop()" to ensure that this code
// doesn't execute more than once (recursively) for the same task.
Stop();
-#ifdef _DEBUG
+#if !defined(NDEBUG)
// verify that stop removed this from its parent
ASSERT(!parent()->IsChildTask(this));
#endif
diff --git a/webrtc/base/task_unittest.cc b/webrtc/base/task_unittest.cc
index 7f67841..7492436 100644
--- a/webrtc/base/task_unittest.cc
+++ b/webrtc/base/task_unittest.cc
@@ -20,6 +20,7 @@
#include "webrtc/base/win32.h"
#endif // WEBRTC_WIN
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
@@ -27,7 +28,6 @@
#include "webrtc/base/taskrunner.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace rtc {
@@ -408,7 +408,7 @@
class TimeoutChangeTest : public sigslot::has_slots<> {
public:
TimeoutChangeTest()
- : task_count_(ARRAY_SIZE(stuck_tasks_)) {}
+ : task_count_(arraysize(stuck_tasks_)) {}
// no need to delete any tasks; the task runner owns them
~TimeoutChangeTest() {}
@@ -463,7 +463,7 @@
private:
void OnTimeoutId(const int id) {
- for (int i = 0; i < ARRAY_SIZE(stuck_tasks_); ++i) {
+ for (size_t i = 0; i < arraysize(stuck_tasks_); ++i) {
if (stuck_tasks_[i] && stuck_tasks_[i]->unique_id() == id) {
task_count_--;
stuck_tasks_[i] = NULL;
diff --git a/webrtc/base/taskparent.cc b/webrtc/base/taskparent.cc
index db6db37..14d236d 100644
--- a/webrtc/base/taskparent.cc
+++ b/webrtc/base/taskparent.cc
@@ -46,7 +46,7 @@
children_->insert(child);
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool TaskParent::IsChildTask(Task *task) {
ASSERT(task != NULL);
return task->parent_ == this && children_->find(task) != children_->end();
@@ -69,7 +69,7 @@
void TaskParent::AbortAllChildren() {
if (children_->size() > 0) {
-#ifdef _DEBUG
+#if !defined(NDEBUG)
runner_->IncrementAbortCount();
#endif
@@ -78,7 +78,7 @@
(*it)->Abort(true); // Note we do not wake
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
runner_->DecrementAbortCount();
#endif
}
diff --git a/webrtc/base/taskparent.h b/webrtc/base/taskparent.h
index e9342c1..41008fa 100644
--- a/webrtc/base/taskparent.h
+++ b/webrtc/base/taskparent.h
@@ -32,7 +32,7 @@
bool AllChildrenDone();
bool AnyChildError();
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool IsChildTask(Task *task);
#endif
diff --git a/webrtc/base/taskrunner.cc b/webrtc/base/taskrunner.cc
index e7278f1..c50c9f8 100644
--- a/webrtc/base/taskrunner.cc
+++ b/webrtc/base/taskrunner.cc
@@ -23,7 +23,7 @@
: TaskParent(this),
next_timeout_task_(NULL),
tasks_running_(false)
-#ifdef _DEBUG
+#if !defined(NDEBUG)
, abort_count_(0),
deleting_task_(NULL)
#endif
@@ -88,11 +88,11 @@
need_timeout_recalc = true;
}
-#ifdef _DEBUG
+#if !defined(NDEBUG)
deleting_task_ = task;
#endif
delete task;
-#ifdef _DEBUG
+#if !defined(NDEBUG)
deleting_task_ = NULL;
#endif
tasks_[i] = NULL;
diff --git a/webrtc/base/taskrunner.h b/webrtc/base/taskrunner.h
index 9a43aac0..e0cf175 100644
--- a/webrtc/base/taskrunner.h
+++ b/webrtc/base/taskrunner.h
@@ -44,7 +44,7 @@
void UpdateTaskTimeout(Task* task, int64_t previous_task_timeout_time);
-#ifdef _DEBUG
+#if !defined(NDEBUG)
bool is_ok_to_delete(Task* task) {
return task == deleting_task_;
}
@@ -87,7 +87,7 @@
std::vector<Task *> tasks_;
Task *next_timeout_task_;
bool tasks_running_;
-#ifdef _DEBUG
+#if !defined(NDEBUG)
int abort_count_;
Task* deleting_task_;
#endif
diff --git a/webrtc/base/testclient_unittest.cc b/webrtc/base/testclient_unittest.cc
index 17bf4e6..bdd06b3 100644
--- a/webrtc/base/testclient_unittest.cc
+++ b/webrtc/base/testclient_unittest.cc
@@ -14,7 +14,6 @@
#include "webrtc/base/testclient.h"
#include "webrtc/base/testechoserver.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using namespace rtc;
diff --git a/webrtc/base/testutils.h b/webrtc/base/testutils.h
index e56895d..6e7e22a 100644
--- a/webrtc/base/testutils.h
+++ b/webrtc/base/testutils.h
@@ -25,6 +25,7 @@
#include <algorithm>
#include <map>
#include <vector>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/asyncsocket.h"
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
@@ -357,7 +358,7 @@
}
void OnReadEvent(AsyncSocket* socket) {
char data[64 * 1024];
- int result = socket_->Recv(data, ARRAY_SIZE(data));
+ int result = socket_->Recv(data, arraysize(data));
if (result > 0) {
recv_buffer_.insert(recv_buffer_.end(), data, data + result);
}
diff --git a/webrtc/base/thread.cc b/webrtc/base/thread.cc
index 8ab381f..4197d28 100644
--- a/webrtc/base/thread.cc
+++ b/webrtc/base/thread.cc
@@ -140,7 +140,6 @@
Thread::Thread(SocketServer* ss)
: MessageQueue(ss),
- priority_(PRIORITY_NORMAL),
running_(true, false),
#if defined(WEBRTC_WIN)
thread_(NULL),
@@ -188,34 +187,6 @@
return true;
}
-bool Thread::SetPriority(ThreadPriority priority) {
-#if defined(WEBRTC_WIN)
- if (running()) {
- ASSERT(thread_ != NULL);
- BOOL ret = FALSE;
- if (priority == PRIORITY_NORMAL) {
- ret = ::SetThreadPriority(thread_, THREAD_PRIORITY_NORMAL);
- } else if (priority == PRIORITY_HIGH) {
- ret = ::SetThreadPriority(thread_, THREAD_PRIORITY_HIGHEST);
- } else if (priority == PRIORITY_ABOVE_NORMAL) {
- ret = ::SetThreadPriority(thread_, THREAD_PRIORITY_ABOVE_NORMAL);
- } else if (priority == PRIORITY_IDLE) {
- ret = ::SetThreadPriority(thread_, THREAD_PRIORITY_IDLE);
- }
- if (!ret) {
- return false;
- }
- }
- priority_ = priority;
- return true;
-#else
- // TODO: Implement for Linux/Mac if possible.
- if (running()) return false;
- priority_ = priority;
- return true;
-#endif
-}
-
bool Thread::Start(Runnable* runnable) {
ASSERT(owned_);
if (!owned_) return false;
@@ -232,18 +203,10 @@
init->thread = this;
init->runnable = runnable;
#if defined(WEBRTC_WIN)
- DWORD flags = 0;
- if (priority_ != PRIORITY_NORMAL) {
- flags = CREATE_SUSPENDED;
- }
- thread_ = CreateThread(NULL, 0, (LPTHREAD_START_ROUTINE)PreRun, init, flags,
+ thread_ = CreateThread(NULL, 0, (LPTHREAD_START_ROUTINE)PreRun, init, 0,
&thread_id_);
if (thread_) {
running_.Set();
- if (priority_ != PRIORITY_NORMAL) {
- SetPriority(priority_);
- ::ResumeThread(thread_);
- }
} else {
return false;
}
@@ -251,37 +214,6 @@
pthread_attr_t attr;
pthread_attr_init(&attr);
- // Thread priorities are not supported in NaCl.
-#if !defined(__native_client__)
- if (priority_ != PRIORITY_NORMAL) {
- if (priority_ == PRIORITY_IDLE) {
- // There is no POSIX-standard way to set a below-normal priority for an
- // individual thread (only whole process), so let's not support it.
- LOG(LS_WARNING) << "PRIORITY_IDLE not supported";
- } else {
- // Set real-time round-robin policy.
- if (pthread_attr_setschedpolicy(&attr, SCHED_RR) != 0) {
- LOG(LS_ERROR) << "pthread_attr_setschedpolicy";
- }
- struct sched_param param;
- if (pthread_attr_getschedparam(&attr, ¶m) != 0) {
- LOG(LS_ERROR) << "pthread_attr_getschedparam";
- } else {
- // The numbers here are arbitrary.
- if (priority_ == PRIORITY_HIGH) {
- param.sched_priority = 6; // 6 = HIGH
- } else {
- ASSERT(priority_ == PRIORITY_ABOVE_NORMAL);
- param.sched_priority = 4; // 4 = ABOVE_NORMAL
- }
- if (pthread_attr_setschedparam(&attr, ¶m) != 0) {
- LOG(LS_ERROR) << "pthread_attr_setschedparam";
- }
- }
- }
- }
-#endif // !defined(__native_client__)
-
int error_code = pthread_create(&thread_, &attr, PreRun, init);
if (0 != error_code) {
LOG(LS_ERROR) << "Unable to create pthread, error " << error_code;
@@ -345,7 +277,7 @@
// static
void Thread::AssertBlockingIsAllowedOnCurrentThread() {
-#ifdef _DEBUG
+#if !defined(NDEBUG)
Thread* current = Thread::Current();
ASSERT(!current || current->blocking_calls_allowed_);
#endif
diff --git a/webrtc/base/thread.h b/webrtc/base/thread.h
index 9cbe8ec..f91aa56 100644
--- a/webrtc/base/thread.h
+++ b/webrtc/base/thread.h
@@ -78,13 +78,6 @@
bool *ready;
};
-enum ThreadPriority {
- PRIORITY_IDLE = -1,
- PRIORITY_NORMAL = 0,
- PRIORITY_ABOVE_NORMAL = 1,
- PRIORITY_HIGH = 2,
-};
-
class Runnable {
public:
virtual ~Runnable() {}
@@ -137,10 +130,6 @@
const std::string& name() const { return name_; }
bool SetName(const std::string& name, const void* obj);
- // Sets the thread's priority. Must be called before Start().
- ThreadPriority priority() const { return priority_; }
- bool SetPriority(ThreadPriority priority);
-
// Starts the execution of the thread.
bool Start(Runnable* runnable = NULL);
@@ -271,7 +260,6 @@
std::list<_SendMessage> sendlist_;
std::string name_;
- ThreadPriority priority_;
Event running_; // Signalled means running.
#if defined(WEBRTC_POSIX)
diff --git a/webrtc/base/thread_checker_impl.cc b/webrtc/base/thread_checker_impl.cc
index ea88308..79be606 100644
--- a/webrtc/base/thread_checker_impl.cc
+++ b/webrtc/base/thread_checker_impl.cc
@@ -12,6 +12,8 @@
#include "webrtc/base/thread_checker_impl.h"
+#include "webrtc/base/platform_thread.h"
+
namespace rtc {
ThreadCheckerImpl::ThreadCheckerImpl() : valid_thread_(CurrentThreadRef()) {
diff --git a/webrtc/base/thread_checker_impl.h b/webrtc/base/thread_checker_impl.h
index 7b39ada..0455835 100644
--- a/webrtc/base/thread_checker_impl.h
+++ b/webrtc/base/thread_checker_impl.h
@@ -14,7 +14,7 @@
#define WEBRTC_BASE_THREAD_CHECKER_IMPL_H_
#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/platform_thread_types.h"
namespace rtc {
diff --git a/webrtc/base/thread_checker_unittest.cc b/webrtc/base/thread_checker_unittest.cc
index bcffb52..3381900 100644
--- a/webrtc/base/thread_checker_unittest.cc
+++ b/webrtc/base/thread_checker_unittest.cc
@@ -15,7 +15,6 @@
#include "webrtc/base/thread.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
// Duplicated from base/threading/thread_checker.h so that we can be
// good citizens there and undef the macro.
diff --git a/webrtc/base/thread_unittest.cc b/webrtc/base/thread_unittest.cc
index e50e45c..7ed4326 100644
--- a/webrtc/base/thread_unittest.cc
+++ b/webrtc/base/thread_unittest.cc
@@ -15,7 +15,6 @@
#include "webrtc/base/physicalsocketserver.h"
#include "webrtc/base/socketaddress.h"
#include "webrtc/base/thread.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
#if defined(WEBRTC_WIN)
#include <comdef.h> // NOLINT
@@ -137,16 +136,48 @@
Event* event_;
};
+// A bool wrapped in a mutex, to avoid data races. Using a volatile
+// bool should be sufficient for correct code ("eventual consistency"
+// between caches is sufficient), but we can't tell the compiler about
+// that, and then tsan complains about a data race.
+
+// See also discussion at
+// http://stackoverflow.com/questions/7223164/is-mutex-needed-to-synchronize-a-simple-flag-between-pthreads
+
+// Using std::atomic<bool> or std::atomic_flag in C++11 is probably
+// the right thing to do, but those features are not yet allowed. Or
+// rtc::AtomicInt, if/when that is added. Since the use isn't
+// performance critical, use a plain critical section for the time
+// being.
+
+class AtomicBool {
+ public:
+ explicit AtomicBool(bool value = false) : flag_(value) {}
+ AtomicBool& operator=(bool value) {
+ CritScope scoped_lock(&cs_);
+ flag_ = value;
+ return *this;
+ }
+ bool get() const {
+ CritScope scoped_lock(&cs_);
+ return flag_;
+ }
+
+ private:
+ mutable CriticalSection cs_;
+ bool flag_;
+};
+
// Function objects to test Thread::Invoke.
struct FunctorA {
int operator()() { return 42; }
};
class FunctorB {
public:
- explicit FunctorB(bool* flag) : flag_(flag) {}
+ explicit FunctorB(AtomicBool* flag) : flag_(flag) {}
void operator()() { if (flag_) *flag_ = true; }
private:
- bool* flag_;
+ AtomicBool* flag_;
};
struct FunctorC {
int operator()() {
@@ -220,33 +251,6 @@
delete thread;
}
-// Test that setting thread priorities doesn't cause a malfunction.
-// There's no easy way to verify the priority was set properly at this time.
-TEST(ThreadTest, Priorities) {
- Thread *thread;
- thread = new Thread();
- EXPECT_TRUE(thread->SetPriority(PRIORITY_HIGH));
- EXPECT_TRUE(thread->Start());
- thread->Stop();
- delete thread;
- thread = new Thread();
- EXPECT_TRUE(thread->SetPriority(PRIORITY_ABOVE_NORMAL));
- EXPECT_TRUE(thread->Start());
- thread->Stop();
- delete thread;
-
- thread = new Thread();
- EXPECT_TRUE(thread->Start());
-#if defined(WEBRTC_WIN)
- EXPECT_TRUE(thread->SetPriority(PRIORITY_ABOVE_NORMAL));
-#else
- EXPECT_FALSE(thread->SetPriority(PRIORITY_ABOVE_NORMAL));
-#endif
- thread->Stop();
- delete thread;
-
-}
-
TEST(ThreadTest, Wrap) {
Thread* current_thread = Thread::Current();
current_thread->UnwrapCurrent();
@@ -266,10 +270,10 @@
thread.Start();
// Try calling functors.
EXPECT_EQ(42, thread.Invoke<int>(FunctorA()));
- bool called = false;
+ AtomicBool called;
FunctorB f2(&called);
thread.Invoke<void>(f2);
- EXPECT_TRUE(called);
+ EXPECT_TRUE(called.get());
// Try calling bare functions.
struct LocalFuncs {
static int Func1() { return 999; }
@@ -408,9 +412,9 @@
Thread thread;
thread.Start();
// Try calling functor.
- bool called = false;
+ AtomicBool called;
invoker.AsyncInvoke<void>(&thread, FunctorB(&called));
- EXPECT_TRUE_WAIT(called, kWaitTimeout);
+ EXPECT_TRUE_WAIT(called.get(), kWaitTimeout);
}
TEST_F(AsyncInvokeTest, WithCallback) {
@@ -478,26 +482,26 @@
TEST_F(AsyncInvokeTest, Flush) {
AsyncInvoker invoker;
- bool flag1 = false;
- bool flag2 = false;
+ AtomicBool flag1;
+ AtomicBool flag2;
// Queue two async calls to the current thread.
invoker.AsyncInvoke<void>(Thread::Current(),
FunctorB(&flag1));
invoker.AsyncInvoke<void>(Thread::Current(),
FunctorB(&flag2));
// Because we haven't pumped messages, these should not have run yet.
- EXPECT_FALSE(flag1);
- EXPECT_FALSE(flag2);
+ EXPECT_FALSE(flag1.get());
+ EXPECT_FALSE(flag2.get());
// Force them to run now.
invoker.Flush(Thread::Current());
- EXPECT_TRUE(flag1);
- EXPECT_TRUE(flag2);
+ EXPECT_TRUE(flag1.get());
+ EXPECT_TRUE(flag2.get());
}
TEST_F(AsyncInvokeTest, FlushWithIds) {
AsyncInvoker invoker;
- bool flag1 = false;
- bool flag2 = false;
+ AtomicBool flag1;
+ AtomicBool flag2;
// Queue two async calls to the current thread, one with a message id.
invoker.AsyncInvoke<void>(Thread::Current(),
FunctorB(&flag1),
@@ -505,17 +509,17 @@
invoker.AsyncInvoke<void>(Thread::Current(),
FunctorB(&flag2));
// Because we haven't pumped messages, these should not have run yet.
- EXPECT_FALSE(flag1);
- EXPECT_FALSE(flag2);
+ EXPECT_FALSE(flag1.get());
+ EXPECT_FALSE(flag2.get());
// Execute pending calls with id == 5.
invoker.Flush(Thread::Current(), 5);
- EXPECT_TRUE(flag1);
- EXPECT_FALSE(flag2);
+ EXPECT_TRUE(flag1.get());
+ EXPECT_FALSE(flag2.get());
flag1 = false;
// Execute all pending calls. The id == 5 call should not execute again.
invoker.Flush(Thread::Current());
- EXPECT_FALSE(flag1);
- EXPECT_TRUE(flag2);
+ EXPECT_FALSE(flag1.get());
+ EXPECT_TRUE(flag2.get());
}
class GuardedAsyncInvokeTest : public testing::Test {
@@ -564,11 +568,11 @@
// Kill |thread|.
thread = nullptr;
// Try calling functor.
- bool called = false;
+ AtomicBool called;
EXPECT_FALSE(invoker->AsyncInvoke<void>(FunctorB(&called)));
// With thread gone, nothing should happen.
- WAIT(called, kWaitTimeout);
- EXPECT_FALSE(called);
+ WAIT(called.get(), kWaitTimeout);
+ EXPECT_FALSE(called.get());
}
// Test that we can call AsyncInvoke with callback after the thread died.
@@ -595,9 +599,9 @@
TEST_F(GuardedAsyncInvokeTest, FireAndForget) {
GuardedAsyncInvoker invoker;
// Try calling functor.
- bool called = false;
+ AtomicBool called;
EXPECT_TRUE(invoker.AsyncInvoke<void>(FunctorB(&called)));
- EXPECT_TRUE_WAIT(called, kWaitTimeout);
+ EXPECT_TRUE_WAIT(called.get(), kWaitTimeout);
}
TEST_F(GuardedAsyncInvokeTest, WithCallback) {
@@ -660,39 +664,39 @@
TEST_F(GuardedAsyncInvokeTest, Flush) {
GuardedAsyncInvoker invoker;
- bool flag1 = false;
- bool flag2 = false;
+ AtomicBool flag1;
+ AtomicBool flag2;
// Queue two async calls to the current thread.
EXPECT_TRUE(invoker.AsyncInvoke<void>(FunctorB(&flag1)));
EXPECT_TRUE(invoker.AsyncInvoke<void>(FunctorB(&flag2)));
// Because we haven't pumped messages, these should not have run yet.
- EXPECT_FALSE(flag1);
- EXPECT_FALSE(flag2);
+ EXPECT_FALSE(flag1.get());
+ EXPECT_FALSE(flag2.get());
// Force them to run now.
EXPECT_TRUE(invoker.Flush());
- EXPECT_TRUE(flag1);
- EXPECT_TRUE(flag2);
+ EXPECT_TRUE(flag1.get());
+ EXPECT_TRUE(flag2.get());
}
TEST_F(GuardedAsyncInvokeTest, FlushWithIds) {
GuardedAsyncInvoker invoker;
- bool flag1 = false;
- bool flag2 = false;
+ AtomicBool flag1;
+ AtomicBool flag2;
// Queue two async calls to the current thread, one with a message id.
EXPECT_TRUE(invoker.AsyncInvoke<void>(FunctorB(&flag1), 5));
EXPECT_TRUE(invoker.AsyncInvoke<void>(FunctorB(&flag2)));
// Because we haven't pumped messages, these should not have run yet.
- EXPECT_FALSE(flag1);
- EXPECT_FALSE(flag2);
+ EXPECT_FALSE(flag1.get());
+ EXPECT_FALSE(flag2.get());
// Execute pending calls with id == 5.
EXPECT_TRUE(invoker.Flush(5));
- EXPECT_TRUE(flag1);
- EXPECT_FALSE(flag2);
+ EXPECT_TRUE(flag1.get());
+ EXPECT_FALSE(flag2.get());
flag1 = false;
// Execute all pending calls. The id == 5 call should not execute again.
EXPECT_TRUE(invoker.Flush());
- EXPECT_FALSE(flag1);
- EXPECT_TRUE(flag2);
+ EXPECT_FALSE(flag1.get());
+ EXPECT_TRUE(flag2.get());
}
#if defined(WEBRTC_WIN)
diff --git a/webrtc/base/timeutils.cc b/webrtc/base/timeutils.cc
index fac5b66..05e9ad8 100644
--- a/webrtc/base/timeutils.cc
+++ b/webrtc/base/timeutils.cc
@@ -204,4 +204,48 @@
return unwrapped_ts;
}
+int64_t TmToSeconds(const std::tm& tm) {
+ static short int mdays[12] = {31, 28, 31, 30, 31, 30, 31, 31, 30, 31, 30, 31};
+ static short int cumul_mdays[12] = {0, 31, 59, 90, 120, 151,
+ 181, 212, 243, 273, 304, 334};
+ int year = tm.tm_year + 1900;
+ int month = tm.tm_mon;
+ int day = tm.tm_mday - 1; // Make 0-based like the rest.
+ int hour = tm.tm_hour;
+ int min = tm.tm_min;
+ int sec = tm.tm_sec;
+
+ bool expiry_in_leap_year = (year % 4 == 0 &&
+ (year % 100 != 0 || year % 400 == 0));
+
+ if (year < 1970)
+ return -1;
+ if (month < 0 || month > 11)
+ return -1;
+ if (day < 0 || day >= mdays[month] + (expiry_in_leap_year && month == 2 - 1))
+ return -1;
+ if (hour < 0 || hour > 23)
+ return -1;
+ if (min < 0 || min > 59)
+ return -1;
+ if (sec < 0 || sec > 59)
+ return -1;
+
+ day += cumul_mdays[month];
+
+ // Add number of leap days between 1970 and the expiration year, inclusive.
+ day += ((year / 4 - 1970 / 4) - (year / 100 - 1970 / 100) +
+ (year / 400 - 1970 / 400));
+
+ // We will have added one day too much above if expiration is during a leap
+ // year, and expiration is in January or February.
+ if (expiry_in_leap_year && month <= 2 - 1) // |month| is zero based.
+ day -= 1;
+
+ // Combine all variables into seconds from 1970-01-01 00:00 (except |month|
+ // which was accumulated into |day| above).
+ return (((static_cast<int64_t>
+ (year - 1970) * 365 + day) * 24 + hour) * 60 + min) * 60 + sec;
+}
+
} // namespace rtc
diff --git a/webrtc/base/timeutils.h b/webrtc/base/timeutils.h
index bdeccc3..3ade430 100644
--- a/webrtc/base/timeutils.h
+++ b/webrtc/base/timeutils.h
@@ -11,6 +11,7 @@
#ifndef WEBRTC_BASE_TIMEUTILS_H_
#define WEBRTC_BASE_TIMEUTILS_H_
+#include <ctime>
#include <time.h>
#include "webrtc/base/basictypes.h"
@@ -93,6 +94,11 @@
int64_t num_wrap_;
};
+// Convert from std::tm, which is relative to 1900-01-01 00:00 to number of
+// seconds from 1970-01-01 00:00 ("epoch"). Don't return time_t since that
+// is still 32 bits on many systems.
+int64_t TmToSeconds(const std::tm& tm);
+
} // namespace rtc
#endif // WEBRTC_BASE_TIMEUTILS_H_
diff --git a/webrtc/base/timeutils_unittest.cc b/webrtc/base/timeutils_unittest.cc
index d1b9ad4..688658b 100644
--- a/webrtc/base/timeutils_unittest.cc
+++ b/webrtc/base/timeutils_unittest.cc
@@ -10,6 +10,7 @@
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
+#include "webrtc/base/helpers.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
@@ -166,4 +167,99 @@
EXPECT_EQ(unwrapped_ts, wraparound_handler_.Unwrap(ts));
}
+class TmToSeconds : public testing::Test {
+ public:
+ TmToSeconds() {
+ // Set use of the test RNG to get deterministic expiration timestamp.
+ rtc::SetRandomTestMode(true);
+ }
+ ~TmToSeconds() {
+ // Put it back for the next test.
+ rtc::SetRandomTestMode(false);
+ }
+
+ void TestTmToSeconds(int times) {
+ static char mdays[12] = {31, 28, 31, 30, 31, 30, 31, 31, 30, 31, 30, 31};
+ for (int i = 0; i < times; i++) {
+
+ // First generate something correct and check that TmToSeconds is happy.
+ int year = rtc::CreateRandomId() % 400 + 1970;
+
+ bool leap_year = false;
+ if (year % 4 == 0)
+ leap_year = true;
+ if (year % 100 == 0)
+ leap_year = false;
+ if (year % 400 == 0)
+ leap_year = true;
+
+ std::tm tm;
+ tm.tm_year = year - 1900; // std::tm is year 1900 based.
+ tm.tm_mon = rtc::CreateRandomId() % 12;
+ tm.tm_mday = rtc::CreateRandomId() % mdays[tm.tm_mon] + 1;
+ tm.tm_hour = rtc::CreateRandomId() % 24;
+ tm.tm_min = rtc::CreateRandomId() % 60;
+ tm.tm_sec = rtc::CreateRandomId() % 60;
+ int64_t t = rtc::TmToSeconds(tm);
+ EXPECT_TRUE(t >= 0);
+
+ // Now damage a random field and check that TmToSeconds is unhappy.
+ switch (rtc::CreateRandomId() % 11) {
+ case 0:
+ tm.tm_year = 1969 - 1900;
+ break;
+ case 1:
+ tm.tm_mon = -1;
+ break;
+ case 2:
+ tm.tm_mon = 12;
+ break;
+ case 3:
+ tm.tm_mday = 0;
+ break;
+ case 4:
+ tm.tm_mday = mdays[tm.tm_mon] + (leap_year && tm.tm_mon == 1) + 1;
+ break;
+ case 5:
+ tm.tm_hour = -1;
+ break;
+ case 6:
+ tm.tm_hour = 24;
+ break;
+ case 7:
+ tm.tm_min = -1;
+ break;
+ case 8:
+ tm.tm_min = 60;
+ break;
+ case 9:
+ tm.tm_sec = -1;
+ break;
+ case 10:
+ tm.tm_sec = 60;
+ break;
+ }
+ EXPECT_EQ(rtc::TmToSeconds(tm), -1);
+ }
+ // Check consistency with the system gmtime_r. With time_t, we can only
+ // portably test dates until 2038, which is achieved by the % 0x80000000.
+ for (int i = 0; i < times; i++) {
+ time_t t = rtc::CreateRandomId() % 0x80000000;
+#if defined(WEBRTC_WIN)
+ std::tm* tm = std::gmtime(&t);
+ EXPECT_TRUE(tm);
+ EXPECT_TRUE(rtc::TmToSeconds(*tm) == t);
+#else
+ std::tm tm;
+ EXPECT_TRUE(gmtime_r(&t, &tm));
+ EXPECT_TRUE(rtc::TmToSeconds(tm) == t);
+#endif
+ }
+ }
+};
+
+TEST_F(TmToSeconds, TestTmToSeconds) {
+ TestTmToSeconds(100000);
+}
+
} // namespace rtc
diff --git a/webrtc/base/trace_event.h b/webrtc/base/trace_event.h
index c14cbff..3916af4 100644
--- a/webrtc/base/trace_event.h
+++ b/webrtc/base/trace_event.h
@@ -701,7 +701,7 @@
explicit TraceID(const void* id, unsigned char* flags)
: data_(static_cast<unsigned long long>(
- reinterpret_cast<unsigned long>(id))) {
+ reinterpret_cast<uintptr_t>(id))) {
*flags |= TRACE_EVENT_FLAG_MANGLE_ID;
}
explicit TraceID(ForceMangle id, unsigned char* flags) : data_(id.data()) {
diff --git a/webrtc/base/unittest_main.cc b/webrtc/base/unittest_main.cc
index f952b2d..167570d 100644
--- a/webrtc/base/unittest_main.cc
+++ b/webrtc/base/unittest_main.cc
@@ -78,12 +78,12 @@
_CrtSetReportHook2(_CRT_RPTHOOK_INSTALL, TestCrtReportHandler);
}
-#ifdef _DEBUG // Turn on memory leak checking on Windows.
+#if !defined(NDEBUG) // Turn on memory leak checking on Windows.
_CrtSetDbgFlag(_CRTDBG_ALLOC_MEM_DF |_CRTDBG_LEAK_CHECK_DF);
if (FLAG_crt_break_alloc >= 0) {
_crtBreakAlloc = FLAG_crt_break_alloc;
}
-#endif // _DEBUG
+#endif
#endif // WEBRTC_WIN
rtc::Filesystem::SetOrganizationName("google");
@@ -93,6 +93,10 @@
rtc::LogMessage::LogTimestamps();
if (*FLAG_log != '\0') {
rtc::LogMessage::ConfigureLogging(FLAG_log);
+ } else if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO) {
+ // Default to LS_INFO, even for release builds to provide better test
+ // logging.
+ rtc::LogMessage::LogToDebug(rtc::LS_INFO);
}
// Initialize SSL which are used by several tests.
diff --git a/webrtc/base/unixfilesystem.cc b/webrtc/base/unixfilesystem.cc
index 30d6e78..b474324 100644
--- a/webrtc/base/unixfilesystem.cc
+++ b/webrtc/base/unixfilesystem.cc
@@ -44,6 +44,7 @@
#include <sys/syslimits.h>
#endif
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/pathutils.h"
#include "webrtc/base/stream.h"
@@ -176,7 +177,7 @@
kCreateFolder, &fr))
return false;
unsigned char buffer[NAME_MAX+1];
- if (0 != FSRefMakePath(&fr, buffer, ARRAY_SIZE(buffer)))
+ if (0 != FSRefMakePath(&fr, buffer, arraysize(buffer)))
return false;
pathname.SetPathname(reinterpret_cast<char*>(buffer), "");
#elif defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
@@ -303,7 +304,7 @@
#endif // WEBRTC_MAC && !defined(WEBRTC_IOS)
#endif // WEBRTC_ANDROID || WEBRTC_IOS
};
- for (size_t i = 0; i < ARRAY_SIZE(kTempPrefixes); ++i) {
+ for (size_t i = 0; i < arraysize(kTempPrefixes); ++i) {
if (0 == strncmp(pathname.pathname().c_str(), kTempPrefixes[i],
strlen(kTempPrefixes[i])))
return true;
@@ -372,12 +373,12 @@
return success;
#elif defined(__native_client__)
return false;
-#elif IOS
+#elif WEBRTC_IOS
IOSAppName(path);
return true;
#else // WEBRTC_MAC && !defined(WEBRTC_IOS)
char buffer[PATH_MAX + 2];
- ssize_t len = readlink("/proc/self/exe", buffer, ARRAY_SIZE(buffer) - 1);
+ ssize_t len = readlink("/proc/self/exe", buffer, arraysize(buffer) - 1);
if ((len <= 0) || (len == PATH_MAX + 1))
return false;
buffer[len] = '\0';
@@ -399,7 +400,7 @@
kCreateFolder, &fr))
return false;
unsigned char buffer[NAME_MAX+1];
- if (0 != FSRefMakePath(&fr, buffer, ARRAY_SIZE(buffer)))
+ if (0 != FSRefMakePath(&fr, buffer, arraysize(buffer)))
return false;
path->SetPathname(reinterpret_cast<char*>(buffer), "");
} else {
@@ -487,7 +488,7 @@
// Create a random directory as /tmp/<appname>-<pid>-<timestamp>
char buffer[128];
- sprintfn(buffer, ARRAY_SIZE(buffer), "-%d-%d",
+ sprintfn(buffer, arraysize(buffer), "-%d-%d",
static_cast<int>(getpid()),
static_cast<int>(time(0)));
std::string folder(application_name_);
diff --git a/webrtc/base/urlencode_unittest.cc b/webrtc/base/urlencode_unittest.cc
index 5216913..6a61db3 100644
--- a/webrtc/base/urlencode_unittest.cc
+++ b/webrtc/base/urlencode_unittest.cc
@@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/common.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/thread.h"
@@ -19,7 +20,7 @@
char source[] = "^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^"
"^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^";
char dest[1];
- ASSERT_EQ(0, UrlEncode(source, dest, ARRAY_SIZE(dest)));
+ ASSERT_EQ(0, UrlEncode(source, dest, arraysize(dest)));
ASSERT_EQ('\0', dest[0]);
dest[0] = 'a';
@@ -30,7 +31,7 @@
TEST(Urlencode, OneCharacterConversion) {
char source[] = "^";
char dest[4];
- ASSERT_EQ(3, UrlEncode(source, dest, ARRAY_SIZE(dest)));
+ ASSERT_EQ(3, UrlEncode(source, dest, arraysize(dest)));
ASSERT_STREQ("%5E", dest);
}
@@ -40,7 +41,7 @@
// hold the text given.
char source[] = "aa";
char dest[3];
- ASSERT_EQ(2, UrlEncode(source, dest, ARRAY_SIZE(dest)));
+ ASSERT_EQ(2, UrlEncode(source, dest, arraysize(dest)));
ASSERT_STREQ("aa", dest);
}
@@ -49,14 +50,14 @@
// big enough to hold the encoding.
char source[] = "&";
char dest[3];
- ASSERT_EQ(0, UrlEncode(source, dest, ARRAY_SIZE(dest)));
+ ASSERT_EQ(0, UrlEncode(source, dest, arraysize(dest)));
ASSERT_EQ('\0', dest[0]);
}
TEST(Urlencode, Encoding1) {
char source[] = "A^ ";
char dest[8];
- ASSERT_EQ(5, UrlEncode(source, dest, ARRAY_SIZE(dest)));
+ ASSERT_EQ(5, UrlEncode(source, dest, arraysize(dest)));
ASSERT_STREQ("A%5E+", dest);
}
@@ -64,7 +65,7 @@
char source[] = "A^ ";
char dest[8];
ASSERT_EQ(7, rtc::UrlEncodeWithoutEncodingSpaceAsPlus(source, dest,
- ARRAY_SIZE(dest)));
+ arraysize(dest)));
ASSERT_STREQ("A%5E%20", dest);
}
diff --git a/webrtc/base/virtualsocket_unittest.cc b/webrtc/base/virtualsocket_unittest.cc
index 694b154..2cd2b5e 100644
--- a/webrtc/base/virtualsocket_unittest.cc
+++ b/webrtc/base/virtualsocket_unittest.cc
@@ -14,6 +14,7 @@
#include <netinet/in.h>
#endif
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/testclient.h"
@@ -21,7 +22,6 @@
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/base/virtualsocketserver.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
using namespace rtc;
@@ -1022,9 +1022,9 @@
const double kTestDev[] = { 0.25, 0.1, 0.01 };
// TODO: The current code only works for 1000 data points or more.
const uint32_t kTestSamples[] = {/*10, 100,*/ 1000};
- for (size_t midx = 0; midx < ARRAY_SIZE(kTestMean); ++midx) {
- for (size_t didx = 0; didx < ARRAY_SIZE(kTestDev); ++didx) {
- for (size_t sidx = 0; sidx < ARRAY_SIZE(kTestSamples); ++sidx) {
+ for (size_t midx = 0; midx < arraysize(kTestMean); ++midx) {
+ for (size_t didx = 0; didx < arraysize(kTestDev); ++didx) {
+ for (size_t sidx = 0; sidx < arraysize(kTestSamples); ++sidx) {
ASSERT_LT(0u, kTestSamples[sidx]);
const uint32_t kStdDev =
static_cast<uint32_t>(kTestDev[didx] * kTestMean[midx]);
diff --git a/webrtc/base/win32.cc b/webrtc/base/win32.cc
index 6e09829..182b84f 100644
--- a/webrtc/base/win32.cc
+++ b/webrtc/base/win32.cc
@@ -14,6 +14,7 @@
#include <ws2tcpip.h>
#include <algorithm>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/common.h"
@@ -87,7 +88,7 @@
int current = 1;
int max = 0;
int maxpos = -1;
- int run_array_size = ARRAY_SIZE(runpos);
+ int run_array_size = arraysize(runpos);
// Run over the address marking runs of 0s.
for (int i = 0; i < run_array_size; ++i) {
if (as_shorts[i] == 0) {
diff --git a/webrtc/base/win32filesystem.cc b/webrtc/base/win32filesystem.cc
index 8ac918f..b731974 100644
--- a/webrtc/base/win32filesystem.cc
+++ b/webrtc/base/win32filesystem.cc
@@ -15,6 +15,7 @@
#include <shlobj.h>
#include <tchar.h>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/pathutils.h"
#include "webrtc/base/scoped_ptr.h"
@@ -197,16 +198,16 @@
bool Win32Filesystem::GetTemporaryFolder(Pathname &pathname, bool create,
const std::string *append) {
wchar_t buffer[MAX_PATH + 1];
- if (!::GetTempPath(ARRAY_SIZE(buffer), buffer))
+ if (!::GetTempPath(arraysize(buffer), buffer))
return false;
if (!IsCurrentProcessLowIntegrity() &&
- !::GetLongPathName(buffer, buffer, ARRAY_SIZE(buffer)))
+ !::GetLongPathName(buffer, buffer, arraysize(buffer)))
return false;
size_t len = strlen(buffer);
if ((len > 0) && (buffer[len-1] != '\\')) {
- len += strcpyn(buffer + len, ARRAY_SIZE(buffer) - len, L"\\");
+ len += strcpyn(buffer + len, arraysize(buffer) - len, L"\\");
}
- if (len >= ARRAY_SIZE(buffer) - 1)
+ if (len >= arraysize(buffer) - 1)
return false;
pathname.clear();
pathname.SetFolder(ToUtf8(buffer));
@@ -295,10 +296,10 @@
bool Win32Filesystem::IsTemporaryPath(const Pathname& pathname) {
TCHAR buffer[MAX_PATH + 1];
- if (!::GetTempPath(ARRAY_SIZE(buffer), buffer))
+ if (!::GetTempPath(arraysize(buffer), buffer))
return false;
if (!IsCurrentProcessLowIntegrity() &&
- !::GetLongPathName(buffer, buffer, ARRAY_SIZE(buffer)))
+ !::GetLongPathName(buffer, buffer, arraysize(buffer)))
return false;
return (::strnicmp(ToUtf16(pathname.pathname()).c_str(),
buffer, strlen(buffer)) == 0);
@@ -337,7 +338,7 @@
bool Win32Filesystem::GetAppPathname(Pathname* path) {
TCHAR buffer[MAX_PATH + 1];
- if (0 == ::GetModuleFileName(NULL, buffer, ARRAY_SIZE(buffer)))
+ if (0 == ::GetModuleFileName(NULL, buffer, arraysize(buffer)))
return false;
path->SetPathname(ToUtf8(buffer));
return true;
@@ -351,20 +352,20 @@
if (!::SHGetSpecialFolderPath(NULL, buffer, csidl, TRUE))
return false;
if (!IsCurrentProcessLowIntegrity() &&
- !::GetLongPathName(buffer, buffer, ARRAY_SIZE(buffer)))
+ !::GetLongPathName(buffer, buffer, arraysize(buffer)))
return false;
- size_t len = strcatn(buffer, ARRAY_SIZE(buffer), __T("\\"));
- len += strcpyn(buffer + len, ARRAY_SIZE(buffer) - len,
+ size_t len = strcatn(buffer, arraysize(buffer), __T("\\"));
+ len += strcpyn(buffer + len, arraysize(buffer) - len,
ToUtf16(organization_name_).c_str());
if ((len > 0) && (buffer[len-1] != __T('\\'))) {
- len += strcpyn(buffer + len, ARRAY_SIZE(buffer) - len, __T("\\"));
+ len += strcpyn(buffer + len, arraysize(buffer) - len, __T("\\"));
}
- len += strcpyn(buffer + len, ARRAY_SIZE(buffer) - len,
+ len += strcpyn(buffer + len, arraysize(buffer) - len,
ToUtf16(application_name_).c_str());
if ((len > 0) && (buffer[len-1] != __T('\\'))) {
- len += strcpyn(buffer + len, ARRAY_SIZE(buffer) - len, __T("\\"));
+ len += strcpyn(buffer + len, arraysize(buffer) - len, __T("\\"));
}
- if (len >= ARRAY_SIZE(buffer) - 1)
+ if (len >= arraysize(buffer) - 1)
return false;
path->clear();
path->SetFolder(ToUtf8(buffer));
diff --git a/webrtc/base/win32regkey_unittest.cc b/webrtc/base/win32regkey_unittest.cc
index 389c3a2..1702ef7 100644
--- a/webrtc/base/win32regkey_unittest.cc
+++ b/webrtc/base/win32regkey_unittest.cc
@@ -10,6 +10,7 @@
// Unittest for registry access API
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/common.h"
#include "webrtc/base/win32regkey.h"
@@ -564,8 +565,8 @@
#ifdef IS_PRIVATE_BUILD
// get a temp file name
wchar_t temp_path[MAX_PATH] = {0};
- EXPECT_LT(::GetTempPath(ARRAY_SIZE(temp_path), temp_path),
- static_cast<DWORD>(ARRAY_SIZE(temp_path)));
+ EXPECT_LT(::GetTempPath(arraysize(temp_path), temp_path),
+ static_cast<DWORD>(arraysize(temp_path)));
wchar_t temp_file[MAX_PATH] = {0};
EXPECT_NE(::GetTempFileName(temp_path, L"rkut_",
::GetTickCount(), temp_file), 0);
diff --git a/webrtc/base/win32socketserver.cc b/webrtc/base/win32socketserver.cc
index f466bf1..72ce4eb 100644
--- a/webrtc/base/win32socketserver.cc
+++ b/webrtc/base/win32socketserver.cc
@@ -55,7 +55,7 @@
static const int ICMP_PING_TIMEOUT_MILLIS = 10000u;
// TODO: Enable for production builds also? Use FormatMessage?
-#ifdef _DEBUG
+#if !defined(NDEBUG)
LPCSTR WSAErrorToString(int error, LPCSTR *description_result) {
LPCSTR string = "Unspecified";
LPCSTR description = "Unspecified description";
@@ -626,7 +626,7 @@
case FD_CONNECT:
if (error != ERROR_SUCCESS) {
ReportWSAError("WSAAsync:connect notify", error, addr_);
-#ifdef _DEBUG
+#if !defined(NDEBUG)
int32_t duration = TimeSince(connect_time_);
LOG(LS_INFO) << "WSAAsync:connect error (" << duration
<< " ms), faking close";
@@ -639,7 +639,7 @@
// though the connect event never did occur.
SignalCloseEvent(this, error);
} else {
-#ifdef _DEBUG
+#if !defined(NDEBUG)
int32_t duration = TimeSince(connect_time_);
LOG(LS_INFO) << "WSAAsync:connect (" << duration << " ms)";
#endif
diff --git a/webrtc/base/win32windowpicker.cc b/webrtc/base/win32windowpicker.cc
index b4550ae..da05a5c 100644
--- a/webrtc/base/win32windowpicker.cc
+++ b/webrtc/base/win32windowpicker.cc
@@ -12,6 +12,7 @@
#include <string>
#include <vector>
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
@@ -58,7 +59,7 @@
}
TCHAR window_title[500];
- GetWindowText(hwnd, window_title, ARRAY_SIZE(window_title));
+ GetWindowText(hwnd, window_title, arraysize(window_title));
std::string title = ToUtf8(window_title);
WindowId id(hwnd);
diff --git a/webrtc/base/win32windowpicker_unittest.cc b/webrtc/base/win32windowpicker_unittest.cc
index 71e8af6..701bb27 100644
--- a/webrtc/base/win32windowpicker_unittest.cc
+++ b/webrtc/base/win32windowpicker_unittest.cc
@@ -7,6 +7,7 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
@@ -71,7 +72,7 @@
EXPECT_EQ(window_picker.visible_window()->handle(), desc.id().id());
TCHAR window_title[500];
GetWindowText(window_picker.visible_window()->handle(), window_title,
- ARRAY_SIZE(window_title));
+ arraysize(window_title));
EXPECT_EQ(0, wcscmp(window_title, kVisibleWindowTitle));
}
diff --git a/webrtc/build/android/AndroidManifest.xml b/webrtc/build/android/AndroidManifest.xml
new file mode 100644
index 0000000..bb6d354
--- /dev/null
+++ b/webrtc/build/android/AndroidManifest.xml
@@ -0,0 +1,14 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+ This is a dummy manifest which is required by:
+ 1. aapt when generating R.java in java.gypi:
+ Nothing in the manifest is used, but it is still required by aapt.
+ 2. lint: [min|target]SdkVersion are required by lint and should
+ be kept up-to-date.
+-->
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+ package="dummy.package">
+
+ <uses-sdk android:minSdkVersion="16" android:targetSdkVersion="23" />
+
+</manifest>
diff --git a/webrtc/build/android/suppressions.xml b/webrtc/build/android/suppressions.xml
new file mode 100644
index 0000000..0fc22e0
--- /dev/null
+++ b/webrtc/build/android/suppressions.xml
@@ -0,0 +1,23 @@
+<?xml version="1.0" encoding="utf-8"?>
+<lint>
+ <!-- These lint settings is for the Android linter that gets run by
+ lint_action.gypi on compile of WebRTC java code. All WebRTC java code
+ should lint cleanly for the issues below. -->
+ <!-- TODO(phoglund): make work with suppress.py or remove printout referring
+ to suppress.py. -->
+ <issue id="NewApi"></issue>
+
+ <issue id="Locale" severity="ignore"/>
+ <issue id="SdCardPath" severity="ignore"/>
+ <issue id="UseValueOf" severity="ignore"/>
+ <issue id="InlinedApi" severity="ignore"/>
+ <issue id="DefaultLocale" severity="ignore"/>
+ <issue id="Assert" severity="ignore"/>
+ <issue id="UseSparseArrays" severity="ignore"/>
+
+ <!-- These are just from the dummy AndroidManifest.xml we use for linting.
+ It's in the same directory as this file. -->
+ <issue id="MissingApplicationIcon" severity="ignore"/>
+ <issue id="AllowBackup" severity="ignore"/>
+ <issue id="MissingVersion" severity="ignore"/>
+</lint>
diff --git a/webrtc/build/android/test_runner.py b/webrtc/build/android/test_runner.py
index 7996982..78a7a19 100755
--- a/webrtc/build/android/test_runner.py
+++ b/webrtc/build/android/test_runner.py
@@ -38,6 +38,8 @@
'webrtc/common_audio/common_audio_unittests.isolate',
'common_video_unittests':
'webrtc/common_video/common_video_unittests.isolate',
+ 'libjingle_peerconnection_unittest':
+ 'talk/libjingle_peerconnection_unittest.isolate',
'modules_tests': 'webrtc/modules/modules_tests.isolate',
'modules_unittests': 'webrtc/modules/modules_unittests.isolate',
'rtc_unittests': 'webrtc/rtc_unittests.isolate',
@@ -48,8 +50,6 @@
'video_capture_tests':
'webrtc/modules/video_capture/video_capture_tests.isolate',
'video_engine_tests': 'webrtc/video_engine_tests.isolate',
- 'video_engine_core_unittests':
- 'webrtc/video_engine/video_engine_core_unittests.isolate',
'voice_engine_unittests':
'webrtc/voice_engine/voice_engine_unittests.isolate',
'webrtc_nonparallel_tests': 'webrtc/webrtc_nonparallel_tests.isolate',
diff --git a/webrtc/build/apk_test.gypi b/webrtc/build/apk_test.gypi
new file mode 100644
index 0000000..a41e436
--- /dev/null
+++ b/webrtc/build/apk_test.gypi
@@ -0,0 +1,40 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+# This is almost an identical copy of src/build/apk_test.gypi with minor
+# modifications to allow test executables starting with "lib".
+# See http://crbug.com/543820 for more details.
+
+{
+ 'dependencies': [
+ '<(DEPTH)/base/base.gyp:base_java',
+ '<(DEPTH)/build/android/pylib/device/commands/commands.gyp:chromium_commands',
+ '<(DEPTH)/build/android/pylib/remote/device/dummy/dummy.gyp:remote_device_dummy_apk',
+ '<(DEPTH)/testing/android/appurify_support.gyp:appurify_support_java',
+ '<(DEPTH)/testing/android/on_device_instrumentation.gyp:reporter_java',
+ '<(DEPTH)/tools/android/android_tools.gyp:android_tools',
+ ],
+ 'conditions': [
+ ['OS == "android"', {
+ 'variables': {
+ # These are used to configure java_apk.gypi included below.
+ 'test_type': 'gtest',
+ 'apk_name': '<(test_suite_name)',
+ 'intermediate_dir': '<(PRODUCT_DIR)/<(test_suite_name)_apk',
+ 'final_apk_path': '<(intermediate_dir)/<(test_suite_name)-debug.apk',
+ 'java_in_dir': '<(DEPTH)/testing/android/native_test/java',
+ 'native_lib_target': '<(test_suite_name)',
+ 'gyp_managed_install': 0,
+ },
+ 'includes': [
+ '../../build/java_apk.gypi',
+ '../../build/android/test_runner.gypi',
+ ],
+ }], # 'OS == "android"
+ ], # conditions
+}
diff --git a/webrtc/build/apk_tests.gyp b/webrtc/build/apk_tests.gyp
index a3833ff..02a1342 100644
--- a/webrtc/build/apk_tests.gyp
+++ b/webrtc/build/apk_tests.gyp
@@ -61,6 +61,23 @@
],
},
{
+ 'target_name': 'libjingle_peerconnection_unittest_apk',
+ 'type': 'none',
+ 'variables': {
+ 'test_suite_name': 'libjingle_peerconnection_unittest',
+ 'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)libjingle_peerconnection_unittest<(SHARED_LIB_SUFFIX)',
+ },
+ 'dependencies': [
+ '<(DEPTH)/talk/libjingle_tests.gyp:libjingle_peerconnection_unittest',
+ '<(DEPTH)/talk/libjingle.gyp:libjingle_peerconnection_java',
+ ],
+ 'includes': [
+ # Use webrtc copy of apk_test.gypi to allow test executables starting
+ # with "lib". See http://crbug.com/543820 for more details.
+ '../build/apk_test.gypi',
+ ],
+ },
+ {
'target_name': 'modules_tests_apk',
'type': 'none',
'variables': {
@@ -146,20 +163,6 @@
],
},
{
- 'target_name': 'video_engine_core_unittests_apk',
- 'type': 'none',
- 'variables': {
- 'test_suite_name': 'video_engine_core_unittests',
- 'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)video_engine_core_unittests<(SHARED_LIB_SUFFIX)',
- },
- 'dependencies': [
- '<(webrtc_root)/video_engine/video_engine_core_unittests.gyp:video_engine_core_unittests',
- ],
- 'includes': [
- '../../build/apk_test.gypi',
- ],
- },
- {
'target_name': 'video_engine_tests_apk',
'type': 'none',
'variables': {
diff --git a/webrtc/build/apk_tests_noop.gyp b/webrtc/build/apk_tests_noop.gyp
index 719bddb..ed9249a 100644
--- a/webrtc/build/apk_tests_noop.gyp
+++ b/webrtc/build/apk_tests_noop.gyp
@@ -22,6 +22,10 @@
'type': 'none',
},
{
+ 'target_name': 'libjingle_peerconnection_unittest_apk',
+ 'type': 'none',
+ },
+ {
'target_name': 'modules_tests_apk',
'type': 'none',
},
@@ -46,10 +50,6 @@
'type': 'none',
},
{
- 'target_name': 'video_engine_core_unittests_apk',
- 'type': 'none',
- },
- {
'target_name': 'video_engine_tests_apk',
'type': 'none',
},
diff --git a/webrtc/build/common.gypi b/webrtc/build/common.gypi
index 2b05168..8d8583f 100644
--- a/webrtc/build/common.gypi
+++ b/webrtc/build/common.gypi
@@ -21,12 +21,10 @@
'conditions': [
['build_with_chromium==1', {
- 'build_with_libjingle': 1,
'webrtc_root%': '<(DEPTH)/third_party/webrtc',
'apk_tests_path%': '<(DEPTH)/third_party/webrtc/build/apk_tests_noop.gyp',
'modules_java_gyp_path%': '<(DEPTH)/third_party/webrtc/modules/modules_java_chromium.gyp',
}, {
- 'build_with_libjingle%': 0,
'webrtc_root%': '<(DEPTH)/webrtc',
'apk_tests_path%': '<(DEPTH)/webrtc/build/apk_tests.gyp',
'modules_java_gyp_path%': '<(DEPTH)/webrtc/modules/modules_java.gyp',
@@ -34,7 +32,6 @@
],
},
'build_with_chromium%': '<(build_with_chromium)',
- 'build_with_libjingle%': '<(build_with_libjingle)',
'webrtc_root%': '<(webrtc_root)',
'apk_tests_path%': '<(apk_tests_path)',
'modules_java_gyp_path%': '<(modules_java_gyp_path)',
@@ -47,7 +44,6 @@
'build_with_mozilla%': 0,
},
'build_with_chromium%': '<(build_with_chromium)',
- 'build_with_libjingle%': '<(build_with_libjingle)',
'build_with_mozilla%': '<(build_with_mozilla)',
'webrtc_root%': '<(webrtc_root)',
'apk_tests_path%': '<(apk_tests_path)',
@@ -89,15 +85,14 @@
# Disable these to not build components which can be externally provided.
'build_expat%': 1,
- 'build_icu%': 1,
'build_json%': 1,
'build_libjpeg%': 1,
'build_libvpx%': 1,
'build_libyuv%': 1,
'build_openmax_dl%': 1,
'build_opus%': 1,
+ 'build_protobuf%': 1,
'build_ssl%': 1,
- 'build_vp9%': 1,
# Disable by default
'have_dbus_glib%': 0,
@@ -129,6 +124,17 @@
# Enabling this may break interop with Android clients that support H264.
'use_objc_h264%': 0,
+ # Enable this to build H.264 encoder/decoder using third party libraries.
+ # Encoding uses OpenH264 and decoding uses FFmpeg. Because of this, OpenH264
+ # and FFmpeg have to be correctly enabled separately.
+ # - use_openh264=1 is required for OpenH264 targets to be defined.
+ # - ffmpeg_branding=Chrome is one way to support H.264 decoding in FFmpeg.
+ # FFmpeg can be built with/without H.264 support, see 'ffmpeg_branding'.
+ # Without it, it compiles but H264DecoderImpl fails to initialize.
+ # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
+ # http://www.openh264.org, https://www.ffmpeg.org/
+ 'use_third_party_h264%': 0, # TODO(hbos): To be used in follow-up CL(s).
+
'conditions': [
['build_with_chromium==1', {
# Exclude pulse audio on Chromium since its prerequisites don't require
@@ -137,6 +143,10 @@
# Exclude internal ADM since Chromium uses its own IO handling.
'include_internal_audio_device%': 0,
+
+ # Remove tests for Chromium to avoid slowing down GYP generation.
+ 'include_tests%': 0,
+ 'restrict_webrtc_logging%': 1,
}, { # Settings for the standalone (not-in-Chromium) build.
# TODO(andrew): For now, disable the Chrome plugins, which causes a
# flood of chromium-style warnings. Investigate enabling them:
@@ -145,17 +155,11 @@
'include_pulse_audio%': 1,
'include_internal_audio_device%': 1,
- }],
- ['build_with_libjingle==1', {
- 'include_tests%': 0,
- 'restrict_webrtc_logging%': 1,
- }, {
'include_tests%': 1,
'restrict_webrtc_logging%': 0,
}],
['OS=="ios"', {
'build_libjpeg%': 0,
- 'enable_protobuf%': 0,
}],
['target_arch=="arm" or target_arch=="arm64"', {
'prefer_fixed_point%': 1,
diff --git a/webrtc/build/protoc.gypi b/webrtc/build/protoc.gypi
index 5e486f1..682bc22 100644
--- a/webrtc/build/protoc.gypi
+++ b/webrtc/build/protoc.gypi
@@ -109,10 +109,6 @@
'process_outputs_as_sources': 1,
},
],
- 'dependencies': [
- '<(DEPTH)/third_party/protobuf/protobuf.gyp:protoc#host',
- '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
- ],
'include_dirs': [
'<(SHARED_INTERMEDIATE_DIR)/protoc_out',
'<(DEPTH)',
@@ -123,12 +119,20 @@
'<(DEPTH)',
]
},
- 'export_dependent_settings': [
- # The generated headers reference headers within protobuf_lite,
- # so dependencies must be able to find those headers too.
- '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
- ],
# This target exports a hard dependency because it generates header
# files.
'hard_dependency': 1,
+ 'conditions': [
+ ['build_protobuf==1', {
+ 'dependencies': [
+ '<(DEPTH)/third_party/protobuf/protobuf.gyp:protoc#host',
+ '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
+ ],
+ 'export_dependent_settings': [
+ # The generated headers reference headers within protobuf_lite,
+ # so dependencies must be able to find those headers too.
+ '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
+ ],
+ }],
+ ],
}
diff --git a/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc b/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc
index 4022160..1150990 100644
--- a/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc
+++ b/webrtc/build/sanitizers/tsan_suppressions_webrtc.cc
@@ -42,6 +42,10 @@
"race:webrtc/modules/audio_processing/aec/aec_core.c\n"
"race:webrtc/modules/audio_processing/aec/aec_rdft.c\n"
+// Race in pulse initialization.
+// https://code.google.com/p/webrtc/issues/detail?id=5152
+"race:webrtc::AudioDeviceLinuxPulse::Init\n"
+
// rtc_unittest
// https://code.google.com/p/webrtc/issues/detail?id=3911 for details.
"race:rtc::AsyncInvoker::OnMessage\n"
@@ -68,6 +72,9 @@
// TODO(jiayl): https://code.google.com/p/webrtc/issues/detail?id=3492
"race:user_sctp_timer_iterate\n"
+// https://code.google.com/p/webrtc/issues/detail?id=5151
+"race:sctp_close\n"
+
// Potential deadlocks detected after roll in r6516.
// https://code.google.com/p/webrtc/issues/detail?id=3509
"deadlock:webrtc::RTCPReceiver::SetSsrcs\n"
@@ -85,7 +92,7 @@
// Race between InitCpuFlags and TestCpuFlag in libyuv.
// https://code.google.com/p/libyuv/issues/detail?id=508
-"race:libyuv::TestCpuFlag\n"
+"race:InitCpuFlags\n"
// End of suppressions.
; // Please keep this semicolon.
diff --git a/webrtc/build/webrtc.gni b/webrtc/build/webrtc.gni
index 1d33e89..c55f423 100644
--- a/webrtc/build/webrtc.gni
+++ b/webrtc/build/webrtc.gni
@@ -36,7 +36,6 @@
# Disable these to not build components which can be externally provided.
rtc_build_expat = true
- rtc_build_icu = true
rtc_build_json = true
rtc_build_libjpeg = true
rtc_build_libvpx = true
@@ -44,7 +43,6 @@
rtc_build_openmax_dl = true
rtc_build_opus = true
rtc_build_ssl = true
- rtc_build_vp9 = true
# Disable by default.
rtc_have_dbus_glib = false
@@ -92,6 +90,17 @@
# Enable this to use HW H.264 encoder/decoder on iOS PeerConnections.
# Enabling this may break interop with Android clients that support H264.
rtc_use_objc_h264 = false
+
+ # Enable this to build H.264 encoder/decoder using third party libraries.
+ # Encoding uses OpenH264 and decoding uses FFmpeg. Because of this, OpenH264
+ # and FFmpeg have to be correctly enabled separately.
+ # - use_openh264=true is required for OpenH264 targets to be defined.
+ # - ffmpeg_branding="Chrome" is one way to support H.264 decoding in FFmpeg.
+ # FFmpeg can be built with/without H.264 support, see 'ffmpeg_branding'.
+ # Without it, it compiles but H264DecoderImpl fails to initialize.
+ # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
+ # http://www.openh264.org, https://www.ffmpeg.org/
+ use_third_party_h264 = false # TODO(hbos): To be used in follow-up CL(s).
}
# Make it possible to provide custom locations for some libraries (move these
diff --git a/webrtc/call.h b/webrtc/call.h
index e6e8cde..313c5e5 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -16,16 +16,14 @@
#include "webrtc/common_types.h"
#include "webrtc/audio_receive_stream.h"
#include "webrtc/audio_send_stream.h"
+#include "webrtc/audio_state.h"
#include "webrtc/base/socket.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
namespace webrtc {
-class AudioDeviceModule;
class AudioProcessing;
-class VoiceEngine;
-class VoiceEngineObserver;
const char* Version();
@@ -74,9 +72,6 @@
struct Config {
static const int kDefaultStartBitrateBps;
- // VoiceEngine used for audio/video synchronization for this Call.
- VoiceEngine* voice_engine = nullptr;
-
// Bitrate config used until valid bitrate estimates are calculated. Also
// used to cap total bitrate used.
struct BitrateConfig {
@@ -85,11 +80,13 @@
int max_bitrate_bps = -1;
} bitrate_config;
- struct AudioConfig {
- AudioDeviceModule* audio_device_module = nullptr;
- AudioProcessing* audio_processing = nullptr;
- VoiceEngineObserver* voice_engine_observer = nullptr;
- } audio_config;
+ // AudioState which is possibly shared between multiple calls.
+ // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+ rtc::scoped_refptr<AudioState> audio_state;
+
+ // Audio Processing Module to be used in this call.
+ // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+ AudioProcessing* audio_processing = nullptr;
};
struct Stats {
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 3abc762..498c724 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -10,6 +10,7 @@
source_set("call") {
sources = [
+ "bitrate_allocator.cc",
"call.cc",
"congestion_controller.cc",
"transport_adapter.cc",
diff --git a/webrtc/call/bitrate_allocator.cc b/webrtc/call/bitrate_allocator.cc
new file mode 100644
index 0000000..b3789d3
--- /dev/null
+++ b/webrtc/call/bitrate_allocator.cc
@@ -0,0 +1,194 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "webrtc/call/bitrate_allocator.h"
+
+#include <algorithm>
+#include <utility>
+
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+
+namespace webrtc {
+
+// Allow packets to be transmitted in up to 2 times max video bitrate if the
+// bandwidth estimate allows it.
+const int kTransmissionMaxBitrateMultiplier = 2;
+const int kDefaultBitrateBps = 300000;
+
+BitrateAllocator::BitrateAllocator()
+ : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
+ bitrate_observers_(),
+ bitrate_observers_modified_(false),
+ enforce_min_bitrate_(true),
+ last_bitrate_bps_(kDefaultBitrateBps),
+ last_fraction_loss_(0),
+ last_rtt_(0) {}
+
+uint32_t BitrateAllocator::OnNetworkChanged(uint32_t bitrate,
+ uint8_t fraction_loss,
+ int64_t rtt) {
+ CriticalSectionScoped lock(crit_sect_.get());
+ last_bitrate_bps_ = bitrate;
+ last_fraction_loss_ = fraction_loss;
+ last_rtt_ = rtt;
+ uint32_t allocated_bitrate_bps = 0;
+ ObserverBitrateMap allocation = AllocateBitrates();
+ for (const auto& kv : allocation) {
+ kv.first->OnNetworkChanged(kv.second, last_fraction_loss_, last_rtt_);
+ allocated_bitrate_bps += kv.second;
+ }
+ return allocated_bitrate_bps;
+}
+
+BitrateAllocator::ObserverBitrateMap BitrateAllocator::AllocateBitrates() {
+ if (bitrate_observers_.empty())
+ return ObserverBitrateMap();
+
+ uint32_t sum_min_bitrates = 0;
+ for (const auto& observer : bitrate_observers_)
+ sum_min_bitrates += observer.second.min_bitrate;
+ if (last_bitrate_bps_ <= sum_min_bitrates)
+ return LowRateAllocation(last_bitrate_bps_);
+ else
+ return NormalRateAllocation(last_bitrate_bps_, sum_min_bitrates);
+}
+
+int BitrateAllocator::AddBitrateObserver(BitrateObserver* observer,
+ uint32_t min_bitrate_bps,
+ uint32_t max_bitrate_bps) {
+ CriticalSectionScoped lock(crit_sect_.get());
+
+ BitrateObserverConfList::iterator it =
+ FindObserverConfigurationPair(observer);
+
+ // Allow the max bitrate to be exceeded for FEC and retransmissions.
+ // TODO(holmer): We have to get rid of this hack as it makes it difficult to
+ // properly allocate bitrate. The allocator should instead distribute any
+ // extra bitrate after all streams have maxed out.
+ max_bitrate_bps *= kTransmissionMaxBitrateMultiplier;
+ if (it != bitrate_observers_.end()) {
+ // Update current configuration.
+ it->second.min_bitrate = min_bitrate_bps;
+ it->second.max_bitrate = max_bitrate_bps;
+ } else {
+ // Add new settings.
+ bitrate_observers_.push_back(BitrateObserverConfiguration(
+ observer, BitrateConfiguration(min_bitrate_bps, max_bitrate_bps)));
+ bitrate_observers_modified_ = true;
+ }
+
+ ObserverBitrateMap allocation = AllocateBitrates();
+ int new_observer_bitrate_bps = 0;
+ for (auto& kv : allocation) {
+ kv.first->OnNetworkChanged(kv.second, last_fraction_loss_, last_rtt_);
+ if (kv.first == observer)
+ new_observer_bitrate_bps = kv.second;
+ }
+ return new_observer_bitrate_bps;
+}
+
+void BitrateAllocator::RemoveBitrateObserver(BitrateObserver* observer) {
+ CriticalSectionScoped lock(crit_sect_.get());
+ BitrateObserverConfList::iterator it =
+ FindObserverConfigurationPair(observer);
+ if (it != bitrate_observers_.end()) {
+ bitrate_observers_.erase(it);
+ bitrate_observers_modified_ = true;
+ }
+}
+
+void BitrateAllocator::GetMinMaxBitrateSumBps(int* min_bitrate_sum_bps,
+ int* max_bitrate_sum_bps) const {
+ *min_bitrate_sum_bps = 0;
+ *max_bitrate_sum_bps = 0;
+
+ CriticalSectionScoped lock(crit_sect_.get());
+ for (const auto& observer : bitrate_observers_) {
+ *min_bitrate_sum_bps += observer.second.min_bitrate;
+ *max_bitrate_sum_bps += observer.second.max_bitrate;
+ }
+}
+
+BitrateAllocator::BitrateObserverConfList::iterator
+BitrateAllocator::FindObserverConfigurationPair(
+ const BitrateObserver* observer) {
+ for (auto it = bitrate_observers_.begin(); it != bitrate_observers_.end();
+ ++it) {
+ if (it->first == observer)
+ return it;
+ }
+ return bitrate_observers_.end();
+}
+
+void BitrateAllocator::EnforceMinBitrate(bool enforce_min_bitrate) {
+ CriticalSectionScoped lock(crit_sect_.get());
+ enforce_min_bitrate_ = enforce_min_bitrate;
+}
+
+BitrateAllocator::ObserverBitrateMap BitrateAllocator::NormalRateAllocation(
+ uint32_t bitrate,
+ uint32_t sum_min_bitrates) {
+ uint32_t number_of_observers =
+ static_cast<uint32_t>(bitrate_observers_.size());
+ uint32_t bitrate_per_observer =
+ (bitrate - sum_min_bitrates) / number_of_observers;
+ // Use map to sort list based on max bitrate.
+ ObserverSortingMap list_max_bitrates;
+ for (const auto& observer : bitrate_observers_) {
+ list_max_bitrates.insert(std::pair<uint32_t, ObserverConfiguration>(
+ observer.second.max_bitrate,
+ ObserverConfiguration(observer.first, observer.second.min_bitrate)));
+ }
+ ObserverBitrateMap allocation;
+ ObserverSortingMap::iterator max_it = list_max_bitrates.begin();
+ while (max_it != list_max_bitrates.end()) {
+ number_of_observers--;
+ uint32_t observer_allowance =
+ max_it->second.min_bitrate + bitrate_per_observer;
+ if (max_it->first < observer_allowance) {
+ // We have more than enough for this observer.
+ // Carry the remainder forward.
+ uint32_t remainder = observer_allowance - max_it->first;
+ if (number_of_observers != 0) {
+ bitrate_per_observer += remainder / number_of_observers;
+ }
+ allocation[max_it->second.observer] = max_it->first;
+ } else {
+ allocation[max_it->second.observer] = observer_allowance;
+ }
+ list_max_bitrates.erase(max_it);
+ // Prepare next iteration.
+ max_it = list_max_bitrates.begin();
+ }
+ return allocation;
+}
+
+BitrateAllocator::ObserverBitrateMap BitrateAllocator::LowRateAllocation(
+ uint32_t bitrate) {
+ ObserverBitrateMap allocation;
+ if (enforce_min_bitrate_) {
+ // Min bitrate to all observers.
+ for (const auto& observer : bitrate_observers_)
+ allocation[observer.first] = observer.second.min_bitrate;
+ } else {
+ // Allocate up to |min_bitrate| to one observer at a time, until
+ // |bitrate| is depleted.
+ uint32_t remainder = bitrate;
+ for (const auto& observer : bitrate_observers_) {
+ uint32_t allocated_bitrate =
+ std::min(remainder, observer.second.min_bitrate);
+ allocation[observer.first] = allocated_bitrate;
+ remainder -= allocated_bitrate;
+ }
+ }
+ return allocation;
+}
+} // namespace webrtc
diff --git a/webrtc/call/bitrate_allocator.h b/webrtc/call/bitrate_allocator.h
new file mode 100644
index 0000000..4a3fd59
--- /dev/null
+++ b/webrtc/call/bitrate_allocator.h
@@ -0,0 +1,102 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ *
+ * Usage: this class will register multiple RtcpBitrateObserver's one at each
+ * RTCP module. It will aggregate the results and run one bandwidth estimation
+ * and push the result to the encoders via BitrateObserver(s).
+ */
+
+#ifndef WEBRTC_CALL_BITRATE_ALLOCATOR_H_
+#define WEBRTC_CALL_BITRATE_ALLOCATOR_H_
+
+#include <list>
+#include <map>
+#include <utility>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+
+class BitrateObserver;
+
+class BitrateAllocator {
+ public:
+ BitrateAllocator();
+
+ // Allocate target_bitrate across the registered BitrateObservers.
+ // Returns actual bitrate allocated (might be higher than target_bitrate if
+ // for instance EnforceMinBitrate() is enabled.
+ uint32_t OnNetworkChanged(uint32_t target_bitrate,
+ uint8_t fraction_loss,
+ int64_t rtt);
+
+ // Set the start and max send bitrate used by the bandwidth management.
+ //
+ // |observer| updates bitrates if already in use.
+ // |min_bitrate_bps| = 0 equals no min bitrate.
+ // |max_bitrate_bps| = 0 equals no max bitrate.
+ // Returns bitrate allocated for the bitrate observer.
+ int AddBitrateObserver(BitrateObserver* observer,
+ uint32_t min_bitrate_bps,
+ uint32_t max_bitrate_bps);
+
+ void RemoveBitrateObserver(BitrateObserver* observer);
+
+ void GetMinMaxBitrateSumBps(int* min_bitrate_sum_bps,
+ int* max_bitrate_sum_bps) const;
+
+ // This method controls the behavior when the available bitrate is lower than
+ // the minimum bitrate, or the sum of minimum bitrates.
+ // When true, the bitrate will never be set lower than the minimum bitrate(s).
+ // When false, the bitrate observers will be allocated rates up to their
+ // respective minimum bitrate, satisfying one observer after the other.
+ void EnforceMinBitrate(bool enforce_min_bitrate);
+
+ private:
+ struct BitrateConfiguration {
+ BitrateConfiguration(uint32_t min_bitrate, uint32_t max_bitrate)
+ : min_bitrate(min_bitrate), max_bitrate(max_bitrate) {}
+ uint32_t min_bitrate;
+ uint32_t max_bitrate;
+ };
+ struct ObserverConfiguration {
+ ObserverConfiguration(BitrateObserver* observer, uint32_t bitrate)
+ : observer(observer), min_bitrate(bitrate) {}
+ BitrateObserver* const observer;
+ uint32_t min_bitrate;
+ };
+ typedef std::pair<BitrateObserver*, BitrateConfiguration>
+ BitrateObserverConfiguration;
+ typedef std::list<BitrateObserverConfiguration> BitrateObserverConfList;
+ typedef std::multimap<uint32_t, ObserverConfiguration> ObserverSortingMap;
+ typedef std::map<BitrateObserver*, int> ObserverBitrateMap;
+
+ BitrateObserverConfList::iterator FindObserverConfigurationPair(
+ const BitrateObserver* observer) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+ ObserverBitrateMap AllocateBitrates() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+ ObserverBitrateMap NormalRateAllocation(uint32_t bitrate,
+ uint32_t sum_min_bitrates)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ ObserverBitrateMap LowRateAllocation(uint32_t bitrate)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+ rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ // Stored in a list to keep track of the insertion order.
+ BitrateObserverConfList bitrate_observers_ GUARDED_BY(crit_sect_);
+ bool bitrate_observers_modified_ GUARDED_BY(crit_sect_);
+ bool enforce_min_bitrate_ GUARDED_BY(crit_sect_);
+ uint32_t last_bitrate_bps_ GUARDED_BY(crit_sect_);
+ uint8_t last_fraction_loss_ GUARDED_BY(crit_sect_);
+ int64_t last_rtt_ GUARDED_BY(crit_sect_);
+};
+} // namespace webrtc
+#endif // WEBRTC_CALL_BITRATE_ALLOCATOR_H_
diff --git a/webrtc/call/bitrate_allocator_unittest.cc b/webrtc/call/bitrate_allocator_unittest.cc
new file mode 100644
index 0000000..86f75a4
--- /dev/null
+++ b/webrtc/call/bitrate_allocator_unittest.cc
@@ -0,0 +1,212 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/call/bitrate_allocator.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+
+namespace webrtc {
+
+class TestBitrateObserver : public BitrateObserver {
+ public:
+ TestBitrateObserver()
+ : last_bitrate_(0), last_fraction_loss_(0), last_rtt_(0) {}
+
+ virtual void OnNetworkChanged(uint32_t bitrate,
+ uint8_t fraction_loss,
+ int64_t rtt) {
+ last_bitrate_ = bitrate;
+ last_fraction_loss_ = fraction_loss;
+ last_rtt_ = rtt;
+ }
+ uint32_t last_bitrate_;
+ uint8_t last_fraction_loss_;
+ int64_t last_rtt_;
+};
+
+class BitrateAllocatorTest : public ::testing::Test {
+ protected:
+ BitrateAllocatorTest() : allocator_(new BitrateAllocator()) {
+ allocator_->OnNetworkChanged(300000u, 0, 0);
+ }
+ ~BitrateAllocatorTest() {}
+
+ rtc::scoped_ptr<BitrateAllocator> allocator_;
+};
+
+TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
+ TestBitrateObserver bitrate_observer;
+ int start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer, 100000, 1500000);
+ EXPECT_EQ(300000, start_bitrate);
+ allocator_->OnNetworkChanged(200000, 0, 0);
+ EXPECT_EQ(200000u, bitrate_observer.last_bitrate_);
+
+ // TODO(pbos): Expect capping to 1.5M instead of 3M when not boosting the max
+ // bitrate for FEC/retransmissions (see todo in BitrateAllocator).
+ allocator_->OnNetworkChanged(4000000, 0, 0);
+ EXPECT_EQ(3000000u, bitrate_observer.last_bitrate_);
+ start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer, 100000, 4000000);
+ EXPECT_EQ(4000000, start_bitrate);
+
+ start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer, 100000, 1500000);
+ EXPECT_EQ(3000000, start_bitrate);
+ EXPECT_EQ(3000000u, bitrate_observer.last_bitrate_);
+ allocator_->OnNetworkChanged(1500000, 0, 0);
+ EXPECT_EQ(1500000u, bitrate_observer.last_bitrate_);
+}
+
+TEST_F(BitrateAllocatorTest, TwoBitrateObserversOneRtcpObserver) {
+ TestBitrateObserver bitrate_observer_1;
+ TestBitrateObserver bitrate_observer_2;
+ int start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 300000);
+ EXPECT_EQ(300000, start_bitrate);
+ start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 300000);
+ EXPECT_EQ(200000, start_bitrate);
+
+ // Test too low start bitrate, hence lower than sum of min. Min bitrates will
+ // be allocated to all observers.
+ allocator_->OnNetworkChanged(200000, 0, 50);
+ EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
+ EXPECT_EQ(0, bitrate_observer_1.last_fraction_loss_);
+ EXPECT_EQ(50, bitrate_observer_1.last_rtt_);
+ EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_);
+ EXPECT_EQ(0, bitrate_observer_2.last_fraction_loss_);
+ EXPECT_EQ(50, bitrate_observer_2.last_rtt_);
+
+ // Test a bitrate which should be distributed equally.
+ allocator_->OnNetworkChanged(500000, 0, 50);
+ const uint32_t kBitrateToShare = 500000 - 200000 - 100000;
+ EXPECT_EQ(100000u + kBitrateToShare / 2, bitrate_observer_1.last_bitrate_);
+ EXPECT_EQ(200000u + kBitrateToShare / 2, bitrate_observer_2.last_bitrate_);
+
+ // Limited by 2x max bitrates since we leave room for FEC and retransmissions.
+ allocator_->OnNetworkChanged(1500000, 0, 50);
+ EXPECT_EQ(600000u, bitrate_observer_1.last_bitrate_);
+ EXPECT_EQ(600000u, bitrate_observer_2.last_bitrate_);
+}
+
+class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
+ protected:
+ BitrateAllocatorTestNoEnforceMin() : allocator_(new BitrateAllocator()) {
+ allocator_->EnforceMinBitrate(false);
+ allocator_->OnNetworkChanged(300000u, 0, 0);
+ }
+ ~BitrateAllocatorTestNoEnforceMin() {}
+
+ rtc::scoped_ptr<BitrateAllocator> allocator_;
+};
+
+// The following three tests verify that the EnforceMinBitrate() method works
+// as intended.
+TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserver) {
+ TestBitrateObserver bitrate_observer_1;
+ int start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 400000);
+ EXPECT_EQ(300000, start_bitrate);
+
+ // High REMB.
+ allocator_->OnNetworkChanged(150000, 0, 0);
+ EXPECT_EQ(150000u, bitrate_observer_1.last_bitrate_);
+
+ // Low REMB.
+ allocator_->OnNetworkChanged(10000, 0, 0);
+ EXPECT_EQ(10000u, bitrate_observer_1.last_bitrate_);
+
+ allocator_->RemoveBitrateObserver(&bitrate_observer_1);
+}
+
+TEST_F(BitrateAllocatorTestNoEnforceMin, ThreeBitrateObservers) {
+ TestBitrateObserver bitrate_observer_1;
+ TestBitrateObserver bitrate_observer_2;
+ TestBitrateObserver bitrate_observer_3;
+ // Set up the observers with min bitrates at 100000, 200000, and 300000.
+ int start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 400000);
+ EXPECT_EQ(300000, start_bitrate);
+
+ start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 400000);
+ EXPECT_EQ(200000, start_bitrate);
+ EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
+
+ start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer_3, 300000, 400000);
+ EXPECT_EQ(0, start_bitrate);
+ EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
+ EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_);
+
+ // High REMB. Make sure the controllers get a fair share of the surplus
+ // (i.e., what is left after each controller gets its min rate).
+ allocator_->OnNetworkChanged(690000, 0, 0);
+ // Verify that each observer gets its min rate (sum of min rates is 600000),
+ // and that the remaining 90000 is divided equally among the three.
+ uint32_t bitrate_to_share = 690000u - 100000u - 200000u - 300000u;
+ EXPECT_EQ(100000u + bitrate_to_share / 3, bitrate_observer_1.last_bitrate_);
+ EXPECT_EQ(200000u + bitrate_to_share / 3, bitrate_observer_2.last_bitrate_);
+ EXPECT_EQ(300000u + bitrate_to_share / 3, bitrate_observer_3.last_bitrate_);
+
+ // High REMB, but below the sum of min bitrates.
+ allocator_->OnNetworkChanged(500000, 0, 0);
+ // Verify that the first and second observers get their min bitrates, and the
+ // third gets the remainder.
+ EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_); // Min bitrate.
+ EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_); // Min bitrate.
+ EXPECT_EQ(200000u, bitrate_observer_3.last_bitrate_); // Remainder.
+
+ // Low REMB.
+ allocator_->OnNetworkChanged(10000, 0, 0);
+ // Verify that the first observer gets all the rate, and the rest get zero.
+ EXPECT_EQ(10000u, bitrate_observer_1.last_bitrate_);
+ EXPECT_EQ(0u, bitrate_observer_2.last_bitrate_);
+ EXPECT_EQ(0u, bitrate_observer_3.last_bitrate_);
+
+ allocator_->RemoveBitrateObserver(&bitrate_observer_1);
+ allocator_->RemoveBitrateObserver(&bitrate_observer_2);
+ allocator_->RemoveBitrateObserver(&bitrate_observer_3);
+}
+
+TEST_F(BitrateAllocatorTest, ThreeBitrateObserversLowRembEnforceMin) {
+ TestBitrateObserver bitrate_observer_1;
+ TestBitrateObserver bitrate_observer_2;
+ TestBitrateObserver bitrate_observer_3;
+ int start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer_1, 100000, 400000);
+ EXPECT_EQ(300000, start_bitrate);
+
+ start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer_2, 200000, 400000);
+ EXPECT_EQ(200000, start_bitrate);
+ EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_);
+
+ start_bitrate =
+ allocator_->AddBitrateObserver(&bitrate_observer_3, 300000, 400000);
+ EXPECT_EQ(300000, start_bitrate);
+ EXPECT_EQ(100000, static_cast<int>(bitrate_observer_1.last_bitrate_));
+ EXPECT_EQ(200000, static_cast<int>(bitrate_observer_2.last_bitrate_));
+
+ // Low REMB. Verify that all observers still get their respective min bitrate.
+ allocator_->OnNetworkChanged(1000, 0, 0);
+ EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_); // Min cap.
+ EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_); // Min cap.
+ EXPECT_EQ(300000u, bitrate_observer_3.last_bitrate_); // Min cap.
+
+ allocator_->RemoveBitrateObserver(&bitrate_observer_1);
+ allocator_->RemoveBitrateObserver(&bitrate_observer_2);
+ allocator_->RemoveBitrateObserver(&bitrate_observer_3);
+}
+} // namespace webrtc
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index 685f3fd..4b24bbd 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -13,66 +13,54 @@
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/audio_state.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/event.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/fake_voice_engine.h"
+#include "webrtc/test/mock_voice_engine.h"
#include "webrtc/test/frame_generator_capturer.h"
namespace webrtc {
namespace {
// Note: If you consider to re-use this class, think twice and instead consider
-// writing tests that don't depend on the trace system.
-class TraceObserver {
+// writing tests that don't depend on the logging system.
+class LogObserver {
public:
- TraceObserver() {
- Trace::set_level_filter(kTraceTerseInfo);
+ LogObserver() { rtc::LogMessage::AddLogToStream(&callback_, rtc::LS_INFO); }
- Trace::CreateTrace();
- Trace::SetTraceCallback(&callback_);
-
- // Call webrtc trace to initialize the tracer that would otherwise trigger a
- // data-race if left to be initialized by multiple threads (i.e. threads
- // spawned by test::DirectTransport members in BitrateEstimatorTest).
- WEBRTC_TRACE(kTraceStateInfo,
- kTraceUtility,
- -1,
- "Instantiate without data races.");
- }
-
- ~TraceObserver() {
- Trace::SetTraceCallback(nullptr);
- Trace::ReturnTrace();
- }
+ ~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }
void PushExpectedLogLine(const std::string& expected_log_line) {
callback_.PushExpectedLogLine(expected_log_line);
}
- EventTypeWrapper Wait() {
- return callback_.Wait();
- }
+ bool Wait() { return callback_.Wait(); }
private:
- class Callback : public TraceCallback {
+ class Callback : public rtc::LogSink {
public:
- Callback() : done_(EventWrapper::Create()) {}
+ Callback() : done_(false, false) {}
- void Print(TraceLevel level, const char* message, int length) override {
+ void OnLogMessage(const std::string& message) override {
rtc::CritScope lock(&crit_sect_);
- std::string msg(message);
- if (msg.find("BitrateEstimator") != std::string::npos) {
- received_log_lines_.push_back(msg);
+ // Ignore log lines that are due to missing AST extensions, these are
+ // logged when we switch back from AST to TOF until the wrapping bitrate
+ // estimator gives up on using AST.
+ if (message.find("BitrateEstimator") != std::string::npos &&
+ message.find("packet is missing") == std::string::npos) {
+ received_log_lines_.push_back(message);
}
+
int num_popped = 0;
while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
std::string a = received_log_lines_.front();
@@ -80,19 +68,17 @@
received_log_lines_.pop_front();
expected_log_lines_.pop_front();
num_popped++;
- EXPECT_TRUE(a.find(b) != std::string::npos);
+ EXPECT_TRUE(a.find(b) != std::string::npos) << a << " != " << b;
}
if (expected_log_lines_.size() <= 0) {
if (num_popped > 0) {
- done_->Set();
+ done_.Set();
}
return;
}
}
- EventTypeWrapper Wait() {
- return done_->Wait(test::CallTest::kDefaultTimeoutMs);
- }
+ bool Wait() { return done_.Wait(test::CallTest::kDefaultTimeoutMs); }
void PushExpectedLogLine(const std::string& expected_log_line) {
rtc::CritScope lock(&crit_sect_);
@@ -104,7 +90,7 @@
rtc::CriticalSection crit_sect_;
Strings received_log_lines_ GUARDED_BY(crit_sect_);
Strings expected_log_lines_ GUARDED_BY(crit_sect_);
- rtc::scoped_ptr<EventWrapper> done_;
+ rtc::Event done_;
};
Callback callback_;
@@ -118,13 +104,13 @@
public:
BitrateEstimatorTest() : receive_config_(nullptr) {}
- virtual ~BitrateEstimatorTest() {
- EXPECT_TRUE(streams_.empty());
- }
+ virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }
virtual void SetUp() {
+ AudioState::Config audio_state_config;
+ audio_state_config.voice_engine = &mock_voice_engine_;
Call::Config config;
- config.voice_engine = &fake_voice_engine_;
+ config.audio_state = AudioState::Create(audio_state_config);
receiver_call_.reset(Call::Create(config));
sender_call_.reset(Call::Create(config));
@@ -133,18 +119,19 @@
receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
receive_transport_->SetReceiver(sender_call_->Receiver());
- send_config_ = VideoSendStream::Config(send_transport_.get());
- send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
+ video_send_config_ = VideoSendStream::Config(send_transport_.get());
+ video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
// Encoders will be set separately per stream.
- send_config_.encoder_settings.encoder = nullptr;
- send_config_.encoder_settings.payload_name = "FAKE";
- send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
- encoder_config_.streams = test::CreateVideoStreams(1);
+ video_send_config_.encoder_settings.encoder = nullptr;
+ video_send_config_.encoder_settings.payload_name = "FAKE";
+ video_send_config_.encoder_settings.payload_type =
+ kFakeVideoSendPayloadType;
+ video_encoder_config_.streams = test::CreateVideoStreams(1);
receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
// receive_config_.decoders will be set by every stream separately.
- receive_config_.rtp.remote_ssrc = send_config_.rtp.ssrcs[0];
- receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
+ receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
+ receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
receive_config_.rtp.remb = true;
receive_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
@@ -154,7 +141,7 @@
virtual void TearDown() {
std::for_each(streams_.begin(), streams_.end(),
- std::mem_fun(&Stream::StopSending));
+ std::mem_fun(&Stream::StopSending));
send_transport_->StopSending();
receive_transport_->StopSending();
@@ -165,6 +152,7 @@
}
receiver_call_.reset();
+ sender_call_.reset();
}
protected:
@@ -181,23 +169,21 @@
frame_generator_capturer_(),
fake_encoder_(Clock::GetRealTimeClock()),
fake_decoder_() {
- test_->send_config_.rtp.ssrcs[0]++;
- test_->send_config_.encoder_settings.encoder = &fake_encoder_;
+ test_->video_send_config_.rtp.ssrcs[0]++;
+ test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
send_stream_ = test_->sender_call_->CreateVideoSendStream(
- test_->send_config_, test_->encoder_config_);
- RTC_DCHECK_EQ(1u, test_->encoder_config_.streams.size());
+ test_->video_send_config_, test_->video_encoder_config_);
+ RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size());
frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
- send_stream_->Input(),
- test_->encoder_config_.streams[0].width,
- test_->encoder_config_.streams[0].height,
- 30,
+ send_stream_->Input(), test_->video_encoder_config_.streams[0].width,
+ test_->video_encoder_config_.streams[0].height, 30,
Clock::GetRealTimeClock()));
send_stream_->Start();
frame_generator_capturer_->Start();
if (receive_audio) {
AudioReceiveStream::Config receive_config;
- receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
+ receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
// Bogus non-default id to prevent hitting a RTC_DCHECK when creating
// the AudioReceiveStream. Every receive stream has to correspond to
// an underlying channel id.
@@ -211,12 +197,13 @@
VideoReceiveStream::Decoder decoder;
decoder.decoder = &fake_decoder_;
decoder.payload_type =
- test_->send_config_.encoder_settings.payload_type;
+ test_->video_send_config_.encoder_settings.payload_type;
decoder.payload_name =
- test_->send_config_.encoder_settings.payload_name;
+ test_->video_send_config_.encoder_settings.payload_name;
+ test_->receive_config_.decoders.clear();
test_->receive_config_.decoders.push_back(decoder);
test_->receive_config_.rtp.remote_ssrc =
- test_->send_config_.rtp.ssrcs[0];
+ test_->video_send_config_.rtp.ssrcs[0];
test_->receive_config_.rtp.local_ssrc++;
video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
test_->receive_config_);
@@ -262,8 +249,8 @@
test::FakeDecoder fake_decoder_;
};
- test::FakeVoiceEngine fake_voice_engine_;
- TraceObserver receiver_trace_;
+ testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_;
+ LogObserver receiver_log_;
rtc::scoped_ptr<test::DirectTransport> send_transport_;
rtc::scoped_ptr<test::DirectTransport> receive_transport_;
rtc::scoped_ptr<Call> sender_call_;
@@ -278,89 +265,89 @@
"RemoteBitrateEstimatorSingleStream: Instantiating.";
TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForAudio) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, true));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTForAudio) {
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, true));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, true));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions[0] =
+ video_send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
TEST_F(BitrateEstimatorTest, SwitchesToASTThenBackToTOFForVideo) {
- send_config_.rtp.extensions.push_back(
+ video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions[0] =
+ video_send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId);
- receiver_trace_.PushExpectedLogLine("Switching to absolute send time RBE.");
- receiver_trace_.PushExpectedLogLine(kAbsSendTimeLog);
+ receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
+ receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
streams_.push_back(new Stream(this, false));
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
- send_config_.rtp.extensions[0] =
+ video_send_config_.rtp.extensions[0] =
RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
- receiver_trace_.PushExpectedLogLine(
+ receiver_log_.PushExpectedLogLine(
"WrappingBitrateEstimator: Switching to transmission time offset RBE.");
- receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
+ receiver_log_.PushExpectedLogLine(kSingleStreamLog);
streams_.push_back(new Stream(this, false));
streams_[0]->StopSending();
streams_[1]->StopSending();
- EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
+ EXPECT_TRUE(receiver_log_.Wait());
}
} // namespace webrtc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 594ddf5..5c46a48 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -15,27 +15,33 @@
#include "webrtc/audio/audio_receive_stream.h"
#include "webrtc/audio/audio_send_stream.h"
+#include "webrtc/audio/audio_state.h"
+#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/call.h"
+#include "webrtc/call/bitrate_allocator.h"
#include "webrtc/call/congestion_controller.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
#include "webrtc/config.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/video/call_stats.h"
#include "webrtc/video/video_receive_stream.h"
#include "webrtc/video/video_send_stream.h"
-#include "webrtc/video_engine/call_stats.h"
#include "webrtc/voice_engine/include/voe_codec.h"
namespace webrtc {
@@ -44,7 +50,8 @@
namespace internal {
-class Call : public webrtc::Call, public PacketReceiver {
+class Call : public webrtc::Call, public PacketReceiver,
+ public BitrateObserver {
public:
explicit Call(const Call::Config& config);
virtual ~Call();
@@ -83,6 +90,10 @@
void OnSentPacket(const rtc::SentPacket& sent_packet) override;
+ // Implements BitrateObserver.
+ void OnNetworkChanged(uint32_t bitrate_bps, uint8_t fraction_loss,
+ int64_t rtt_ms) override;
+
private:
DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
size_t length);
@@ -94,14 +105,28 @@
void ConfigureSync(const std::string& sync_group)
EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
+ VoiceEngine* voice_engine() {
+ internal::AudioState* audio_state =
+ static_cast<internal::AudioState*>(config_.audio_state.get());
+ if (audio_state)
+ return audio_state->voice_engine();
+ else
+ return nullptr;
+ }
+
+ void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
+ void UpdateReceiveHistograms();
+
+ Clock* const clock_;
+
const int num_cpu_cores_;
const rtc::scoped_ptr<ProcessThread> module_process_thread_;
const rtc::scoped_ptr<CallStats> call_stats_;
- const rtc::scoped_ptr<CongestionController> congestion_controller_;
+ const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
Call::Config config_;
rtc::ThreadChecker configuration_thread_checker_;
- bool network_enabled_;
+ bool network_enabled_;
rtc::scoped_ptr<RWLockWrapper> receive_crit_;
// Audio and Video receive streams are owned by the client that creates them.
@@ -123,7 +148,25 @@
VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;
RtcEventLog* event_log_ = nullptr;
- VoECodec* voe_codec_ = nullptr;
+
+ // The following members are only accessed (exclusively) from one thread and
+ // from the destructor, and therefore doesn't need any explicit
+ // synchronization.
+ int64_t received_video_bytes_;
+ int64_t received_audio_bytes_;
+ int64_t received_rtcp_bytes_;
+ int64_t first_rtp_packet_received_ms_;
+ int64_t last_rtp_packet_received_ms_;
+ int64_t first_packet_sent_ms_;
+
+ // TODO(holmer): Remove this lock once BitrateController no longer calls
+ // OnNetworkChanged from multiple threads.
+ rtc::CriticalSection bitrate_crit_;
+ int64_t estimated_send_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
+ int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_);
+ int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
+
+ const rtc::scoped_ptr<CongestionController> congestion_controller_;
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
};
@@ -136,15 +179,29 @@
namespace internal {
Call::Call(const Call::Config& config)
- : num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
+ : clock_(Clock::GetRealTimeClock()),
+ num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
- call_stats_(new CallStats()),
- congestion_controller_(new CongestionController(
- module_process_thread_.get(), call_stats_.get())),
+ call_stats_(new CallStats(clock_)),
+ bitrate_allocator_(new BitrateAllocator()),
config_(config),
network_enabled_(true),
receive_crit_(RWLockWrapper::CreateRWLock()),
- send_crit_(RWLockWrapper::CreateRWLock()) {
+ send_crit_(RWLockWrapper::CreateRWLock()),
+ received_video_bytes_(0),
+ received_audio_bytes_(0),
+ received_rtcp_bytes_(0),
+ first_rtp_packet_received_ms_(-1),
+ last_rtp_packet_received_ms_(-1),
+ first_packet_sent_ms_(-1),
+ estimated_send_bitrate_sum_kbits_(0),
+ pacer_bitrate_sum_kbits_(0),
+ num_bitrate_updates_(0),
+ congestion_controller_(
+ new CongestionController(module_process_thread_.get(),
+ call_stats_.get(),
+ this)) {
+ RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
config.bitrate_config.min_bitrate_bps);
@@ -152,12 +209,9 @@
RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
config.bitrate_config.start_bitrate_bps);
}
- if (config.voice_engine) {
- // Keep a reference to VoECodec, so we're sure the VoiceEngine lives for the
- // duration of the call.
- voe_codec_ = VoECodec::GetInterface(config.voice_engine);
- if (voe_codec_)
- event_log_ = voe_codec_->GetEventLog();
+ if (config.audio_state.get()) {
+ ScopedVoEInterface<VoECodec> voe_codec(voice_engine());
+ event_log_ = voe_codec->GetEventLog();
}
Trace::CreateTrace();
@@ -168,10 +222,14 @@
config_.bitrate_config.min_bitrate_bps,
config_.bitrate_config.start_bitrate_bps,
config_.bitrate_config.max_bitrate_bps);
+
+ congestion_controller_->GetBitrateController()->SetEventLog(event_log_);
}
Call::~Call() {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
+ UpdateSendHistograms();
+ UpdateReceiveHistograms();
RTC_CHECK(audio_send_ssrcs_.empty());
RTC_CHECK(video_send_ssrcs_.empty());
RTC_CHECK(video_send_streams_.empty());
@@ -182,9 +240,53 @@
module_process_thread_->DeRegisterModule(call_stats_.get());
module_process_thread_->Stop();
Trace::ReturnTrace();
+}
- if (voe_codec_)
- voe_codec_->Release();
+void Call::UpdateSendHistograms() {
+ if (num_bitrate_updates_ == 0 || first_packet_sent_ms_ == -1)
+ return;
+ int64_t elapsed_sec =
+ (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
+ if (elapsed_sec < metrics::kMinRunTimeInSeconds)
+ return;
+ int send_bitrate_kbps =
+ estimated_send_bitrate_sum_kbits_ / num_bitrate_updates_;
+ int pacer_bitrate_kbps = pacer_bitrate_sum_kbits_ / num_bitrate_updates_;
+ if (send_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
+ send_bitrate_kbps);
+ }
+ if (pacer_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.PacerBitrateInKbps",
+ pacer_bitrate_kbps);
+ }
+}
+
+void Call::UpdateReceiveHistograms() {
+ if (first_rtp_packet_received_ms_ == -1)
+ return;
+ int64_t elapsed_sec =
+ (last_rtp_packet_received_ms_ - first_rtp_packet_received_ms_) / 1000;
+ if (elapsed_sec < metrics::kMinRunTimeInSeconds)
+ return;
+ int audio_bitrate_kbps = received_audio_bytes_ * 8 / elapsed_sec / 1000;
+ int video_bitrate_kbps = received_video_bytes_ * 8 / elapsed_sec / 1000;
+ int rtcp_bitrate_bps = received_rtcp_bytes_ * 8 / elapsed_sec;
+ if (video_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
+ video_bitrate_kbps);
+ }
+ if (audio_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
+ audio_bitrate_kbps);
+ }
+ if (rtcp_bitrate_bps > 0) {
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
+ rtcp_bitrate_bps);
+ }
+ RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+ "WebRTC.Call.BitrateReceivedInKbps",
+ audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000);
}
PacketReceiver* Call::Receiver() {
@@ -198,8 +300,8 @@
const webrtc::AudioSendStream::Config& config) {
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
- AudioSendStream* send_stream =
- new AudioSendStream(config, config_.voice_engine);
+ AudioSendStream* send_stream = new AudioSendStream(
+ config, config_.audio_state, congestion_controller_.get());
if (!network_enabled_)
send_stream->SignalNetworkState(kNetworkDown);
{
@@ -234,8 +336,7 @@
TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
AudioReceiveStream* receive_stream = new AudioReceiveStream(
- congestion_controller_->GetRemoteBitrateEstimator(false), config,
- config_.voice_engine);
+ congestion_controller_.get(), config, config_.audio_state);
{
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ -279,8 +380,8 @@
// the call has already started.
VideoSendStream* send_stream = new VideoSendStream(
num_cpu_cores_, module_process_thread_.get(), call_stats_.get(),
- congestion_controller_.get(), config, encoder_config,
- suspended_video_send_ssrcs_);
+ congestion_controller_.get(), bitrate_allocator_.get(), config,
+ encoder_config, suspended_video_send_ssrcs_);
if (!network_enabled_)
send_stream->SignalNetworkState(kNetworkDown);
@@ -338,7 +439,7 @@
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
VideoReceiveStream* receive_stream = new VideoReceiveStream(
num_cpu_cores_, congestion_controller_.get(), config,
- config_.voice_engine, module_process_thread_.get(), call_stats_.get());
+ voice_engine(), module_process_thread_.get(), call_stats_.get());
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ -463,12 +564,48 @@
}
void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
+ if (first_packet_sent_ms_ == -1)
+ first_packet_sent_ms_ = clock_->TimeInMilliseconds();
congestion_controller_->OnSentPacket(sent_packet);
}
+void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
+ int64_t rtt_ms) {
+ uint32_t allocated_bitrate_bps = bitrate_allocator_->OnNetworkChanged(
+ target_bitrate_bps, fraction_loss, rtt_ms);
+
+ int pad_up_to_bitrate_bps = 0;
+ {
+ ReadLockScoped read_lock(*send_crit_);
+ // No need to update as long as we're not sending.
+ if (video_send_streams_.empty())
+ return;
+
+ for (VideoSendStream* stream : video_send_streams_)
+ pad_up_to_bitrate_bps += stream->GetPaddingNeededBps();
+ }
+ // Allocated bitrate might be higher than bitrate estimate if enforcing min
+ // bitrate, or lower if estimate is higher than the sum of max bitrates, so
+ // set the pacer bitrate to the maximum of the two.
+ uint32_t pacer_bitrate_bps =
+ std::max(target_bitrate_bps, allocated_bitrate_bps);
+ {
+ rtc::CritScope lock(&bitrate_crit_);
+ // We only update these stats if we have send streams, and assume that
+ // OnNetworkChanged is called roughly with a fixed frequency.
+ estimated_send_bitrate_sum_kbits_ += target_bitrate_bps / 1000;
+ pacer_bitrate_sum_kbits_ += pacer_bitrate_bps / 1000;
+ ++num_bitrate_updates_;
+ }
+ congestion_controller_->UpdatePacerBitrate(
+ target_bitrate_bps / 1000,
+ PacedSender::kDefaultPaceMultiplier * pacer_bitrate_bps / 1000,
+ pad_up_to_bitrate_bps / 1000);
+}
+
void Call::ConfigureSync(const std::string& sync_group) {
// Set sync only if there was no previous one.
- if (config_.voice_engine == nullptr || sync_group.empty())
+ if (voice_engine() == nullptr || sync_group.empty())
return;
AudioReceiveStream* sync_audio_stream = nullptr;
@@ -506,10 +643,10 @@
}
// Only sync the first A/V pair within this sync group.
if (sync_audio_stream != nullptr && num_synced_streams == 1) {
- video_stream->SetSyncChannel(config_.voice_engine,
+ video_stream->SetSyncChannel(voice_engine(),
sync_audio_stream->config().voe_channel_id);
} else {
- video_stream->SetSyncChannel(config_.voice_engine, -1);
+ video_stream->SetSyncChannel(voice_engine(), -1);
}
}
}
@@ -517,10 +654,12 @@
PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
const uint8_t* packet,
size_t length) {
+ TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
// TODO(pbos): Figure out what channel needs it actually.
// Do NOT broadcast! Also make sure it's a valid packet.
// Return DELIVERY_UNKNOWN_SSRC if it can be determined that
// there's no receiver of the packet.
+ received_rtcp_bytes_ += length;
bool rtcp_delivered = false;
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
ReadLockScoped read_lock(*receive_crit_);
@@ -549,16 +688,21 @@
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) {
+ TRACE_EVENT0("webrtc", "Call::DeliverRtp");
// Minimum RTP header size.
if (length < 12)
return DELIVERY_PACKET_ERROR;
- uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
+ last_rtp_packet_received_ms_ = clock_->TimeInMilliseconds();
+ if (first_rtp_packet_received_ms_ == -1)
+ first_rtp_packet_received_ms_ = last_rtp_packet_received_ms_;
+ uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
ReadLockScoped read_lock(*receive_crit_);
if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
auto it = audio_receive_ssrcs_.find(ssrc);
if (it != audio_receive_ssrcs_.end()) {
+ received_audio_bytes_ += length;
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
@@ -570,6 +714,7 @@
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
auto it = video_receive_ssrcs_.find(ssrc);
if (it != video_receive_ssrcs_.end()) {
+ received_video_bytes_ += length;
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index c37b83b..3adcb10 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -18,8 +18,10 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/common.h"
+#include "webrtc/config.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
@@ -172,7 +174,7 @@
false);
}
if (time_since_creation > kMinRunTimeMs)
- observation_complete_->Set();
+ observation_complete_.Set();
}
}
@@ -189,6 +191,8 @@
void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
const char* kSyncGroup = "av_sync";
+ const uint32_t kAudioSendSsrc = 1234;
+ const uint32_t kAudioRecvSsrc = 5678;
class AudioPacketReceiver : public PacketReceiver {
public:
AudioPacketReceiver(int channel, VoENetwork* voe_network)
@@ -228,35 +232,45 @@
test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
audio_filename);
EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
- int channel = voe_base->CreateChannel();
+ Config voe_config;
+ voe_config.Set<VoicePacing>(new VoicePacing(true));
+ int send_channel_id = voe_base->CreateChannel(voe_config);
+ int recv_channel_id = voe_base->CreateChannel();
SyncRtcpObserver audio_observer;
+ AudioState::Config send_audio_state_config;
+ send_audio_state_config.voice_engine = voice_engine;
+ Call::Config sender_config;
+ sender_config.audio_state = AudioState::Create(send_audio_state_config);
Call::Config receiver_config;
- receiver_config.voice_engine = voice_engine;
- CreateCalls(Call::Config(), receiver_config);
+ receiver_config.audio_state = sender_config.audio_state;
+ CreateCalls(sender_config, receiver_config);
- CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
- EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac));
-
- AudioPacketReceiver voe_packet_receiver(channel, voe_network);
+ AudioPacketReceiver voe_send_packet_receiver(send_channel_id, voe_network);
+ AudioPacketReceiver voe_recv_packet_receiver(recv_channel_id, voe_network);
FakeNetworkPipe::Config net_config;
net_config.queue_delay_ms = 500;
net_config.loss_percent = 5;
test::PacketTransport audio_send_transport(
nullptr, &audio_observer, test::PacketTransport::kSender, net_config);
- audio_send_transport.SetReceiver(&voe_packet_receiver);
+ audio_send_transport.SetReceiver(&voe_recv_packet_receiver);
test::PacketTransport audio_receive_transport(
nullptr, &audio_observer, test::PacketTransport::kReceiver, net_config);
- audio_receive_transport.SetReceiver(&voe_packet_receiver);
+ audio_receive_transport.SetReceiver(&voe_send_packet_receiver);
- internal::TransportAdapter transport_adapter(&audio_send_transport);
- transport_adapter.Enable();
- EXPECT_EQ(0,
- voe_network->RegisterExternalTransport(channel, transport_adapter));
+ internal::TransportAdapter send_transport_adapter(&audio_send_transport);
+ send_transport_adapter.Enable();
+ EXPECT_EQ(0, voe_network->RegisterExternalTransport(send_channel_id,
+ send_transport_adapter));
- VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), channel,
+ internal::TransportAdapter recv_transport_adapter(&audio_receive_transport);
+ recv_transport_adapter.Enable();
+ EXPECT_EQ(0, voe_network->RegisterExternalTransport(recv_channel_id,
+ recv_transport_adapter));
+
+ VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), recv_channel_id,
voe_sync, &audio_observer);
test::PacketTransport sync_send_transport(sender_call_.get(), &observer,
@@ -270,34 +284,45 @@
test::FakeDecoder fake_decoder;
- CreateSendConfig(1, &sync_send_transport);
+ CreateSendConfig(1, 0, &sync_send_transport);
CreateMatchingReceiveConfigs(&sync_receive_transport);
- send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+ AudioSendStream::Config audio_send_config(&audio_send_transport);
+ audio_send_config.voe_channel_id = send_channel_id;
+ audio_send_config.rtp.ssrc = kAudioSendSsrc;
+ AudioSendStream* audio_send_stream =
+ sender_call_->CreateAudioSendStream(audio_send_config);
+
+ CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
+ EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac));
+
+ video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
if (fec) {
- send_config_.rtp.fec.red_payload_type = kRedPayloadType;
- send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
- receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
- receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+ video_send_config_.rtp.fec.red_payload_type = kRedPayloadType;
+ video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
+ video_receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
+ video_receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
}
- receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
- receive_configs_[0].renderer = &observer;
- receive_configs_[0].sync_group = kSyncGroup;
+ video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
+ video_receive_configs_[0].renderer = &observer;
+ video_receive_configs_[0].sync_group = kSyncGroup;
- AudioReceiveStream::Config audio_config;
- audio_config.voe_channel_id = channel;
- audio_config.sync_group = kSyncGroup;
+ AudioReceiveStream::Config audio_recv_config;
+ audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
+ audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
+ audio_recv_config.voe_channel_id = recv_channel_id;
+ audio_recv_config.sync_group = kSyncGroup;
- AudioReceiveStream* audio_receive_stream = nullptr;
+ AudioReceiveStream* audio_receive_stream;
if (create_audio_first) {
audio_receive_stream =
- receiver_call_->CreateAudioReceiveStream(audio_config);
- CreateStreams();
+ receiver_call_->CreateAudioReceiveStream(audio_recv_config);
+ CreateVideoStreams();
} else {
- CreateStreams();
+ CreateVideoStreams();
audio_receive_stream =
- receiver_call_->CreateAudioReceiveStream(audio_config);
+ receiver_call_->CreateAudioReceiveStream(audio_recv_config);
}
CreateFrameGeneratorCapturer();
@@ -305,16 +330,16 @@
Start();
fake_audio_device.Start();
- EXPECT_EQ(0, voe_base->StartPlayout(channel));
- EXPECT_EQ(0, voe_base->StartReceive(channel));
- EXPECT_EQ(0, voe_base->StartSend(channel));
+ EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
+ EXPECT_EQ(0, voe_base->StartReceive(recv_channel_id));
+ EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
- EXPECT_EQ(kEventSignaled, observer.Wait())
+ EXPECT_TRUE(observer.Wait())
<< "Timed out while waiting for audio and video to be synchronized.";
- EXPECT_EQ(0, voe_base->StopSend(channel));
- EXPECT_EQ(0, voe_base->StopReceive(channel));
- EXPECT_EQ(0, voe_base->StopPlayout(channel));
+ EXPECT_EQ(0, voe_base->StopSend(send_channel_id));
+ EXPECT_EQ(0, voe_base->StopReceive(recv_channel_id));
+ EXPECT_EQ(0, voe_base->StopPlayout(recv_channel_id));
fake_audio_device.Stop();
Stop();
@@ -323,16 +348,18 @@
audio_send_transport.StopSending();
audio_receive_transport.StopSending();
- voe_base->DeleteChannel(channel);
+ DestroyStreams();
+
+ sender_call_->DestroyAudioSendStream(audio_send_stream);
+ receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
+
+ voe_base->DeleteChannel(send_channel_id);
+ voe_base->DeleteChannel(recv_channel_id);
voe_base->Release();
voe_codec->Release();
voe_network->Release();
voe_sync->Release();
- DestroyStreams();
-
- receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
-
DestroyCalls();
VoiceEngine::Delete(voice_engine);
@@ -357,8 +384,12 @@
class CaptureNtpTimeObserver : public test::EndToEndTest,
public VideoRenderer {
public:
- CaptureNtpTimeObserver(int threshold_ms, int start_time_ms, int run_time_ms)
+ CaptureNtpTimeObserver(const FakeNetworkPipe::Config& net_config,
+ int threshold_ms,
+ int start_time_ms,
+ int run_time_ms)
: EndToEndTest(kLongTimeoutMs),
+ net_config_(net_config),
clock_(Clock::GetRealTimeClock()),
threshold_ms_(threshold_ms),
start_time_ms_(start_time_ms),
@@ -369,6 +400,16 @@
rtp_start_timestamp_(0) {}
private:
+ test::PacketTransport* CreateSendTransport(Call* sender_call) override {
+ return new test::PacketTransport(
+ sender_call, this, test::PacketTransport::kSender, net_config_);
+ }
+
+ test::PacketTransport* CreateReceiveTransport() override {
+ return new test::PacketTransport(
+ nullptr, this, test::PacketTransport::kReceiver, net_config_);
+ }
+
void RenderFrame(const VideoFrame& video_frame,
int time_to_render_ms) override {
rtc::CritScope lock(&crit_);
@@ -386,7 +427,7 @@
}
if (time_since_creation > run_time_ms_) {
- observation_complete_->Set();
+ observation_complete_.Set();
}
FrameCaptureTimeList::iterator iter =
@@ -437,21 +478,23 @@
capturer_ = frame_generator_capturer;
}
- void ModifyConfigs(VideoSendStream::Config* send_config,
- std::vector<VideoReceiveStream::Config>* receive_configs,
- VideoEncoderConfig* encoder_config) override {
+ void ModifyVideoConfigs(
+ VideoSendStream::Config* send_config,
+ std::vector<VideoReceiveStream::Config>* receive_configs,
+ VideoEncoderConfig* encoder_config) override {
(*receive_configs)[0].renderer = this;
// Enable the receiver side rtt calculation.
(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
}
void PerformTest() override {
- EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for "
- "estimated capture NTP time to be "
- "within bounds.";
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for "
+ "estimated capture NTP time to be "
+ "within bounds.";
}
rtc::CriticalSection crit_;
+ const FakeNetworkPipe::Config net_config_;
Clock* const clock_;
int threshold_ms_;
int start_time_ms_;
@@ -462,9 +505,9 @@
uint32_t rtp_start_timestamp_;
typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
FrameCaptureTimeList capture_time_list_ GUARDED_BY(&crit_);
- } test(threshold_ms, start_time_ms, run_time_ms);
+ } test(net_config, threshold_ms, start_time_ms, run_time_ms);
- RunBaseTest(&test, net_config);
+ RunBaseTest(&test);
}
TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
@@ -501,26 +544,26 @@
void OnLoadUpdate(Load load) override {
if (load == tested_load_)
- observation_complete_->Set();
+ observation_complete_.Set();
}
- void ModifyConfigs(VideoSendStream::Config* send_config,
- std::vector<VideoReceiveStream::Config>* receive_configs,
- VideoEncoderConfig* encoder_config) override {
+ void ModifyVideoConfigs(
+ VideoSendStream::Config* send_config,
+ std::vector<VideoReceiveStream::Config>* receive_configs,
+ VideoEncoderConfig* encoder_config) override {
send_config->overuse_callback = this;
send_config->encoder_settings.encoder = &encoder_;
}
void PerformTest() override {
- EXPECT_EQ(kEventSignaled, Wait())
- << "Timed out before receiving an overuse callback.";
+ EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback.";
}
LoadObserver::Load tested_load_;
test::DelayedEncoder encoder_;
} test(tested_load, encode_delay_ms);
- RunBaseTest(&test, FakeNetworkPipe::Config());
+ RunBaseTest(&test);
}
TEST_F(CallPerfTest, ReceivesCpuUnderuse) {
@@ -581,21 +624,22 @@
}
if (num_bitrate_observations_in_range_ ==
kNumBitrateObservationsInRange)
- observation_complete_->Set();
+ observation_complete_.Set();
}
}
return SEND_PACKET;
}
- void OnStreamsCreated(
+ void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
- void ModifyConfigs(VideoSendStream::Config* send_config,
- std::vector<VideoReceiveStream::Config>* receive_configs,
- VideoEncoderConfig* encoder_config) override {
+ void ModifyVideoConfigs(
+ VideoSendStream::Config* send_config,
+ std::vector<VideoReceiveStream::Config>* receive_configs,
+ VideoEncoderConfig* encoder_config) override {
if (pad_to_min_bitrate_) {
encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
} else {
@@ -604,8 +648,7 @@
}
void PerformTest() override {
- EXPECT_EQ(kEventSignaled, Wait())
- << "Timeout while waiting for send-bitrate stats.";
+ EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats.";
}
VideoSendStream* send_stream_;
@@ -614,7 +657,7 @@
} test(pad_to_min_bitrate);
fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps);
- RunBaseTest(&test, FakeNetworkPipe::Config());
+ RunBaseTest(&test);
}
TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); }
@@ -633,7 +676,7 @@
BitrateObserver()
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
- time_to_reconfigure_(webrtc::EventWrapper::Create()),
+ time_to_reconfigure_(false, false),
encoder_inits_(0),
last_set_bitrate_(0),
send_stream_(nullptr) {}
@@ -652,7 +695,7 @@
last_set_bitrate_,
kPermittedReconfiguredBitrateDiffKbps)
<< "Encoder reconfigured with bitrate too far away from last set.";
- observation_complete_->Set();
+ observation_complete_.Set();
}
return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
}
@@ -662,7 +705,7 @@
last_set_bitrate_ = new_target_bitrate_kbps;
if (encoder_inits_ == 1 &&
new_target_bitrate_kbps > kReconfigureThresholdKbps) {
- time_to_reconfigure_->Set();
+ time_to_reconfigure_.Set();
}
return FakeEncoder::SetRates(new_target_bitrate_kbps, framerate);
}
@@ -673,9 +716,10 @@
return config;
}
- void ModifyConfigs(VideoSendStream::Config* send_config,
- std::vector<VideoReceiveStream::Config>* receive_configs,
- VideoEncoderConfig* encoder_config) override {
+ void ModifyVideoConfigs(
+ VideoSendStream::Config* send_config,
+ std::vector<VideoReceiveStream::Config>* receive_configs,
+ VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder = this;
encoder_config->streams[0].min_bitrate_bps = 50000;
encoder_config->streams[0].target_bitrate_bps =
@@ -684,32 +728,32 @@
encoder_config_ = *encoder_config;
}
- void OnStreamsCreated(
+ void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void PerformTest() override {
- ASSERT_EQ(kEventSignaled, time_to_reconfigure_->Wait(kDefaultTimeoutMs))
+ ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs))
<< "Timed out before receiving an initial high bitrate.";
encoder_config_.streams[0].width *= 2;
encoder_config_.streams[0].height *= 2;
EXPECT_TRUE(send_stream_->ReconfigureVideoEncoder(encoder_config_));
- EXPECT_EQ(kEventSignaled, Wait())
+ EXPECT_TRUE(Wait())
<< "Timed out while waiting for a couple of high bitrate estimates "
"after reconfiguring the send stream.";
}
private:
- rtc::scoped_ptr<webrtc::EventWrapper> time_to_reconfigure_;
+ rtc::Event time_to_reconfigure_;
int encoder_inits_;
uint32_t last_set_bitrate_;
VideoSendStream* send_stream_;
VideoEncoderConfig encoder_config_;
} test;
- RunBaseTest(&test, FakeNetworkPipe::Config());
+ RunBaseTest(&test);
}
} // namespace webrtc
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index 9819b53..75c8238 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -12,22 +12,25 @@
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/audio_state.h"
#include "webrtc/call.h"
-#include "webrtc/test/fake_voice_engine.h"
+#include "webrtc/test/mock_voice_engine.h"
namespace {
struct CallHelper {
- CallHelper() : voice_engine_(new webrtc::test::FakeVoiceEngine()) {
+ CallHelper() {
+ webrtc::AudioState::Config audio_state_config;
+ audio_state_config.voice_engine = &voice_engine_;
webrtc::Call::Config config;
- config.voice_engine = voice_engine_.get();
+ config.audio_state = webrtc::AudioState::Create(audio_state_config);
call_.reset(webrtc::Call::Create(config));
}
webrtc::Call* operator->() { return call_.get(); }
private:
- rtc::scoped_ptr<webrtc::test::FakeVoiceEngine> voice_engine_;
+ testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
rtc::scoped_ptr<webrtc::Call> call_;
};
} // namespace
diff --git a/webrtc/call/congestion_controller.cc b/webrtc/call/congestion_controller.cc
index 1ec361e..c442667 100644
--- a/webrtc/call/congestion_controller.cc
+++ b/webrtc/call/congestion_controller.cc
@@ -11,23 +11,24 @@
#include "webrtc/call/congestion_controller.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common.h"
-#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/pacing/include/packet_router.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/pacing/packet_router.h"
#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h"
#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
-#include "webrtc/video_engine/call_stats.h"
-#include "webrtc/video_engine/payload_router.h"
-#include "webrtc/video_engine/vie_encoder.h"
-#include "webrtc/video_engine/vie_remb.h"
+#include "webrtc/video/call_stats.h"
+#include "webrtc/video/payload_router.h"
+#include "webrtc/video/vie_encoder.h"
+#include "webrtc/video/vie_remb.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
namespace webrtc {
@@ -144,9 +145,9 @@
} // namespace
CongestionController::CongestionController(ProcessThread* process_thread,
- CallStats* call_stats)
- : remb_(new VieRemb()),
- bitrate_allocator_(new BitrateAllocator()),
+ CallStats* call_stats,
+ BitrateObserver* bitrate_observer)
+ : remb_(new VieRemb(Clock::GetRealTimeClock())),
packet_router_(new PacketRouter()),
pacer_(new PacedSender(Clock::GetRealTimeClock(),
packet_router_.get(),
@@ -166,7 +167,7 @@
// construction.
bitrate_controller_(
BitrateController::CreateBitrateController(Clock::GetRealTimeClock(),
- this)),
+ bitrate_observer)),
min_bitrate_bps_(RemoteBitrateEstimator::kDefaultMinBitrateBps) {
call_stats_->RegisterStatsObserver(remote_bitrate_estimator_.get());
@@ -249,6 +250,12 @@
return transport_feedback_adapter_.get();
}
+void CongestionController::UpdatePacerBitrate(int bitrate_kbps,
+ int max_bitrate_kbps,
+ int min_bitrate_kbps) {
+ pacer_->UpdateBitrate(bitrate_kbps, max_bitrate_kbps, min_bitrate_kbps);
+}
+
int64_t CongestionController::GetPacerQueuingDelayMs() const {
return pacer_->QueueInMs();
}
@@ -278,23 +285,6 @@
}
}
-// TODO(mflodman): Move this logic out from CongestionController.
-void CongestionController::OnNetworkChanged(uint32_t target_bitrate_bps,
- uint8_t fraction_loss,
- int64_t rtt) {
- bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt);
- int pad_up_to_bitrate_bps = 0;
- {
- rtc::CritScope lock(&encoder_crit_);
- for (const auto& encoder : encoders_)
- pad_up_to_bitrate_bps += encoder->GetPaddingNeededBps();
- }
- pacer_->UpdateBitrate(
- target_bitrate_bps / 1000,
- PacedSender::kDefaultPaceMultiplier * target_bitrate_bps / 1000,
- pad_up_to_bitrate_bps / 1000);
-}
-
void CongestionController::OnSentPacket(const rtc::SentPacket& sent_packet) {
if (transport_feedback_adapter_) {
transport_feedback_adapter_->OnSentPacket(sent_packet.packet_id,
diff --git a/webrtc/call/congestion_controller.h b/webrtc/call/congestion_controller.h
index b424234..b77c46f 100644
--- a/webrtc/call/congestion_controller.h
+++ b/webrtc/call/congestion_controller.h
@@ -16,12 +16,12 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/socket.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/stream.h"
namespace webrtc {
-class BitrateAllocator;
+class BitrateController;
+class BitrateObserver;
class CallStats;
class Config;
class PacedSender;
@@ -32,43 +32,43 @@
class RtpRtcp;
class SendStatisticsProxy;
class TransportFeedbackAdapter;
+class TransportFeedbackObserver;
class ViEEncoder;
class VieRemb;
-class CongestionController : public BitrateObserver {
+class CongestionController {
public:
- CongestionController(ProcessThread* process_thread, CallStats* call_stats);
- ~CongestionController();
- void AddEncoder(ViEEncoder* encoder);
- void RemoveEncoder(ViEEncoder* encoder);
- void SetBweBitrates(int min_bitrate_bps,
- int start_bitrate_bps,
- int max_bitrate_bps);
+ CongestionController(ProcessThread* process_thread, CallStats* call_stats,
+ BitrateObserver* bitrate_observer);
+ virtual ~CongestionController();
+ virtual void AddEncoder(ViEEncoder* encoder);
+ virtual void RemoveEncoder(ViEEncoder* encoder);
+ virtual void SetBweBitrates(int min_bitrate_bps,
+ int start_bitrate_bps,
+ int max_bitrate_bps);
- void SetChannelRembStatus(bool sender, bool receiver, RtpRtcp* rtp_module);
+ virtual void SetChannelRembStatus(bool sender,
+ bool receiver,
+ RtpRtcp* rtp_module);
- void SignalNetworkState(NetworkState state);
+ virtual void SignalNetworkState(NetworkState state);
- BitrateController* GetBitrateController() const;
- RemoteBitrateEstimator* GetRemoteBitrateEstimator(bool send_side_bwe) const;
- int64_t GetPacerQueuingDelayMs() const;
- PacedSender* pacer() const { return pacer_.get(); }
- PacketRouter* packet_router() const { return packet_router_.get(); }
- BitrateAllocator* bitrate_allocator() const {
- return bitrate_allocator_.get(); }
- TransportFeedbackObserver* GetTransportFeedbackObserver();
+ virtual BitrateController* GetBitrateController() const;
+ virtual RemoteBitrateEstimator* GetRemoteBitrateEstimator(
+ bool send_side_bwe) const;
+ virtual int64_t GetPacerQueuingDelayMs() const;
+ virtual PacedSender* pacer() const { return pacer_.get(); }
+ virtual PacketRouter* packet_router() const { return packet_router_.get(); }
+ virtual TransportFeedbackObserver* GetTransportFeedbackObserver();
- // Implements BitrateObserver.
- void OnNetworkChanged(uint32_t target_bitrate_bps,
- uint8_t fraction_loss,
- int64_t rtt) override;
+ virtual void UpdatePacerBitrate(int bitrate_kbps,
+ int max_bitrate_kbps,
+ int min_bitrate_kbps);
- void OnSentPacket(const rtc::SentPacket& sent_packet);
+ virtual void OnSentPacket(const rtc::SentPacket& sent_packet);
private:
rtc::scoped_ptr<VieRemb> remb_;
- // TODO(mflodman): Move bitrate_allocator_ to Call.
- rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
rtc::scoped_ptr<PacketRouter> packet_router_;
rtc::scoped_ptr<PacedSender> pacer_;
rtc::scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
@@ -86,6 +86,8 @@
rtc::scoped_ptr<BitrateController> bitrate_controller_;
rtc::scoped_ptr<TransportFeedbackAdapter> transport_feedback_adapter_;
int min_bitrate_bps_;
+
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CongestionController);
};
} // namespace webrtc
diff --git a/webrtc/call/mock/mock_congestion_controller.h b/webrtc/call/mock/mock_congestion_controller.h
new file mode 100644
index 0000000..54014da
--- /dev/null
+++ b/webrtc/call/mock/mock_congestion_controller.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_MOCK_MOCK_CONGESTION_CONTROLLER_H_
+#define WEBRTC_CALL_MOCK_MOCK_CONGESTION_CONTROLLER_H_
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/call/congestion_controller.h"
+
+namespace webrtc {
+namespace test {
+
+class MockCongestionController : public CongestionController {
+ public:
+ MockCongestionController(ProcessThread* process_thread,
+ CallStats* call_stats,
+ BitrateObserver* bitrate_observer)
+ : CongestionController(process_thread, call_stats, bitrate_observer) {}
+ MOCK_METHOD1(AddEncoder, void(ViEEncoder* encoder));
+ MOCK_METHOD1(RemoveEncoder, void(ViEEncoder* encoder));
+ MOCK_METHOD3(SetBweBitrates,
+ void(int min_bitrate_bps,
+ int start_bitrate_bps,
+ int max_bitrate_bps));
+ MOCK_METHOD3(SetChannelRembStatus,
+ void(bool sender, bool receiver, RtpRtcp* rtp_module));
+ MOCK_METHOD1(SignalNetworkState, void(NetworkState state));
+ MOCK_CONST_METHOD0(GetBitrateController, BitrateController*());
+ MOCK_CONST_METHOD1(GetRemoteBitrateEstimator,
+ RemoteBitrateEstimator*(bool send_side_bwe));
+ MOCK_CONST_METHOD0(GetPacerQueuingDelayMs, int64_t());
+ MOCK_CONST_METHOD0(pacer, PacedSender*());
+ MOCK_CONST_METHOD0(packet_router, PacketRouter*());
+ MOCK_METHOD0(GetTransportFeedbackObserver, TransportFeedbackObserver*());
+ MOCK_METHOD3(UpdatePacerBitrate,
+ void(int bitrate_kbps,
+ int max_bitrate_kbps,
+ int min_bitrate_kbps));
+ MOCK_METHOD1(OnSentPacket, void(const rtc::SentPacket& sent_packet));
+
+ RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MockCongestionController);
+};
+} // namespace test
+} // namespace webrtc
+#endif // WEBRTC_CALL_MOCK_MOCK_CONGESTION_CONTROLLER_H_
diff --git a/webrtc/call/packet_injection_tests.cc b/webrtc/call/packet_injection_tests.cc
index 18ca058..277cd3e 100644
--- a/webrtc/call/packet_injection_tests.cc
+++ b/webrtc/call/packet_injection_tests.cc
@@ -40,22 +40,22 @@
CreateReceiverCall(Call::Config());
test::NullTransport null_transport;
- CreateSendConfig(1, &null_transport);
+ CreateSendConfig(1, 0, &null_transport);
CreateMatchingReceiveConfigs(&null_transport);
- receive_configs_[0].decoders[0].payload_type = payload_type;
+ video_receive_configs_[0].decoders[0].payload_type = payload_type;
switch (codec_type) {
case CodecType::kVp8:
- receive_configs_[0].decoders[0].payload_name = "VP8";
+ video_receive_configs_[0].decoders[0].payload_name = "VP8";
break;
case CodecType::kH264:
- receive_configs_[0].decoders[0].payload_name = "H264";
+ video_receive_configs_[0].decoders[0].payload_name = "H264";
break;
}
- CreateStreams();
+ CreateVideoStreams();
RTPHeader header;
EXPECT_TRUE(rtp_header_parser_->Parse(packet, length, &header));
- EXPECT_EQ(kSendSsrcs[0], header.ssrc)
+ EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc)
<< "Packet should have configured SSRC to not be dropped early.";
EXPECT_EQ(payload_type, header.payloadType);
Start();
diff --git a/webrtc/call/rampup_tests.cc b/webrtc/call/rampup_tests.cc
new file mode 100644
index 0000000..81f1e81
--- /dev/null
+++ b/webrtc/call/rampup_tests.cc
@@ -0,0 +1,587 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/call/rampup_tests.h"
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/test/testsupport/perf_test.h"
+
+namespace webrtc {
+namespace {
+
+static const int64_t kPollIntervalMs = 20;
+
+std::vector<uint32_t> GenerateSsrcs(size_t num_streams, uint32_t ssrc_offset) {
+ std::vector<uint32_t> ssrcs;
+ for (size_t i = 0; i != num_streams; ++i)
+ ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i));
+ return ssrcs;
+}
+} // namespace
+
+RampUpTester::RampUpTester(size_t num_video_streams,
+ size_t num_audio_streams,
+ unsigned int start_bitrate_bps,
+ const std::string& extension_type,
+ bool rtx,
+ bool red)
+ : EndToEndTest(test::CallTest::kLongTimeoutMs),
+ event_(false, false),
+ clock_(Clock::GetRealTimeClock()),
+ num_video_streams_(num_video_streams),
+ num_audio_streams_(num_audio_streams),
+ rtx_(rtx),
+ red_(red),
+ send_stream_(nullptr),
+ start_bitrate_bps_(start_bitrate_bps),
+ start_bitrate_verified_(false),
+ expected_bitrate_bps_(0),
+ test_start_ms_(-1),
+ ramp_up_finished_ms_(-1),
+ extension_type_(extension_type),
+ video_ssrcs_(GenerateSsrcs(num_video_streams_, 100)),
+ video_rtx_ssrcs_(GenerateSsrcs(num_video_streams_, 200)),
+ audio_ssrcs_(GenerateSsrcs(num_audio_streams_, 300)),
+ poller_thread_(&BitrateStatsPollingThread,
+ this,
+ "BitrateStatsPollingThread"),
+ sender_call_(nullptr) {
+ EXPECT_LE(num_audio_streams_, 1u);
+ if (rtx_) {
+ for (size_t i = 0; i < video_ssrcs_.size(); ++i)
+ rtx_ssrc_map_[video_rtx_ssrcs_[i]] = video_ssrcs_[i];
+ }
+}
+
+RampUpTester::~RampUpTester() {
+ event_.Set();
+}
+
+Call::Config RampUpTester::GetSenderCallConfig() {
+ Call::Config call_config;
+ if (start_bitrate_bps_ != 0) {
+ call_config.bitrate_config.start_bitrate_bps = start_bitrate_bps_;
+ }
+ call_config.bitrate_config.min_bitrate_bps = 10000;
+ return call_config;
+}
+
+void RampUpTester::OnVideoStreamsCreated(
+ VideoSendStream* send_stream,
+ const std::vector<VideoReceiveStream*>& receive_streams) {
+ send_stream_ = send_stream;
+}
+
+test::PacketTransport* RampUpTester::CreateSendTransport(Call* sender_call) {
+ send_transport_ = new test::PacketTransport(sender_call, this,
+ test::PacketTransport::kSender,
+ forward_transport_config_);
+ return send_transport_;
+}
+
+size_t RampUpTester::GetNumVideoStreams() const {
+ return num_video_streams_;
+}
+
+size_t RampUpTester::GetNumAudioStreams() const {
+ return num_audio_streams_;
+}
+
+void RampUpTester::ModifyVideoConfigs(
+ VideoSendStream::Config* send_config,
+ std::vector<VideoReceiveStream::Config>* receive_configs,
+ VideoEncoderConfig* encoder_config) {
+ send_config->suspend_below_min_bitrate = true;
+
+ if (num_video_streams_ == 1) {
+ encoder_config->streams[0].target_bitrate_bps =
+ encoder_config->streams[0].max_bitrate_bps = 2000000;
+ // For single stream rampup until 1mbps
+ expected_bitrate_bps_ = kSingleStreamTargetBps;
+ } else {
+ // For multi stream rampup until all streams are being sent. That means
+ // enough birate to send all the target streams plus the min bitrate of
+ // the last one.
+ expected_bitrate_bps_ = encoder_config->streams.back().min_bitrate_bps;
+ for (size_t i = 0; i < encoder_config->streams.size() - 1; ++i) {
+ expected_bitrate_bps_ += encoder_config->streams[i].target_bitrate_bps;
+ }
+ }
+
+ send_config->rtp.extensions.clear();
+
+ bool remb;
+ bool transport_cc;
+ if (extension_type_ == RtpExtension::kAbsSendTime) {
+ remb = true;
+ transport_cc = false;
+ send_config->rtp.extensions.push_back(
+ RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
+ } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) {
+ remb = false;
+ transport_cc = true;
+ send_config->rtp.extensions.push_back(RtpExtension(
+ extension_type_.c_str(), kTransportSequenceNumberExtensionId));
+ } else {
+ remb = true;
+ transport_cc = false;
+ send_config->rtp.extensions.push_back(RtpExtension(
+ extension_type_.c_str(), kTransmissionTimeOffsetExtensionId));
+ }
+
+ send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs;
+ send_config->rtp.ssrcs = video_ssrcs_;
+ if (rtx_) {
+ send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
+ send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_;
+ }
+ if (red_) {
+ send_config->rtp.fec.ulpfec_payload_type =
+ test::CallTest::kUlpfecPayloadType;
+ send_config->rtp.fec.red_payload_type = test::CallTest::kRedPayloadType;
+ }
+
+ size_t i = 0;
+ for (VideoReceiveStream::Config& recv_config : *receive_configs) {
+ recv_config.rtp.remb = remb;
+ recv_config.rtp.transport_cc = transport_cc;
+ recv_config.rtp.extensions = send_config->rtp.extensions;
+
+ recv_config.rtp.remote_ssrc = video_ssrcs_[i];
+ recv_config.rtp.nack.rtp_history_ms = send_config->rtp.nack.rtp_history_ms;
+
+ if (red_) {
+ recv_config.rtp.fec.red_payload_type =
+ send_config->rtp.fec.red_payload_type;
+ recv_config.rtp.fec.ulpfec_payload_type =
+ send_config->rtp.fec.ulpfec_payload_type;
+ }
+
+ if (rtx_) {
+ recv_config.rtp.rtx[send_config->encoder_settings.payload_type].ssrc =
+ video_rtx_ssrcs_[i];
+ recv_config.rtp.rtx[send_config->encoder_settings.payload_type]
+ .payload_type = send_config->rtp.rtx.payload_type;
+ }
+ ++i;
+ }
+}
+
+void RampUpTester::ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) {
+ if (num_audio_streams_ == 0)
+ return;
+
+ EXPECT_NE(RtpExtension::kTOffset, extension_type_)
+ << "Audio BWE not supported with toffset.";
+
+ send_config->rtp.ssrc = audio_ssrcs_[0];
+ send_config->rtp.extensions.clear();
+
+ bool transport_cc = false;
+ if (extension_type_ == RtpExtension::kAbsSendTime) {
+ transport_cc = false;
+ send_config->rtp.extensions.push_back(
+ RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
+ } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) {
+ transport_cc = true;
+ send_config->rtp.extensions.push_back(RtpExtension(
+ extension_type_.c_str(), kTransportSequenceNumberExtensionId));
+ }
+
+ for (AudioReceiveStream::Config& recv_config : *receive_configs) {
+ recv_config.combined_audio_video_bwe = true;
+ recv_config.rtp.transport_cc = transport_cc;
+ recv_config.rtp.extensions = send_config->rtp.extensions;
+ recv_config.rtp.remote_ssrc = send_config->rtp.ssrc;
+ }
+}
+
+void RampUpTester::OnCallsCreated(Call* sender_call, Call* receiver_call) {
+ sender_call_ = sender_call;
+}
+
+bool RampUpTester::BitrateStatsPollingThread(void* obj) {
+ return static_cast<RampUpTester*>(obj)->PollStats();
+}
+
+bool RampUpTester::PollStats() {
+ if (sender_call_) {
+ Call::Stats stats = sender_call_->GetStats();
+
+ RTC_DCHECK_GT(expected_bitrate_bps_, 0);
+ if (!start_bitrate_verified_ && start_bitrate_bps_ != 0) {
+ // For tests with an explicitly set start bitrate, verify the first
+ // bitrate estimate is close to the start bitrate and lower than the
+ // test target bitrate. This is to verify a call respects the configured
+ // start bitrate, but due to the BWE implementation we can't guarantee the
+ // first estimate really is as high as the start bitrate.
+ EXPECT_GT(stats.send_bandwidth_bps, 0.9 * start_bitrate_bps_);
+ start_bitrate_verified_ = true;
+ }
+ if (stats.send_bandwidth_bps >= expected_bitrate_bps_) {
+ ramp_up_finished_ms_ = clock_->TimeInMilliseconds();
+ observation_complete_.Set();
+ }
+ }
+
+ return !event_.Wait(kPollIntervalMs);
+}
+
+void RampUpTester::ReportResult(const std::string& measurement,
+ size_t value,
+ const std::string& units) const {
+ webrtc::test::PrintResult(
+ measurement, "",
+ ::testing::UnitTest::GetInstance()->current_test_info()->name(), value,
+ units, false);
+}
+
+void RampUpTester::AccumulateStats(const VideoSendStream::StreamStats& stream,
+ size_t* total_packets_sent,
+ size_t* total_sent,
+ size_t* padding_sent,
+ size_t* media_sent) const {
+ *total_packets_sent += stream.rtp_stats.transmitted.packets +
+ stream.rtp_stats.retransmitted.packets +
+ stream.rtp_stats.fec.packets;
+ *total_sent += stream.rtp_stats.transmitted.TotalBytes() +
+ stream.rtp_stats.retransmitted.TotalBytes() +
+ stream.rtp_stats.fec.TotalBytes();
+ *padding_sent += stream.rtp_stats.transmitted.padding_bytes +
+ stream.rtp_stats.retransmitted.padding_bytes +
+ stream.rtp_stats.fec.padding_bytes;
+ *media_sent += stream.rtp_stats.MediaPayloadBytes();
+}
+
+void RampUpTester::TriggerTestDone() {
+ RTC_DCHECK_GE(test_start_ms_, 0);
+
+ // TODO(holmer): Add audio send stats here too when those APIs are available.
+ VideoSendStream::Stats send_stats = send_stream_->GetStats();
+
+ size_t total_packets_sent = 0;
+ size_t total_sent = 0;
+ size_t padding_sent = 0;
+ size_t media_sent = 0;
+ for (uint32_t ssrc : video_ssrcs_) {
+ AccumulateStats(send_stats.substreams[ssrc], &total_packets_sent,
+ &total_sent, &padding_sent, &media_sent);
+ }
+
+ size_t rtx_total_packets_sent = 0;
+ size_t rtx_total_sent = 0;
+ size_t rtx_padding_sent = 0;
+ size_t rtx_media_sent = 0;
+ for (uint32_t rtx_ssrc : video_rtx_ssrcs_) {
+ AccumulateStats(send_stats.substreams[rtx_ssrc], &rtx_total_packets_sent,
+ &rtx_total_sent, &rtx_padding_sent, &rtx_media_sent);
+ }
+
+ ReportResult("ramp-up-total-packets-sent", total_packets_sent, "packets");
+ ReportResult("ramp-up-total-sent", total_sent, "bytes");
+ ReportResult("ramp-up-media-sent", media_sent, "bytes");
+ ReportResult("ramp-up-padding-sent", padding_sent, "bytes");
+ ReportResult("ramp-up-rtx-total-packets-sent", rtx_total_packets_sent,
+ "packets");
+ ReportResult("ramp-up-rtx-total-sent", rtx_total_sent, "bytes");
+ ReportResult("ramp-up-rtx-media-sent", rtx_media_sent, "bytes");
+ ReportResult("ramp-up-rtx-padding-sent", rtx_padding_sent, "bytes");
+ if (ramp_up_finished_ms_ >= 0) {
+ ReportResult("ramp-up-time", ramp_up_finished_ms_ - test_start_ms_,
+ "milliseconds");
+ }
+ ReportResult("ramp-up-average-network-latency",
+ send_transport_->GetAverageDelayMs(), "milliseconds");
+}
+
+void RampUpTester::PerformTest() {
+ test_start_ms_ = clock_->TimeInMilliseconds();
+ poller_thread_.Start();
+ EXPECT_TRUE(Wait()) << "Timed out while waiting for ramp-up to complete.";
+ TriggerTestDone();
+ poller_thread_.Stop();
+}
+
+RampUpDownUpTester::RampUpDownUpTester(size_t num_video_streams,
+ size_t num_audio_streams,
+ unsigned int start_bitrate_bps,
+ const std::string& extension_type,
+ bool rtx,
+ bool red)
+ : RampUpTester(num_video_streams,
+ num_audio_streams,
+ start_bitrate_bps,
+ extension_type,
+ rtx,
+ red),
+ test_state_(kFirstRampup),
+ state_start_ms_(clock_->TimeInMilliseconds()),
+ interval_start_ms_(clock_->TimeInMilliseconds()),
+ sent_bytes_(0) {
+ forward_transport_config_.link_capacity_kbps = kHighBandwidthLimitBps / 1000;
+}
+
+RampUpDownUpTester::~RampUpDownUpTester() {}
+
+bool RampUpDownUpTester::PollStats() {
+ if (send_stream_) {
+ webrtc::VideoSendStream::Stats stats = send_stream_->GetStats();
+ int transmit_bitrate_bps = 0;
+ for (auto it : stats.substreams) {
+ transmit_bitrate_bps += it.second.total_bitrate_bps;
+ }
+
+ EvolveTestState(transmit_bitrate_bps, stats.suspended);
+ }
+
+ return !event_.Wait(kPollIntervalMs);
+}
+
+Call::Config RampUpDownUpTester::GetReceiverCallConfig() {
+ Call::Config config;
+ config.bitrate_config.min_bitrate_bps = 10000;
+ return config;
+}
+
+std::string RampUpDownUpTester::GetModifierString() const {
+ std::string str("_");
+ if (num_video_streams_ > 0) {
+ std::ostringstream s;
+ s << num_video_streams_;
+ str += s.str();
+ str += "stream";
+ str += (num_video_streams_ > 1 ? "s" : "");
+ str += "_";
+ }
+ if (num_audio_streams_ > 0) {
+ std::ostringstream s;
+ s << num_audio_streams_;
+ str += s.str();
+ str += "stream";
+ str += (num_audio_streams_ > 1 ? "s" : "");
+ str += "_";
+ }
+ str += (rtx_ ? "" : "no");
+ str += "rtx";
+ return str;
+}
+
+void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
+ int64_t now = clock_->TimeInMilliseconds();
+ switch (test_state_) {
+ case kFirstRampup: {
+ EXPECT_FALSE(suspended);
+ if (bitrate_bps > kExpectedHighBitrateBps) {
+ // The first ramp-up has reached the target bitrate. Change the
+ // channel limit, and move to the next test state.
+ forward_transport_config_.link_capacity_kbps =
+ kLowBandwidthLimitBps / 1000;
+ send_transport_->SetConfig(forward_transport_config_);
+ test_state_ = kLowRate;
+ webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
+ "first_rampup", now - state_start_ms_, "ms",
+ false);
+ state_start_ms_ = now;
+ interval_start_ms_ = now;
+ sent_bytes_ = 0;
+ }
+ break;
+ }
+ case kLowRate: {
+ if (bitrate_bps < kExpectedLowBitrateBps && suspended) {
+ // The ramp-down was successful. Change the channel limit back to a
+ // high value, and move to the next test state.
+ forward_transport_config_.link_capacity_kbps =
+ kHighBandwidthLimitBps / 1000;
+ send_transport_->SetConfig(forward_transport_config_);
+ test_state_ = kSecondRampup;
+ webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
+ "rampdown", now - state_start_ms_, "ms",
+ false);
+ state_start_ms_ = now;
+ interval_start_ms_ = now;
+ sent_bytes_ = 0;
+ }
+ break;
+ }
+ case kSecondRampup: {
+ if (bitrate_bps > kExpectedHighBitrateBps && !suspended) {
+ webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
+ "second_rampup", now - state_start_ms_, "ms",
+ false);
+ ReportResult("ramp-up-down-up-average-network-latency",
+ send_transport_->GetAverageDelayMs(), "milliseconds");
+ observation_complete_.Set();
+ }
+ break;
+ }
+ }
+}
+
+class RampUpTest : public test::CallTest {
+ public:
+ RampUpTest() {}
+
+ virtual ~RampUpTest() {
+ EXPECT_EQ(nullptr, video_send_stream_);
+ EXPECT_TRUE(video_receive_streams_.empty());
+ }
+};
+
+TEST_F(RampUpTest, SingleStream) {
+ RampUpTester test(1, 0, 0, RtpExtension::kTOffset, false, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, Simulcast) {
+ RampUpTester test(3, 0, 0, RtpExtension::kTOffset, false, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, SimulcastWithRtx) {
+ RampUpTester test(3, 0, 0, RtpExtension::kTOffset, true, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, SimulcastByRedWithRtx) {
+ RampUpTester test(3, 0, 0, RtpExtension::kTOffset, true, true);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) {
+ RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps, RtpExtension::kTOffset,
+ false, false);
+ RunBaseTest(&test);
+}
+
+// Disabled on Mac due to flakiness, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5407
+#ifndef WEBRTC_MAC
+
+static const uint32_t kStartBitrateBps = 60000;
+
+TEST_F(RampUpTest, UpDownUpOneStream) {
+ RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+ false, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, UpDownUpThreeStreams) {
+ RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+ false, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, UpDownUpOneStreamRtx) {
+ RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+ true, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) {
+ RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+ true, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, UpDownUpOneStreamByRedRtx) {
+ RampUpDownUpTester test(1, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+ true, true);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, UpDownUpThreeStreamsByRedRtx) {
+ RampUpDownUpTester test(3, 0, kStartBitrateBps, RtpExtension::kAbsSendTime,
+ true, true);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, SendSideVideoUpDownUpRtx) {
+ RampUpDownUpTester test(3, 0, kStartBitrateBps,
+ RtpExtension::kTransportSequenceNumber, true, false);
+ RunBaseTest(&test);
+}
+
+// TODO(holmer): Enable when audio bitrates are included in the bitrate
+// allocation.
+TEST_F(RampUpTest, DISABLED_SendSideAudioVideoUpDownUpRtx) {
+ RampUpDownUpTester test(3, 1, kStartBitrateBps,
+ RtpExtension::kTransportSequenceNumber, true, false);
+ RunBaseTest(&test);
+}
+
+#endif
+
+TEST_F(RampUpTest, AbsSendTimeSingleStream) {
+ RampUpTester test(1, 0, 0, RtpExtension::kAbsSendTime, false, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSimulcast) {
+ RampUpTester test(3, 0, 0, RtpExtension::kAbsSendTime, false, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSimulcastWithRtx) {
+ RampUpTester test(3, 0, 0, RtpExtension::kAbsSendTime, true, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSimulcastByRedWithRtx) {
+ RampUpTester test(3, 0, 0, RtpExtension::kAbsSendTime, true, true);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, AbsSendTimeSingleStreamWithHighStartBitrate) {
+ RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
+ RtpExtension::kAbsSendTime, false, false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, TransportSequenceNumberSingleStream) {
+ RampUpTester test(1, 0, 0, RtpExtension::kTransportSequenceNumber, false,
+ false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, TransportSequenceNumberSimulcast) {
+ RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumber, false,
+ false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, TransportSequenceNumberSimulcastWithRtx) {
+ RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumber, true,
+ false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, AudioVideoTransportSequenceNumberSimulcastWithRtx) {
+ RampUpTester test(3, 1, 0, RtpExtension::kTransportSequenceNumber, true,
+ false);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, TransportSequenceNumberSimulcastByRedWithRtx) {
+ RampUpTester test(3, 0, 0, RtpExtension::kTransportSequenceNumber, true,
+ true);
+ RunBaseTest(&test);
+}
+
+TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
+ RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
+ RtpExtension::kTransportSequenceNumber, false, false);
+ RunBaseTest(&test);
+}
+} // namespace webrtc
diff --git a/webrtc/call/rampup_tests.h b/webrtc/call/rampup_tests.h
new file mode 100644
index 0000000..31a0a02
--- /dev/null
+++ b/webrtc/call/rampup_tests.h
@@ -0,0 +1,137 @@
+/*
+ * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_CALL_RAMPUP_TESTS_H_
+#define WEBRTC_CALL_RAMPUP_TESTS_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "webrtc/base/event.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/call.h"
+#include "webrtc/test/call_test.h"
+
+namespace webrtc {
+
+static const int kTransmissionTimeOffsetExtensionId = 6;
+static const int kAbsSendTimeExtensionId = 7;
+static const int kTransportSequenceNumberExtensionId = 8;
+static const unsigned int kSingleStreamTargetBps = 1000000;
+
+class Clock;
+
+class RampUpTester : public test::EndToEndTest {
+ public:
+ RampUpTester(size_t num_video_streams,
+ size_t num_audio_streams,
+ unsigned int start_bitrate_bps,
+ const std::string& extension_type,
+ bool rtx,
+ bool red);
+ ~RampUpTester() override;
+
+ size_t GetNumVideoStreams() const override;
+ size_t GetNumAudioStreams() const override;
+
+ void PerformTest() override;
+
+ protected:
+ virtual bool PollStats();
+
+ void AccumulateStats(const VideoSendStream::StreamStats& stream,
+ size_t* total_packets_sent,
+ size_t* total_sent,
+ size_t* padding_sent,
+ size_t* media_sent) const;
+
+ void ReportResult(const std::string& measurement,
+ size_t value,
+ const std::string& units) const;
+ void TriggerTestDone();
+
+ rtc::Event event_;
+ Clock* const clock_;
+ FakeNetworkPipe::Config forward_transport_config_;
+ const size_t num_video_streams_;
+ const size_t num_audio_streams_;
+ const bool rtx_;
+ const bool red_;
+ VideoSendStream* send_stream_;
+ test::PacketTransport* send_transport_;
+
+ private:
+ typedef std::map<uint32_t, uint32_t> SsrcMap;
+
+ Call::Config GetSenderCallConfig() override;
+ void OnVideoStreamsCreated(
+ VideoSendStream* send_stream,
+ const std::vector<VideoReceiveStream*>& receive_streams) override;
+ test::PacketTransport* CreateSendTransport(Call* sender_call) override;
+ void ModifyVideoConfigs(
+ VideoSendStream::Config* send_config,
+ std::vector<VideoReceiveStream::Config>* receive_configs,
+ VideoEncoderConfig* encoder_config) override;
+ void ModifyAudioConfigs(
+ AudioSendStream::Config* send_config,
+ std::vector<AudioReceiveStream::Config>* receive_configs) override;
+ void OnCallsCreated(Call* sender_call, Call* receiver_call) override;
+
+ static bool BitrateStatsPollingThread(void* obj);
+
+ const int start_bitrate_bps_;
+ bool start_bitrate_verified_;
+ int expected_bitrate_bps_;
+ int64_t test_start_ms_;
+ int64_t ramp_up_finished_ms_;
+
+ const std::string extension_type_;
+ std::vector<uint32_t> video_ssrcs_;
+ std::vector<uint32_t> video_rtx_ssrcs_;
+ std::vector<uint32_t> audio_ssrcs_;
+ SsrcMap rtx_ssrc_map_;
+
+ rtc::PlatformThread poller_thread_;
+ Call* sender_call_;
+};
+
+class RampUpDownUpTester : public RampUpTester {
+ public:
+ RampUpDownUpTester(size_t num_video_streams,
+ size_t num_audio_streams,
+ unsigned int start_bitrate_bps,
+ const std::string& extension_type,
+ bool rtx,
+ bool red);
+ ~RampUpDownUpTester() override;
+
+ protected:
+ bool PollStats() override;
+
+ private:
+ static const int kHighBandwidthLimitBps = 80000;
+ static const int kExpectedHighBitrateBps = 60000;
+ static const int kLowBandwidthLimitBps = 20000;
+ static const int kExpectedLowBitrateBps = 20000;
+ enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
+
+ Call::Config GetReceiverCallConfig() override;
+
+ std::string GetModifierString() const;
+ void EvolveTestState(int bitrate_bps, bool suspended);
+
+ TestStates test_state_;
+ int64_t state_start_ms_;
+ int64_t interval_start_ms_;
+ int sent_bytes_;
+};
+} // namespace webrtc
+#endif // WEBRTC_CALL_RAMPUP_TESTS_H_
diff --git a/webrtc/call/rtc_event_log.cc b/webrtc/call/rtc_event_log.cc
index 550b556..9f592ce 100644
--- a/webrtc/call/rtc_event_log.cc
+++ b/webrtc/call/rtc_event_log.cc
@@ -17,7 +17,9 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/call.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
@@ -54,6 +56,9 @@
const uint8_t* packet,
size_t length) override {}
void LogAudioPlayout(uint32_t ssrc) override {}
+ void LogBwePacketLossEvent(int32_t bitrate,
+ uint8_t fraction_loss,
+ int32_t total_packets) override {}
};
#else // ENABLE_RTC_EVENT_LOG is defined
@@ -78,6 +83,9 @@
const uint8_t* packet,
size_t length) override;
void LogAudioPlayout(uint32_t ssrc) override;
+ void LogBwePacketLossEvent(int32_t bitrate,
+ uint8_t fraction_loss,
+ int32_t total_packets) override;
private:
// Starts logging. This function assumes the file_ has been opened succesfully
@@ -254,8 +262,7 @@
rtc::CritScope lock(&crit_);
rtclog::Event event;
- const int64_t timestamp = clock_->TimeInMicroseconds();
- event.set_timestamp_us(timestamp);
+ event.set_timestamp_us(clock_->TimeInMicroseconds());
event.set_type(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
rtclog::VideoReceiveConfig* receiver_config =
@@ -264,9 +271,6 @@
receiver_config->set_local_ssrc(config.rtp.local_ssrc);
receiver_config->set_rtcp_mode(ConvertRtcpMode(config.rtp.rtcp_mode));
-
- receiver_config->set_receiver_reference_time_report(
- config.rtp.rtcp_xr.receiver_reference_time_report);
receiver_config->set_remb(config.rtp.remb);
for (const auto& kv : config.rtp.rtx) {
@@ -296,8 +300,7 @@
rtc::CritScope lock(&crit_);
rtclog::Event event;
- const int64_t timestamp = clock_->TimeInMicroseconds();
- event.set_timestamp_us(timestamp);
+ event.set_timestamp_us(clock_->TimeInMicroseconds());
event.set_type(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
rtclog::VideoSendConfig* sender_config = event.mutable_video_sender_config();
@@ -318,8 +321,6 @@
}
sender_config->set_rtx_payload_type(config.rtp.rtx.payload_type);
- sender_config->set_c_name(config.rtp.c_name);
-
rtclog::EncoderConfig* encoder = sender_config->mutable_encoder();
encoder->set_name(config.encoder_settings.payload_name);
encoder->set_payload_type(config.encoder_settings.payload_type);
@@ -348,8 +349,7 @@
rtc::CritScope lock(&crit_);
rtclog::Event rtp_event;
- const int64_t timestamp = clock_->TimeInMicroseconds();
- rtp_event.set_timestamp_us(timestamp);
+ rtp_event.set_timestamp_us(clock_->TimeInMicroseconds());
rtp_event.set_type(rtclog::Event::RTP_EVENT);
rtp_event.mutable_rtp_packet()->set_incoming(incoming);
rtp_event.mutable_rtp_packet()->set_type(ConvertMediaType(media_type));
@@ -364,33 +364,89 @@
size_t length) {
rtc::CritScope lock(&crit_);
rtclog::Event rtcp_event;
- const int64_t timestamp = clock_->TimeInMicroseconds();
- rtcp_event.set_timestamp_us(timestamp);
+ rtcp_event.set_timestamp_us(clock_->TimeInMicroseconds());
rtcp_event.set_type(rtclog::Event::RTCP_EVENT);
rtcp_event.mutable_rtcp_packet()->set_incoming(incoming);
rtcp_event.mutable_rtcp_packet()->set_type(ConvertMediaType(media_type));
- rtcp_event.mutable_rtcp_packet()->set_packet_data(packet, length);
+
+ RTCPUtility::RtcpCommonHeader header;
+ const uint8_t* block_begin = packet;
+ const uint8_t* packet_end = packet + length;
+ RTC_DCHECK(length <= IP_PACKET_SIZE);
+ uint8_t buffer[IP_PACKET_SIZE];
+ uint32_t buffer_length = 0;
+ while (block_begin < packet_end) {
+ if (!RtcpParseCommonHeader(block_begin, packet_end - block_begin,
+ &header)) {
+ break; // Incorrect message header.
+ }
+ uint32_t block_size = header.BlockSize();
+ switch (header.packet_type) {
+ case RTCPUtility::PT_SR:
+ FALLTHROUGH();
+ case RTCPUtility::PT_RR:
+ FALLTHROUGH();
+ case RTCPUtility::PT_BYE:
+ FALLTHROUGH();
+ case RTCPUtility::PT_IJ:
+ FALLTHROUGH();
+ case RTCPUtility::PT_RTPFB:
+ FALLTHROUGH();
+ case RTCPUtility::PT_PSFB:
+ FALLTHROUGH();
+ case RTCPUtility::PT_XR:
+ // We log sender reports, receiver reports, bye messages
+ // inter-arrival jitter, third-party loss reports, payload-specific
+ // feedback and extended reports.
+ memcpy(buffer + buffer_length, block_begin, block_size);
+ buffer_length += block_size;
+ break;
+ case RTCPUtility::PT_SDES:
+ FALLTHROUGH();
+ case RTCPUtility::PT_APP:
+ FALLTHROUGH();
+ default:
+ // We don't log sender descriptions, application defined messages
+ // or message blocks of unknown type.
+ break;
+ }
+
+ block_begin += block_size;
+ }
+ rtcp_event.mutable_rtcp_packet()->set_packet_data(buffer, buffer_length);
HandleEvent(&rtcp_event);
}
void RtcEventLogImpl::LogAudioPlayout(uint32_t ssrc) {
rtc::CritScope lock(&crit_);
rtclog::Event event;
- const int64_t timestamp = clock_->TimeInMicroseconds();
- event.set_timestamp_us(timestamp);
+ event.set_timestamp_us(clock_->TimeInMicroseconds());
event.set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT);
auto playout_event = event.mutable_audio_playout_event();
playout_event->set_local_ssrc(ssrc);
HandleEvent(&event);
}
+void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate,
+ uint8_t fraction_loss,
+ int32_t total_packets) {
+ rtc::CritScope lock(&crit_);
+ rtclog::Event event;
+ event.set_timestamp_us(clock_->TimeInMicroseconds());
+ event.set_type(rtclog::Event::BWE_PACKET_LOSS_EVENT);
+ auto bwe_event = event.mutable_bwe_packet_loss_event();
+ bwe_event->set_bitrate(bitrate);
+ bwe_event->set_fraction_loss(fraction_loss);
+ bwe_event->set_total_packets(total_packets);
+ HandleEvent(&event);
+}
+
void RtcEventLogImpl::StopLoggingLocked() {
if (currently_logging_) {
currently_logging_ = false;
// Create a LogEnd event
rtclog::Event event;
- int64_t timestamp = clock_->TimeInMicroseconds();
- event.set_timestamp_us(timestamp);
+ event.set_timestamp_us(clock_->TimeInMicroseconds());
event.set_type(rtclog::Event::LOG_END);
// Store the event and close the file
RTC_DCHECK(file_->Open());
diff --git a/webrtc/call/rtc_event_log.h b/webrtc/call/rtc_event_log.h
index 85d7525..489687a 100644
--- a/webrtc/call/rtc_event_log.h
+++ b/webrtc/call/rtc_event_log.h
@@ -77,6 +77,11 @@
// Logs an audio playout event
virtual void LogAudioPlayout(uint32_t ssrc) = 0;
+ // Logs a bitrate update from the bandwidth estimator based on packet loss.
+ virtual void LogBwePacketLossEvent(int32_t bitrate,
+ uint8_t fraction_loss,
+ int32_t total_packets) = 0;
+
// Reads an RtcEventLog file and returns true when reading was successful.
// The result is stored in the given EventStream object.
static bool ParseRtcEventLog(const std::string& file_name,
diff --git a/webrtc/call/rtc_event_log.proto b/webrtc/call/rtc_event_log.proto
index 6bdea7b..b14306e 100644
--- a/webrtc/call/rtc_event_log.proto
+++ b/webrtc/call/rtc_event_log.proto
@@ -34,10 +34,12 @@
RTP_EVENT = 3;
RTCP_EVENT = 4;
AUDIO_PLAYOUT_EVENT = 5;
- VIDEO_RECEIVER_CONFIG_EVENT = 6;
- VIDEO_SENDER_CONFIG_EVENT = 7;
- AUDIO_RECEIVER_CONFIG_EVENT = 8;
- AUDIO_SENDER_CONFIG_EVENT = 9;
+ BWE_PACKET_LOSS_EVENT = 6;
+ BWE_PACKET_DELAY_EVENT = 7;
+ VIDEO_RECEIVER_CONFIG_EVENT = 8;
+ VIDEO_SENDER_CONFIG_EVENT = 9;
+ AUDIO_RECEIVER_CONFIG_EVENT = 10;
+ AUDIO_SENDER_CONFIG_EVENT = 11;
}
// required - Indicates the type of this event
@@ -52,17 +54,20 @@
// optional - but required if type == AUDIO_PLAYOUT_EVENT
optional AudioPlayoutEvent audio_playout_event = 5;
+ // optional - but required if type == BWE_PACKET_LOSS_EVENT
+ optional BwePacketLossEvent bwe_packet_loss_event = 6;
+
// optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT
- optional VideoReceiveConfig video_receiver_config = 6;
+ optional VideoReceiveConfig video_receiver_config = 8;
// optional - but required if type == VIDEO_SENDER_CONFIG_EVENT
- optional VideoSendConfig video_sender_config = 7;
+ optional VideoSendConfig video_sender_config = 9;
// optional - but required if type == AUDIO_RECEIVER_CONFIG_EVENT
- optional AudioReceiveConfig audio_receiver_config = 8;
+ optional AudioReceiveConfig audio_receiver_config = 10;
// optional - but required if type == AUDIO_SENDER_CONFIG_EVENT
- optional AudioSendConfig audio_sender_config = 9;
+ optional AudioSendConfig audio_sender_config = 11;
}
@@ -99,6 +104,19 @@
optional uint32 local_ssrc = 2;
}
+message BwePacketLossEvent {
+ // required - Bandwidth estimate (in bps) after the update.
+ optional int32 bitrate = 1;
+
+ // required - Fraction of lost packets since last receiver report
+ // computed as floor( 256 * (#lost_packets / #total_packets) ).
+ // The possible values range from 0 to 255.
+ optional uint32 fraction_loss = 2;
+
+ // TODO(terelius): Is this really needed? Remove or make optional?
+ // required - Total number of packets that the BWE update is based on.
+ optional int32 total_packets = 3;
+}
// TODO(terelius): Video and audio streams could in principle share SSRC,
// so identifying a stream based only on SSRC might not work.
@@ -119,20 +137,17 @@
// required - RTCP mode to use.
optional RtcpMode rtcp_mode = 3;
- // required - Extended RTCP settings.
- optional bool receiver_reference_time_report = 4;
-
// required - Receiver estimated maximum bandwidth.
- optional bool remb = 5;
+ optional bool remb = 4;
// Map from video RTP payload type -> RTX config.
- repeated RtxMap rtx_map = 6;
+ repeated RtxMap rtx_map = 5;
// RTP header extensions used for the received stream.
- repeated RtpHeaderExtension header_extensions = 7;
+ repeated RtpHeaderExtension header_extensions = 6;
// List of decoders associated with the stream.
- repeated DecoderConfig decoders = 8;
+ repeated DecoderConfig decoders = 7;
}
@@ -142,7 +157,7 @@
optional string name = 1;
// required
- optional sint32 payload_type = 2;
+ optional int32 payload_type = 2;
}
@@ -152,7 +167,7 @@
optional string name = 1;
// required
- optional sint32 id = 2;
+ optional int32 id = 2;
}
@@ -163,13 +178,13 @@
optional uint32 rtx_ssrc = 1;
// required - Payload type to use for the RTX stream.
- optional sint32 rtx_payload_type = 2;
+ optional int32 rtx_payload_type = 2;
}
message RtxMap {
// required
- optional sint32 payload_type = 1;
+ optional int32 payload_type = 1;
// required
optional RtxConfig config = 2;
@@ -189,13 +204,10 @@
repeated uint32 rtx_ssrcs = 3;
// required if rtx_ssrcs is used - Payload type for retransmitted packets.
- optional sint32 rtx_payload_type = 4;
-
- // required - Canonical end-point identifier.
- optional string c_name = 5;
+ optional int32 rtx_payload_type = 4;
// required - Encoder associated with the stream.
- optional EncoderConfig encoder = 6;
+ optional EncoderConfig encoder = 5;
}
@@ -205,7 +217,7 @@
optional string name = 1;
// required
- optional sint32 payload_type = 2;
+ optional int32 payload_type = 2;
}
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc
index a4fdd13..f590f66 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -10,22 +10,23 @@
#ifdef ENABLE_RTC_EVENT_LOG
-#include <stdio.h>
#include <string>
+#include <utility>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/random.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread.h"
#include "webrtc/call.h"
#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/test_suite.h"
#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
// Files generated at build-time by the protobuf compiler.
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
@@ -138,9 +139,6 @@
else
EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
receiver_config.rtcp_mode());
- ASSERT_TRUE(receiver_config.has_receiver_reference_time_report());
- EXPECT_EQ(config.rtp.rtcp_xr.receiver_reference_time_report,
- receiver_config.receiver_reference_time_report());
ASSERT_TRUE(receiver_config.has_remb());
EXPECT_EQ(config.rtp.remb, receiver_config.remb());
// Check RTX map.
@@ -214,9 +212,6 @@
ASSERT_TRUE(sender_config.has_rtx_payload_type());
EXPECT_EQ(config.rtp.rtx.payload_type, sender_config.rtx_payload_type());
}
- // Check CNAME.
- ASSERT_TRUE(sender_config.has_c_name());
- EXPECT_EQ(config.rtp.c_name, sender_config.c_name());
// Check encoder.
ASSERT_TRUE(sender_config.has_encoder());
ASSERT_TRUE(sender_config.encoder().has_name());
@@ -230,7 +225,7 @@
void VerifyRtpEvent(const rtclog::Event& event,
bool incoming,
MediaType media_type,
- uint8_t* header,
+ const uint8_t* header,
size_t header_size,
size_t total_size) {
ASSERT_TRUE(IsValidBasicEvent(event));
@@ -252,7 +247,7 @@
void VerifyRtcpEvent(const rtclog::Event& event,
bool incoming,
MediaType media_type,
- uint8_t* packet,
+ const uint8_t* packet,
size_t total_size) {
ASSERT_TRUE(IsValidBasicEvent(event));
ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type());
@@ -276,6 +271,21 @@
EXPECT_EQ(ssrc, playout_event.local_ssrc());
}
+void VerifyBweLossEvent(const rtclog::Event& event,
+ int32_t bitrate,
+ uint8_t fraction_loss,
+ int32_t total_packets) {
+ ASSERT_TRUE(IsValidBasicEvent(event));
+ ASSERT_EQ(rtclog::Event::BWE_PACKET_LOSS_EVENT, event.type());
+ const rtclog::BwePacketLossEvent& bwe_event = event.bwe_packet_loss_event();
+ ASSERT_TRUE(bwe_event.has_bitrate());
+ EXPECT_EQ(bitrate, bwe_event.bitrate());
+ ASSERT_TRUE(bwe_event.has_fraction_loss());
+ EXPECT_EQ(fraction_loss, bwe_event.fraction_loss());
+ ASSERT_TRUE(bwe_event.has_total_packets());
+ EXPECT_EQ(total_packets, bwe_event.total_packets());
+}
+
void VerifyLogStartEvent(const rtclog::Event& event) {
ASSERT_TRUE(IsValidBasicEvent(event));
EXPECT_EQ(rtclog::Event::LOG_START, event.type());
@@ -289,7 +299,8 @@
size_t GenerateRtpPacket(uint32_t extensions_bitvector,
uint32_t csrcs_count,
uint8_t* packet,
- size_t packet_size) {
+ size_t packet_size,
+ Random* prng) {
RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
Clock* clock = Clock::GetRealTimeClock();
@@ -306,12 +317,12 @@
std::vector<uint32_t> csrcs;
for (unsigned i = 0; i < csrcs_count; i++) {
- csrcs.push_back(rand());
+ csrcs.push_back(prng->Rand<uint32_t>());
}
rtp_sender.SetCsrcs(csrcs);
- rtp_sender.SetSSRC(rand());
- rtp_sender.SetStartTimestamp(rand(), true);
- rtp_sender.SetSequenceNumber(rand());
+ rtp_sender.SetSSRC(prng->Rand<uint32_t>());
+ rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true);
+ rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>());
for (unsigned i = 0; i < kNumExtensions; i++) {
if (extensions_bitvector & (1u << i)) {
@@ -319,76 +330,84 @@
}
}
- int8_t payload_type = rand() % 128;
- bool marker_bit = (rand() % 2 == 1);
- uint32_t capture_timestamp = rand();
- int64_t capture_time_ms = rand();
- bool timestamp_provided = (rand() % 2 == 1);
- bool inc_sequence_number = (rand() % 2 == 1);
+ int8_t payload_type = prng->Rand(0, 127);
+ bool marker_bit = prng->Rand<bool>();
+ uint32_t capture_timestamp = prng->Rand<uint32_t>();
+ int64_t capture_time_ms = prng->Rand<uint32_t>();
+ bool timestamp_provided = prng->Rand<bool>();
+ bool inc_sequence_number = prng->Rand<bool>();
size_t header_size = rtp_sender.BuildRTPheader(
packet, payload_type, marker_bit, capture_timestamp, capture_time_ms,
timestamp_provided, inc_sequence_number);
for (size_t i = header_size; i < packet_size; i++) {
- packet[i] = rand();
+ packet[i] = prng->Rand<uint8_t>();
}
return header_size;
}
-void GenerateRtcpPacket(uint8_t* packet, size_t packet_size) {
- for (size_t i = 0; i < packet_size; i++) {
- packet[i] = rand();
- }
+rtc::scoped_ptr<rtcp::RawPacket> GenerateRtcpPacket(Random* prng) {
+ rtcp::ReportBlock report_block;
+ report_block.To(prng->Rand<uint32_t>()); // Remote SSRC.
+ report_block.WithFractionLost(prng->Rand(50));
+
+ rtcp::SenderReport sender_report;
+ sender_report.From(prng->Rand<uint32_t>()); // Sender SSRC.
+ sender_report.WithNtpSec(prng->Rand<uint32_t>());
+ sender_report.WithNtpFrac(prng->Rand<uint32_t>());
+ sender_report.WithPacketCount(prng->Rand<uint32_t>());
+ sender_report.WithReportBlock(report_block);
+
+ return sender_report.Build();
}
void GenerateVideoReceiveConfig(uint32_t extensions_bitvector,
- VideoReceiveStream::Config* config) {
+ VideoReceiveStream::Config* config,
+ Random* prng) {
// Create a map from a payload type to an encoder name.
VideoReceiveStream::Decoder decoder;
- decoder.payload_type = rand();
- decoder.payload_name = (rand() % 2 ? "VP8" : "H264");
+ decoder.payload_type = prng->Rand(0, 127);
+ decoder.payload_name = (prng->Rand<bool>() ? "VP8" : "H264");
config->decoders.push_back(decoder);
// Add SSRCs for the stream.
- config->rtp.remote_ssrc = rand();
- config->rtp.local_ssrc = rand();
+ config->rtp.remote_ssrc = prng->Rand<uint32_t>();
+ config->rtp.local_ssrc = prng->Rand<uint32_t>();
// Add extensions and settings for RTCP.
config->rtp.rtcp_mode =
- rand() % 2 ? RtcpMode::kCompound : RtcpMode::kReducedSize;
- config->rtp.rtcp_xr.receiver_reference_time_report = (rand() % 2 == 1);
- config->rtp.remb = (rand() % 2 == 1);
+ prng->Rand<bool>() ? RtcpMode::kCompound : RtcpMode::kReducedSize;
+ config->rtp.remb = prng->Rand<bool>();
// Add a map from a payload type to a new ssrc and a new payload type for RTX.
VideoReceiveStream::Config::Rtp::Rtx rtx_pair;
- rtx_pair.ssrc = rand();
- rtx_pair.payload_type = rand();
- config->rtp.rtx.insert(std::make_pair(rand(), rtx_pair));
+ rtx_pair.ssrc = prng->Rand<uint32_t>();
+ rtx_pair.payload_type = prng->Rand(0, 127);
+ config->rtp.rtx.insert(std::make_pair(prng->Rand(0, 127), rtx_pair));
// Add header extensions.
for (unsigned i = 0; i < kNumExtensions; i++) {
if (extensions_bitvector & (1u << i)) {
config->rtp.extensions.push_back(
- RtpExtension(kExtensionNames[i], rand()));
+ RtpExtension(kExtensionNames[i], prng->Rand<int>()));
}
}
}
void GenerateVideoSendConfig(uint32_t extensions_bitvector,
- VideoSendStream::Config* config) {
+ VideoSendStream::Config* config,
+ Random* prng) {
// Create a map from a payload type to an encoder name.
- config->encoder_settings.payload_type = rand();
- config->encoder_settings.payload_name = (rand() % 2 ? "VP8" : "H264");
+ config->encoder_settings.payload_type = prng->Rand(0, 127);
+ config->encoder_settings.payload_name = (prng->Rand<bool>() ? "VP8" : "H264");
// Add SSRCs for the stream.
- config->rtp.ssrcs.push_back(rand());
+ config->rtp.ssrcs.push_back(prng->Rand<uint32_t>());
// Add a map from a payload type to new ssrcs and a new payload type for RTX.
- config->rtp.rtx.ssrcs.push_back(rand());
- config->rtp.rtx.payload_type = rand();
- // Add a CNAME.
- config->rtp.c_name = "some.user@some.host";
+ config->rtp.rtx.ssrcs.push_back(prng->Rand<uint32_t>());
+ config->rtp.rtx.payload_type = prng->Rand(0, 127);
// Add header extensions.
for (unsigned i = 0; i < kNumExtensions; i++) {
if (extensions_bitvector & (1u << i)) {
config->rtp.extensions.push_back(
- RtpExtension(kExtensionNames[i], rand()));
+ RtpExtension(kExtensionNames[i], prng->Rand<int>()));
}
}
}
@@ -398,42 +417,49 @@
void LogSessionAndReadBack(size_t rtp_count,
size_t rtcp_count,
size_t playout_count,
+ size_t bwe_loss_count,
uint32_t extensions_bitvector,
uint32_t csrcs_count,
unsigned int random_seed) {
ASSERT_LE(rtcp_count, rtp_count);
ASSERT_LE(playout_count, rtp_count);
+ ASSERT_LE(bwe_loss_count, rtp_count);
std::vector<rtc::Buffer> rtp_packets;
- std::vector<rtc::Buffer> rtcp_packets;
+ std::vector<rtc::scoped_ptr<rtcp::RawPacket> > rtcp_packets;
std::vector<size_t> rtp_header_sizes;
std::vector<uint32_t> playout_ssrcs;
+ std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates;
VideoReceiveStream::Config receiver_config(nullptr);
VideoSendStream::Config sender_config(nullptr);
- srand(random_seed);
+ Random prng(random_seed);
// Create rtp_count RTP packets containing random data.
for (size_t i = 0; i < rtp_count; i++) {
- size_t packet_size = 1000 + rand() % 64;
+ size_t packet_size = prng.Rand(1000, 1100);
rtp_packets.push_back(rtc::Buffer(packet_size));
- size_t header_size = GenerateRtpPacket(extensions_bitvector, csrcs_count,
- rtp_packets[i].data(), packet_size);
+ size_t header_size =
+ GenerateRtpPacket(extensions_bitvector, csrcs_count,
+ rtp_packets[i].data(), packet_size, &prng);
rtp_header_sizes.push_back(header_size);
}
// Create rtcp_count RTCP packets containing random data.
for (size_t i = 0; i < rtcp_count; i++) {
- size_t packet_size = 1000 + rand() % 64;
- rtcp_packets.push_back(rtc::Buffer(packet_size));
- GenerateRtcpPacket(rtcp_packets[i].data(), packet_size);
+ rtcp_packets.push_back(GenerateRtcpPacket(&prng));
}
// Create playout_count random SSRCs to use when logging AudioPlayout events.
for (size_t i = 0; i < playout_count; i++) {
- playout_ssrcs.push_back(static_cast<uint32_t>(rand()));
+ playout_ssrcs.push_back(prng.Rand<uint32_t>());
+ }
+ // Create bwe_loss_count random bitrate updates for BwePacketLoss.
+ for (size_t i = 0; i < bwe_loss_count; i++) {
+ bwe_loss_updates.push_back(
+ std::make_pair(prng.Rand<int32_t>(), prng.Rand<uint8_t>()));
}
// Create configurations for the video streams.
- GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config);
- GenerateVideoSendConfig(extensions_bitvector, &sender_config);
+ GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng);
+ GenerateVideoSendConfig(extensions_bitvector, &sender_config, &prng);
const int config_count = 2;
// Find the name of the current test, in order to use it as a temporary
@@ -448,7 +474,9 @@
rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
log_dumper->LogVideoReceiveStreamConfig(receiver_config);
log_dumper->LogVideoSendStreamConfig(sender_config);
- size_t rtcp_index = 1, playout_index = 1;
+ size_t rtcp_index = 1;
+ size_t playout_index = 1;
+ size_t bwe_loss_index = 1;
for (size_t i = 1; i <= rtp_count; i++) {
log_dumper->LogRtpHeader(
(i % 2 == 0), // Every second packet is incoming.
@@ -458,14 +486,20 @@
log_dumper->LogRtcpPacket(
rtcp_index % 2 == 0, // Every second packet is incoming
rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
- rtcp_packets[rtcp_index - 1].data(),
- rtcp_packets[rtcp_index - 1].size());
+ rtcp_packets[rtcp_index - 1]->Buffer(),
+ rtcp_packets[rtcp_index - 1]->Length());
rtcp_index++;
}
if (i * playout_count >= playout_index * rtp_count) {
log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]);
playout_index++;
}
+ if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
+ log_dumper->LogBwePacketLossEvent(
+ bwe_loss_updates[bwe_loss_index - 1].first,
+ bwe_loss_updates[bwe_loss_index - 1].second, i);
+ bwe_loss_index++;
+ }
if (i == rtp_count / 2) {
log_dumper->StartLogging(temp_filename, 10000000);
}
@@ -480,12 +514,15 @@
// Verify that what we read back from the event log is the same as
// what we wrote down. For RTCP we log the full packets, but for
// RTP we should only log the header.
- const int event_count =
- config_count + playout_count + rtcp_count + rtp_count + 1;
+ const int event_count = config_count + playout_count + bwe_loss_count +
+ rtcp_count + rtp_count + 1;
EXPECT_EQ(event_count, parsed_stream.stream_size());
VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config);
VerifySendStreamConfig(parsed_stream.stream(1), sender_config);
- size_t event_index = config_count, rtcp_index = 1, playout_index = 1;
+ size_t event_index = config_count;
+ size_t rtcp_index = 1;
+ size_t playout_index = 1;
+ size_t bwe_loss_index = 1;
for (size_t i = 1; i <= rtp_count; i++) {
VerifyRtpEvent(parsed_stream.stream(event_index),
(i % 2 == 0), // Every second packet is incoming.
@@ -497,8 +534,8 @@
VerifyRtcpEvent(parsed_stream.stream(event_index),
rtcp_index % 2 == 0, // Every second packet is incoming.
rtcp_index % 3 == 0 ? MediaType::AUDIO : MediaType::VIDEO,
- rtcp_packets[rtcp_index - 1].data(),
- rtcp_packets[rtcp_index - 1].size());
+ rtcp_packets[rtcp_index - 1]->Buffer(),
+ rtcp_packets[rtcp_index - 1]->Length());
event_index++;
rtcp_index++;
}
@@ -508,6 +545,13 @@
event_index++;
playout_index++;
}
+ if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
+ VerifyBweLossEvent(parsed_stream.stream(event_index),
+ bwe_loss_updates[bwe_loss_index - 1].first,
+ bwe_loss_updates[bwe_loss_index - 1].second, i);
+ event_index++;
+ bwe_loss_index++;
+ }
if (i == rtp_count / 2) {
VerifyLogStartEvent(parsed_stream.stream(event_index));
event_index++;
@@ -519,10 +563,11 @@
}
TEST(RtcEventLogTest, LogSessionAndReadBack) {
- // Log 5 RTP, 2 RTCP, and 0 playout events with no header extensions or CSRCS.
- LogSessionAndReadBack(5, 2, 0, 0, 0, 321);
+ // Log 5 RTP, 2 RTCP, 0 playout events and 0 BWE events
+ // with no header extensions or CSRCS.
+ LogSessionAndReadBack(5, 2, 0, 0, 0, 0, 321);
- // Enable AbsSendTime and TransportSequenceNumbers
+ // Enable AbsSendTime and TransportSequenceNumbers.
uint32_t extensions = 0;
for (uint32_t i = 0; i < kNumExtensions; i++) {
if (kExtensionTypes[i] == RTPExtensionType::kRtpExtensionAbsoluteSendTime ||
@@ -531,20 +576,21 @@
extensions |= 1u << i;
}
}
- LogSessionAndReadBack(8, 2, 0, extensions, 0, 3141592653u);
+ LogSessionAndReadBack(8, 2, 0, 0, extensions, 0, 3141592653u);
- extensions = (1u << kNumExtensions) - 1; // Enable all header extensions
- LogSessionAndReadBack(9, 2, 3, extensions, 2, 2718281828u);
+ extensions = (1u << kNumExtensions) - 1; // Enable all header extensions.
+ LogSessionAndReadBack(9, 2, 3, 2, extensions, 2, 2718281828u);
// Try all combinations of header extensions and up to 2 CSRCS.
for (extensions = 0; extensions < (1u << kNumExtensions); extensions++) {
for (uint32_t csrcs_count = 0; csrcs_count < 3; csrcs_count++) {
LogSessionAndReadBack(5 + extensions, // Number of RTP packets.
2 + csrcs_count, // Number of RTCP packets.
- 3 + csrcs_count, // Number of playout events
- extensions, // Bit vector choosing extensions
- csrcs_count, // Number of contributing sources
- rand());
+ 3 + csrcs_count, // Number of playout events.
+ 1 + csrcs_count, // Number of BWE loss events.
+ extensions, // Bit vector choosing extensions.
+ csrcs_count, // Number of contributing sources.
+ extensions * 3 + csrcs_count + 1); // Random seed.
}
}
}
@@ -556,35 +602,32 @@
unsigned int random_seed) {
rtc::Buffer old_rtp_packet;
rtc::Buffer recent_rtp_packet;
- rtc::Buffer old_rtcp_packet;
- rtc::Buffer recent_rtcp_packet;
+ rtc::scoped_ptr<rtcp::RawPacket> old_rtcp_packet;
+ rtc::scoped_ptr<rtcp::RawPacket> recent_rtcp_packet;
VideoReceiveStream::Config receiver_config(nullptr);
VideoSendStream::Config sender_config(nullptr);
- srand(random_seed);
+ Random prng(random_seed);
// Create two RTP packets containing random data.
- size_t packet_size = 1000 + rand() % 64;
+ size_t packet_size = prng.Rand(1000, 1100);
old_rtp_packet.SetSize(packet_size);
GenerateRtpPacket(extensions_bitvector, csrcs_count, old_rtp_packet.data(),
- packet_size);
- packet_size = 1000 + rand() % 64;
+ packet_size, &prng);
+ packet_size = prng.Rand(1000, 1100);
recent_rtp_packet.SetSize(packet_size);
- size_t recent_header_size = GenerateRtpPacket(
- extensions_bitvector, csrcs_count, recent_rtp_packet.data(), packet_size);
+ size_t recent_header_size =
+ GenerateRtpPacket(extensions_bitvector, csrcs_count,
+ recent_rtp_packet.data(), packet_size, &prng);
// Create two RTCP packets containing random data.
- packet_size = 1000 + rand() % 64;
- old_rtcp_packet.SetSize(packet_size);
- GenerateRtcpPacket(old_rtcp_packet.data(), packet_size);
- packet_size = 1000 + rand() % 64;
- recent_rtcp_packet.SetSize(packet_size);
- GenerateRtcpPacket(recent_rtcp_packet.data(), packet_size);
+ old_rtcp_packet = GenerateRtcpPacket(&prng);
+ recent_rtcp_packet = GenerateRtcpPacket(&prng);
// Create configurations for the video streams.
- GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config);
- GenerateVideoSendConfig(extensions_bitvector, &sender_config);
+ GenerateVideoReceiveConfig(extensions_bitvector, &receiver_config, &prng);
+ GenerateVideoSendConfig(extensions_bitvector, &sender_config, &prng);
// Find the name of the current test, in order to use it as a temporary
// filename.
@@ -601,16 +644,16 @@
log_dumper->LogVideoSendStreamConfig(sender_config);
log_dumper->LogRtpHeader(false, MediaType::AUDIO, old_rtp_packet.data(),
old_rtp_packet.size());
- log_dumper->LogRtcpPacket(true, MediaType::AUDIO, old_rtcp_packet.data(),
- old_rtcp_packet.size());
+ log_dumper->LogRtcpPacket(true, MediaType::AUDIO, old_rtcp_packet->Buffer(),
+ old_rtcp_packet->Length());
// Sleep 55 ms to let old events be removed from the queue.
rtc::Thread::SleepMs(55);
log_dumper->StartLogging(temp_filename, 10000000);
log_dumper->LogRtpHeader(true, MediaType::VIDEO, recent_rtp_packet.data(),
recent_rtp_packet.size());
log_dumper->LogRtcpPacket(false, MediaType::VIDEO,
- recent_rtcp_packet.data(),
- recent_rtcp_packet.size());
+ recent_rtcp_packet->Buffer(),
+ recent_rtcp_packet->Length());
}
// Read the generated file from disk.
@@ -628,7 +671,7 @@
recent_rtp_packet.data(), recent_header_size,
recent_rtp_packet.size());
VerifyRtcpEvent(parsed_stream.stream(4), false, MediaType::VIDEO,
- recent_rtcp_packet.data(), recent_rtcp_packet.size());
+ recent_rtcp_packet->Buffer(), recent_rtcp_packet->Length());
// Clean up temporary file - can be pretty slow.
remove(temp_filename.c_str());
diff --git a/webrtc/call/webrtc_call.gypi b/webrtc/call/webrtc_call.gypi
index fd70ae8..0c3efff 100644
--- a/webrtc/call/webrtc_call.gypi
+++ b/webrtc/call/webrtc_call.gypi
@@ -14,6 +14,7 @@
'<(webrtc_root)/webrtc.gyp:rtc_event_log',
],
'webrtc_call_sources': [
+ 'call/bitrate_allocator.cc',
'call/call.cc',
'call/congestion_controller.cc',
'call/transport_adapter.cc',
diff --git a/webrtc/codereview.settings b/webrtc/codereview.settings
index 97bee14..c441cc6 100644
--- a/webrtc/codereview.settings
+++ b/webrtc/codereview.settings
@@ -1,4 +1,5 @@
-Creating CLs from this location is not supported!
-Please create a full WebRTC checkout using 'fetch webrtc'
-or by cloning https://chromium.googlesource.com/external/webrtc
+Creating CLs from this location is not supported! Please make sure the current
+working directory is the parent directory of this directory.
+If you're working with a Chromium checkout, you'll have to create a full WebRTC
+checkout and upload a CL from that. See http://www.webrtc.org for instructions.
diff --git a/webrtc/common.h b/webrtc/common.h
index dda045e..d705d4b 100644
--- a/webrtc/common.h
+++ b/webrtc/common.h
@@ -17,6 +17,23 @@
namespace webrtc {
+// Only add new values to the end of the enumeration and never remove (only
+// deprecate) to maintain binary compatibility.
+enum class ConfigOptionID {
+ kMyExperimentForTest,
+ kAlgo1CostFunctionForTest,
+ kTemporalLayersFactory,
+ kNetEqCapacityConfig,
+ kNetEqFastAccelerate,
+ kVoicePacing,
+ kExtendedFilter,
+ kDelayAgnostic,
+ kExperimentalAgc,
+ kExperimentalNs,
+ kBeamforming,
+ kIntelligibility
+};
+
// Class Config is designed to ease passing a set of options across webrtc code.
// Options are identified by typename in order to avoid incorrect casts.
//
@@ -61,8 +78,6 @@
}
private:
- typedef void* OptionIdentifier;
-
struct BaseOption {
virtual ~BaseOption() {}
};
@@ -76,11 +91,9 @@
T* value;
};
- // Own implementation of rtti-subset to avoid depending on rtti and its costs.
template<typename T>
- static OptionIdentifier identifier() {
- static char id_placeholder;
- return &id_placeholder;
+ static ConfigOptionID identifier() {
+ return T::identifier;
}
// Used to instantiate a default constructed object that doesn't needs to be
@@ -92,7 +105,7 @@
return def;
}
- typedef std::map<OptionIdentifier, BaseOption*> OptionMap;
+ typedef std::map<ConfigOptionID, BaseOption*> OptionMap;
OptionMap options_;
// RTC_DISALLOW_COPY_AND_ASSIGN
diff --git a/webrtc/common_audio/BUILD.gn b/webrtc/common_audio/BUILD.gn
index b01b318..b4ec1d7 100644
--- a/webrtc/common_audio/BUILD.gn
+++ b/webrtc/common_audio/BUILD.gn
@@ -87,6 +87,7 @@
"signal_processing/vector_scaling_operations.c",
"sparse_fir_filter.cc",
"sparse_fir_filter.h",
+ "swap_queue.h",
"vad/include/vad.h",
"vad/include/webrtc_vad.h",
"vad/vad.cc",
diff --git a/webrtc/common_audio/OWNERS b/webrtc/common_audio/OWNERS
index 20f6400..208a7c5 100644
--- a/webrtc/common_audio/OWNERS
+++ b/webrtc/common_audio/OWNERS
@@ -1,4 +1,3 @@
-andrew@webrtc.org
henrik.lundin@webrtc.org
jan.skoglund@webrtc.org
kwiberg@webrtc.org
diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc
index f1709ae..9ebfabc 100644
--- a/webrtc/common_audio/audio_converter.cc
+++ b/webrtc/common_audio/audio_converter.cc
@@ -11,6 +11,7 @@
#include "webrtc/common_audio/audio_converter.h"
#include <cstring>
+#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
@@ -24,7 +25,7 @@
class CopyConverter : public AudioConverter {
public:
- CopyConverter(int src_channels, size_t src_frames, int dst_channels,
+ CopyConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
~CopyConverter() override {};
@@ -33,7 +34,7 @@
size_t dst_capacity) override {
CheckSizes(src_size, dst_capacity);
if (src != dst) {
- for (int i = 0; i < src_channels(); ++i)
+ for (size_t i = 0; i < src_channels(); ++i)
std::memcpy(dst[i], src[i], dst_frames() * sizeof(*dst[i]));
}
}
@@ -41,7 +42,7 @@
class UpmixConverter : public AudioConverter {
public:
- UpmixConverter(int src_channels, size_t src_frames, int dst_channels,
+ UpmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
~UpmixConverter() override {};
@@ -51,7 +52,7 @@
CheckSizes(src_size, dst_capacity);
for (size_t i = 0; i < dst_frames(); ++i) {
const float value = src[0][i];
- for (int j = 0; j < dst_channels(); ++j)
+ for (size_t j = 0; j < dst_channels(); ++j)
dst[j][i] = value;
}
}
@@ -59,7 +60,7 @@
class DownmixConverter : public AudioConverter {
public:
- DownmixConverter(int src_channels, size_t src_frames, int dst_channels,
+ DownmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
}
@@ -71,7 +72,7 @@
float* dst_mono = dst[0];
for (size_t i = 0; i < src_frames(); ++i) {
float sum = 0;
- for (int j = 0; j < src_channels(); ++j)
+ for (size_t j = 0; j < src_channels(); ++j)
sum += src[j][i];
dst_mono[i] = sum / src_channels();
}
@@ -80,11 +81,11 @@
class ResampleConverter : public AudioConverter {
public:
- ResampleConverter(int src_channels, size_t src_frames, int dst_channels,
+ ResampleConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
size_t dst_frames)
: AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
resamplers_.reserve(src_channels);
- for (int i = 0; i < src_channels; ++i)
+ for (size_t i = 0; i < src_channels; ++i)
resamplers_.push_back(new PushSincResampler(src_frames, dst_frames));
}
~ResampleConverter() override {};
@@ -105,7 +106,7 @@
class CompositionConverter : public AudioConverter {
public:
CompositionConverter(ScopedVector<AudioConverter> converters)
- : converters_(converters.Pass()) {
+ : converters_(std::move(converters)) {
RTC_CHECK_GE(converters_.size(), 2u);
// We need an intermediate buffer after every converter.
for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
@@ -135,9 +136,9 @@
ScopedVector<ChannelBuffer<float>> buffers_;
};
-rtc::scoped_ptr<AudioConverter> AudioConverter::Create(int src_channels,
+rtc::scoped_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
size_t src_frames,
- int dst_channels,
+ size_t dst_channels,
size_t dst_frames) {
rtc::scoped_ptr<AudioConverter> sp;
if (src_channels > dst_channels) {
@@ -147,7 +148,7 @@
dst_channels, src_frames));
converters.push_back(new ResampleConverter(dst_channels, src_frames,
dst_channels, dst_frames));
- sp.reset(new CompositionConverter(converters.Pass()));
+ sp.reset(new CompositionConverter(std::move(converters)));
} else {
sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
dst_frames));
@@ -159,7 +160,7 @@
src_channels, dst_frames));
converters.push_back(new UpmixConverter(src_channels, dst_frames,
dst_channels, dst_frames));
- sp.reset(new CompositionConverter(converters.Pass()));
+ sp.reset(new CompositionConverter(std::move(converters)));
} else {
sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
dst_frames));
@@ -172,7 +173,7 @@
dst_frames));
}
- return sp.Pass();
+ return sp;
}
// For CompositionConverter.
@@ -182,8 +183,8 @@
dst_channels_(0),
dst_frames_(0) {}
-AudioConverter::AudioConverter(int src_channels, size_t src_frames,
- int dst_channels, size_t dst_frames)
+AudioConverter::AudioConverter(size_t src_channels, size_t src_frames,
+ size_t dst_channels, size_t dst_frames)
: src_channels_(src_channels),
src_frames_(src_frames),
dst_channels_(dst_channels),
diff --git a/webrtc/common_audio/audio_converter.h b/webrtc/common_audio/audio_converter.h
index 7d1513b..c5f08c1 100644
--- a/webrtc/common_audio/audio_converter.h
+++ b/webrtc/common_audio/audio_converter.h
@@ -26,9 +26,9 @@
public:
// Returns a new AudioConverter, which will use the supplied format for its
// lifetime. Caller is responsible for the memory.
- static rtc::scoped_ptr<AudioConverter> Create(int src_channels,
+ static rtc::scoped_ptr<AudioConverter> Create(size_t src_channels,
size_t src_frames,
- int dst_channels,
+ size_t dst_channels,
size_t dst_frames);
virtual ~AudioConverter() {};
@@ -39,23 +39,23 @@
virtual void Convert(const float* const* src, size_t src_size,
float* const* dst, size_t dst_capacity) = 0;
- int src_channels() const { return src_channels_; }
+ size_t src_channels() const { return src_channels_; }
size_t src_frames() const { return src_frames_; }
- int dst_channels() const { return dst_channels_; }
+ size_t dst_channels() const { return dst_channels_; }
size_t dst_frames() const { return dst_frames_; }
protected:
AudioConverter();
- AudioConverter(int src_channels, size_t src_frames, int dst_channels,
+ AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
size_t dst_frames);
// Helper to RTC_CHECK that inputs are correctly sized.
void CheckSizes(size_t src_size, size_t dst_capacity) const;
private:
- const int src_channels_;
+ const size_t src_channels_;
const size_t src_frames_;
- const int dst_channels_;
+ const size_t dst_channels_;
const size_t dst_frames_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
diff --git a/webrtc/common_audio/audio_converter_unittest.cc b/webrtc/common_audio/audio_converter_unittest.cc
index c85b96e..dace0bd 100644
--- a/webrtc/common_audio/audio_converter_unittest.cc
+++ b/webrtc/common_audio/audio_converter_unittest.cc
@@ -13,6 +13,7 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/audio_converter.h"
@@ -24,11 +25,11 @@
typedef rtc::scoped_ptr<ChannelBuffer<float>> ScopedBuffer;
// Sets the signal value to increase by |data| with every sample.
-ScopedBuffer CreateBuffer(const std::vector<float>& data, int frames) {
- const int num_channels = static_cast<int>(data.size());
+ScopedBuffer CreateBuffer(const std::vector<float>& data, size_t frames) {
+ const size_t num_channels = data.size();
ScopedBuffer sb(new ChannelBuffer<float>(frames, num_channels));
- for (int i = 0; i < num_channels; ++i)
- for (int j = 0; j < frames; ++j)
+ for (size_t i = 0; i < num_channels; ++i)
+ for (size_t j = 0; j < frames; ++j)
sb->channels()[i][j] = data[i] * j;
return sb;
}
@@ -56,7 +57,7 @@
float mse = 0;
float variance = 0;
float mean = 0;
- for (int i = 0; i < ref.num_channels(); ++i) {
+ for (size_t i = 0; i < ref.num_channels(); ++i) {
for (size_t j = 0; j < ref.num_frames() - delay; ++j) {
float error = ref.channels()[i][j] - test.channels()[i][j + delay];
mse += error * error;
@@ -85,9 +86,9 @@
// Sets the source to a linearly increasing signal for which we can easily
// generate a reference. Runs the AudioConverter and ensures the output has
// sufficiently high SNR relative to the reference.
-void RunAudioConverterTest(int src_channels,
+void RunAudioConverterTest(size_t src_channels,
int src_sample_rate_hz,
- int dst_channels,
+ size_t dst_channels,
int dst_sample_rate_hz) {
const float kSrcLeft = 0.0002f;
const float kSrcRight = 0.0001f;
@@ -96,8 +97,8 @@
const float dst_left = resampling_factor * kSrcLeft;
const float dst_right = resampling_factor * kSrcRight;
const float dst_mono = (dst_left + dst_right) / 2;
- const int src_frames = src_sample_rate_hz / 100;
- const int dst_frames = dst_sample_rate_hz / 100;
+ const size_t src_frames = static_cast<size_t>(src_sample_rate_hz / 100);
+ const size_t dst_frames = static_cast<size_t>(dst_sample_rate_hz / 100);
std::vector<float> src_data(1, kSrcLeft);
if (src_channels == 2)
@@ -127,8 +128,9 @@
static_cast<size_t>(
PushSincResampler::AlgorithmicDelaySeconds(src_sample_rate_hz) *
dst_sample_rate_hz);
- printf("(%d, %d Hz) -> (%d, %d Hz) ", // SNR reported on the same line later.
- src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
+ // SNR reported on the same line later.
+ printf("(%" PRIuS ", %d Hz) -> (%" PRIuS ", %d Hz) ",
+ src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
rtc::scoped_ptr<AudioConverter> converter = AudioConverter::Create(
src_channels, src_frames, dst_channels, dst_frames);
@@ -141,13 +143,13 @@
TEST(AudioConverterTest, ConversionsPassSNRThreshold) {
const int kSampleRates[] = {8000, 16000, 32000, 44100, 48000};
- const int kSampleRatesSize = sizeof(kSampleRates) / sizeof(*kSampleRates);
- const int kChannels[] = {1, 2};
- const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels);
- for (int src_rate = 0; src_rate < kSampleRatesSize; ++src_rate) {
- for (int dst_rate = 0; dst_rate < kSampleRatesSize; ++dst_rate) {
- for (int src_channel = 0; src_channel < kChannelsSize; ++src_channel) {
- for (int dst_channel = 0; dst_channel < kChannelsSize; ++dst_channel) {
+ const size_t kChannels[] = {1, 2};
+ for (size_t src_rate = 0; src_rate < arraysize(kSampleRates); ++src_rate) {
+ for (size_t dst_rate = 0; dst_rate < arraysize(kSampleRates); ++dst_rate) {
+ for (size_t src_channel = 0; src_channel < arraysize(kChannels);
+ ++src_channel) {
+ for (size_t dst_channel = 0; dst_channel < arraysize(kChannels);
+ ++dst_channel) {
RunAudioConverterTest(kChannels[src_channel], kSampleRates[src_rate],
kChannels[dst_channel], kSampleRates[dst_rate]);
}
diff --git a/webrtc/common_audio/blocker.cc b/webrtc/common_audio/blocker.cc
index 0133550..13432f2 100644
--- a/webrtc/common_audio/blocker.cc
+++ b/webrtc/common_audio/blocker.cc
@@ -22,10 +22,10 @@
const float* const* b,
int b_start_index,
size_t num_frames,
- int num_channels,
+ size_t num_channels,
float* const* result,
size_t result_start_index) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
for (size_t j = 0; j < num_frames; ++j) {
result[i][j + result_start_index] =
a[i][j + a_start_index] + b[i][j + b_start_index];
@@ -37,10 +37,10 @@
void CopyFrames(const float* const* src,
size_t src_start_index,
size_t num_frames,
- int num_channels,
+ size_t num_channels,
float* const* dst,
size_t dst_start_index) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
memcpy(&dst[i][dst_start_index],
&src[i][src_start_index],
num_frames * sizeof(dst[i][dst_start_index]));
@@ -51,10 +51,10 @@
void MoveFrames(const float* const* src,
size_t src_start_index,
size_t num_frames,
- int num_channels,
+ size_t num_channels,
float* const* dst,
size_t dst_start_index) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
memmove(&dst[i][dst_start_index],
&src[i][src_start_index],
num_frames * sizeof(dst[i][dst_start_index]));
@@ -64,8 +64,8 @@
void ZeroOut(float* const* buffer,
size_t starting_idx,
size_t num_frames,
- int num_channels) {
- for (int i = 0; i < num_channels; ++i) {
+ size_t num_channels) {
+ for (size_t i = 0; i < num_channels; ++i) {
memset(&buffer[i][starting_idx], 0,
num_frames * sizeof(buffer[i][starting_idx]));
}
@@ -75,9 +75,9 @@
// stored in |frames|.
void ApplyWindow(const float* window,
size_t num_frames,
- int num_channels,
+ size_t num_channels,
float* const* frames) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
for (size_t j = 0; j < num_frames; ++j) {
frames[i][j] = frames[i][j] * window[j];
}
@@ -100,8 +100,8 @@
Blocker::Blocker(size_t chunk_size,
size_t block_size,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
const float* window,
size_t shift_amount,
BlockerCallback* callback)
@@ -166,8 +166,8 @@
// TODO(claguna): Look at using ring buffers to eliminate some copies.
void Blocker::ProcessChunk(const float* const* input,
size_t chunk_size,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output) {
RTC_CHECK_EQ(chunk_size, chunk_size_);
RTC_CHECK_EQ(num_input_channels, num_input_channels_);
diff --git a/webrtc/common_audio/blocker.h b/webrtc/common_audio/blocker.h
index 025638a..3a67c13 100644
--- a/webrtc/common_audio/blocker.h
+++ b/webrtc/common_audio/blocker.h
@@ -26,8 +26,8 @@
virtual void ProcessBlock(const float* const* input,
size_t num_frames,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output) = 0;
};
@@ -65,23 +65,23 @@
public:
Blocker(size_t chunk_size,
size_t block_size,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
const float* window,
size_t shift_amount,
BlockerCallback* callback);
void ProcessChunk(const float* const* input,
size_t chunk_size,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output);
private:
const size_t chunk_size_;
const size_t block_size_;
- const int num_input_channels_;
- const int num_output_channels_;
+ const size_t num_input_channels_;
+ const size_t num_output_channels_;
// The number of frames of delay to add at the beginning of the first chunk.
const size_t initial_delay_;
diff --git a/webrtc/common_audio/blocker_unittest.cc b/webrtc/common_audio/blocker_unittest.cc
index 397e269..a5a7b56 100644
--- a/webrtc/common_audio/blocker_unittest.cc
+++ b/webrtc/common_audio/blocker_unittest.cc
@@ -11,6 +11,7 @@
#include "webrtc/common_audio/blocker.h"
#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/arraysize.h"
namespace {
@@ -19,10 +20,10 @@
public:
void ProcessBlock(const float* const* input,
size_t num_frames,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output) override {
- for (int i = 0; i < num_output_channels; ++i) {
+ for (size_t i = 0; i < num_output_channels; ++i) {
for (size_t j = 0; j < num_frames; ++j) {
output[i][j] = input[i][j] + 3;
}
@@ -35,10 +36,10 @@
public:
void ProcessBlock(const float* const* input,
size_t num_frames,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output) override {
- for (int i = 0; i < num_output_channels; ++i) {
+ for (size_t i = 0; i < num_output_channels; ++i) {
for (size_t j = 0; j < num_frames; ++j) {
output[i][j] = input[i][j];
}
@@ -56,16 +57,16 @@
class BlockerTest : public ::testing::Test {
protected:
void RunTest(Blocker* blocker,
- int chunk_size,
- int num_frames,
+ size_t chunk_size,
+ size_t num_frames,
const float* const* input,
float* const* input_chunk,
float* const* output,
float* const* output_chunk,
- int num_input_channels,
- int num_output_channels) {
- int start = 0;
- int end = chunk_size - 1;
+ size_t num_input_channels,
+ size_t num_output_channels) {
+ size_t start = 0;
+ size_t end = chunk_size - 1;
while (end < num_frames) {
CopyTo(input_chunk, 0, start, num_input_channels, chunk_size, input);
blocker->ProcessChunk(input_chunk,
@@ -75,28 +76,28 @@
output_chunk);
CopyTo(output, start, 0, num_output_channels, chunk_size, output_chunk);
- start = start + chunk_size;
- end = end + chunk_size;
+ start += chunk_size;
+ end += chunk_size;
}
}
void ValidateSignalEquality(const float* const* expected,
const float* const* actual,
- int num_channels,
- int num_frames) {
- for (int i = 0; i < num_channels; ++i) {
- for (int j = 0; j < num_frames; ++j) {
+ size_t num_channels,
+ size_t num_frames) {
+ for (size_t i = 0; i < num_channels; ++i) {
+ for (size_t j = 0; j < num_frames; ++j) {
EXPECT_FLOAT_EQ(expected[i][j], actual[i][j]);
}
}
}
void ValidateInitialDelay(const float* const* output,
- int num_channels,
- int num_frames,
- int initial_delay) {
- for (int i = 0; i < num_channels; ++i) {
- for (int j = 0; j < num_frames; ++j) {
+ size_t num_channels,
+ size_t num_frames,
+ size_t initial_delay) {
+ for (size_t i = 0; i < num_channels; ++i) {
+ for (size_t j = 0; j < num_frames; ++j) {
if (j < initial_delay) {
EXPECT_FLOAT_EQ(output[i][j], 0.f);
} else {
@@ -107,12 +108,12 @@
}
static void CopyTo(float* const* dst,
- int start_index_dst,
- int start_index_src,
- int num_channels,
- int num_frames,
+ size_t start_index_dst,
+ size_t start_index_src,
+ size_t num_channels,
+ size_t num_frames,
const float* const* src) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
memcpy(&dst[i][start_index_dst],
&src[i][start_index_src],
num_frames * sizeof(float));
@@ -121,12 +122,12 @@
};
TEST_F(BlockerTest, TestBlockerMutuallyPrimeChunkandBlockSize) {
- const int kNumInputChannels = 3;
- const int kNumOutputChannels = 2;
- const int kNumFrames = 10;
- const int kBlockSize = 4;
- const int kChunkSize = 5;
- const int kShiftAmount = 2;
+ const size_t kNumInputChannels = 3;
+ const size_t kNumOutputChannels = 2;
+ const size_t kNumFrames = 10;
+ const size_t kBlockSize = 4;
+ const size_t kChunkSize = 5;
+ const size_t kShiftAmount = 2;
const float kInput[kNumInputChannels][kNumFrames] = {
{1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
@@ -174,12 +175,12 @@
}
TEST_F(BlockerTest, TestBlockerMutuallyPrimeShiftAndBlockSize) {
- const int kNumInputChannels = 3;
- const int kNumOutputChannels = 2;
- const int kNumFrames = 12;
- const int kBlockSize = 4;
- const int kChunkSize = 6;
- const int kShiftAmount = 3;
+ const size_t kNumInputChannels = 3;
+ const size_t kNumOutputChannels = 2;
+ const size_t kNumFrames = 12;
+ const size_t kBlockSize = 4;
+ const size_t kChunkSize = 6;
+ const size_t kShiftAmount = 3;
const float kInput[kNumInputChannels][kNumFrames] = {
{1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
@@ -227,12 +228,12 @@
}
TEST_F(BlockerTest, TestBlockerNoOverlap) {
- const int kNumInputChannels = 3;
- const int kNumOutputChannels = 2;
- const int kNumFrames = 12;
- const int kBlockSize = 4;
- const int kChunkSize = 4;
- const int kShiftAmount = 4;
+ const size_t kNumInputChannels = 3;
+ const size_t kNumOutputChannels = 2;
+ const size_t kNumFrames = 12;
+ const size_t kBlockSize = 4;
+ const size_t kChunkSize = 4;
+ const size_t kShiftAmount = 4;
const float kInput[kNumInputChannels][kNumFrames] = {
{1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
@@ -280,21 +281,21 @@
}
TEST_F(BlockerTest, InitialDelaysAreMinimum) {
- const int kNumInputChannels = 3;
- const int kNumOutputChannels = 2;
- const int kNumFrames = 1280;
- const int kChunkSize[] =
+ const size_t kNumInputChannels = 3;
+ const size_t kNumOutputChannels = 2;
+ const size_t kNumFrames = 1280;
+ const size_t kChunkSize[] =
{80, 80, 80, 80, 80, 80, 160, 160, 160, 160, 160, 160};
- const int kBlockSize[] =
+ const size_t kBlockSize[] =
{64, 64, 64, 128, 128, 128, 128, 128, 128, 256, 256, 256};
- const int kShiftAmount[] =
+ const size_t kShiftAmount[] =
{16, 32, 64, 32, 64, 128, 32, 64, 128, 64, 128, 256};
- const int kInitialDelay[] =
+ const size_t kInitialDelay[] =
{48, 48, 48, 112, 112, 112, 96, 96, 96, 224, 224, 224};
float input[kNumInputChannels][kNumFrames];
- for (int i = 0; i < kNumInputChannels; ++i) {
- for (int j = 0; j < kNumFrames; ++j) {
+ for (size_t i = 0; i < kNumInputChannels; ++i) {
+ for (size_t j = 0; j < kNumFrames; ++j) {
input[i][j] = i + 1;
}
}
@@ -305,9 +306,9 @@
CopyBlockerCallback callback;
- for (size_t i = 0; i < (sizeof(kChunkSize) / sizeof(*kChunkSize)); ++i) {
+ for (size_t i = 0; i < arraysize(kChunkSize); ++i) {
rtc::scoped_ptr<float[]> window(new float[kBlockSize[i]]);
- for (int j = 0; j < kBlockSize[i]; ++j) {
+ for (size_t j = 0; j < kBlockSize[i]; ++j) {
window[j] = 1.f;
}
diff --git a/webrtc/common_audio/channel_buffer.cc b/webrtc/common_audio/channel_buffer.cc
index d3dc7c0..44520c6 100644
--- a/webrtc/common_audio/channel_buffer.cc
+++ b/webrtc/common_audio/channel_buffer.cc
@@ -13,7 +13,7 @@
namespace webrtc {
IFChannelBuffer::IFChannelBuffer(size_t num_frames,
- int num_channels,
+ size_t num_channels,
size_t num_bands)
: ivalid_(true),
ibuf_(num_frames, num_channels, num_bands),
@@ -47,7 +47,7 @@
assert(ivalid_);
const int16_t* const* int_channels = ibuf_.channels();
float* const* float_channels = fbuf_.channels();
- for (int i = 0; i < ibuf_.num_channels(); ++i) {
+ for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
for (size_t j = 0; j < ibuf_.num_frames(); ++j) {
float_channels[i][j] = int_channels[i][j];
}
@@ -61,7 +61,7 @@
assert(fvalid_);
int16_t* const* int_channels = ibuf_.channels();
const float* const* float_channels = fbuf_.channels();
- for (int i = 0; i < ibuf_.num_channels(); ++i) {
+ for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
FloatS16ToS16(float_channels[i],
ibuf_.num_frames(),
int_channels[i]);
diff --git a/webrtc/common_audio/channel_buffer.h b/webrtc/common_audio/channel_buffer.h
index 6050090..d906916 100644
--- a/webrtc/common_audio/channel_buffer.h
+++ b/webrtc/common_audio/channel_buffer.h
@@ -40,7 +40,7 @@
class ChannelBuffer {
public:
ChannelBuffer(size_t num_frames,
- int num_channels,
+ size_t num_channels,
size_t num_bands = 1)
: data_(new T[num_frames * num_channels]()),
channels_(new T*[num_channels * num_bands]),
@@ -49,7 +49,7 @@
num_frames_per_band_(num_frames / num_bands),
num_channels_(num_channels),
num_bands_(num_bands) {
- for (int i = 0; i < num_channels_; ++i) {
+ for (size_t i = 0; i < num_channels_; ++i) {
for (size_t j = 0; j < num_bands_; ++j) {
channels_[j * num_channels_ + i] =
&data_[i * num_frames_ + j * num_frames_per_band_];
@@ -90,12 +90,12 @@
// 0 <= channel < |num_channels_|
// 0 <= band < |num_bands_|
// 0 <= sample < |num_frames_per_band_|
- const T* const* bands(int channel) const {
+ const T* const* bands(size_t channel) const {
RTC_DCHECK_LT(channel, num_channels_);
- RTC_DCHECK_GE(channel, 0);
+ RTC_DCHECK_GE(channel, 0u);
return &bands_[channel * num_bands_];
}
- T* const* bands(int channel) {
+ T* const* bands(size_t channel) {
const ChannelBuffer<T>* t = this;
return const_cast<T* const*>(t->bands(channel));
}
@@ -104,7 +104,7 @@
// Returns |slice| for convenience.
const T* const* Slice(T** slice, size_t start_frame) const {
RTC_DCHECK_LT(start_frame, num_frames_);
- for (int i = 0; i < num_channels_; ++i)
+ for (size_t i = 0; i < num_channels_; ++i)
slice[i] = &channels_[i][start_frame];
return slice;
}
@@ -115,7 +115,7 @@
size_t num_frames() const { return num_frames_; }
size_t num_frames_per_band() const { return num_frames_per_band_; }
- int num_channels() const { return num_channels_; }
+ size_t num_channels() const { return num_channels_; }
size_t num_bands() const { return num_bands_; }
size_t size() const {return num_frames_ * num_channels_; }
@@ -130,7 +130,7 @@
rtc::scoped_ptr<T* []> bands_;
const size_t num_frames_;
const size_t num_frames_per_band_;
- const int num_channels_;
+ const size_t num_channels_;
const size_t num_bands_;
};
@@ -142,7 +142,7 @@
// fbuf() until the next call to any of the other functions.
class IFChannelBuffer {
public:
- IFChannelBuffer(size_t num_frames, int num_channels, size_t num_bands = 1);
+ IFChannelBuffer(size_t num_frames, size_t num_channels, size_t num_bands = 1);
ChannelBuffer<int16_t>* ibuf();
ChannelBuffer<float>* fbuf();
@@ -151,7 +151,7 @@
size_t num_frames() const { return ibuf_.num_frames(); }
size_t num_frames_per_band() const { return ibuf_.num_frames_per_band(); }
- int num_channels() const { return ibuf_.num_channels(); }
+ size_t num_channels() const { return ibuf_.num_channels(); }
size_t num_bands() const { return ibuf_.num_bands(); }
private:
diff --git a/webrtc/common_audio/common_audio.gyp b/webrtc/common_audio/common_audio.gyp
index 884a8af..f0a6fc9 100644
--- a/webrtc/common_audio/common_audio.gyp
+++ b/webrtc/common_audio/common_audio.gyp
@@ -101,6 +101,7 @@
'signal_processing/vector_scaling_operations.c',
'sparse_fir_filter.cc',
'sparse_fir_filter.h',
+ 'swap_queue.h',
'vad/include/vad.h',
'vad/include/webrtc_vad.h',
'vad/vad.cc',
@@ -227,9 +228,10 @@
},
], # targets
}],
- ['include_tests==1', {
+ ['include_tests==1 and OS!="ios"', {
'targets' : [
{
+ # Does not compile on iOS: webrtc:4755.
'target_name': 'common_audio_unittests',
'type': '<(gtest_target_type)',
'dependencies': [
@@ -256,6 +258,7 @@
'signal_processing/real_fft_unittest.cc',
'signal_processing/signal_processing_unittest.cc',
'sparse_fir_filter_unittest.cc',
+ 'swap_queue_unittest.cc',
'vad/vad_core_unittest.cc',
'vad/vad_filterbank_unittest.cc',
'vad/vad_gmm_unittest.cc',
diff --git a/webrtc/common_audio/include/audio_util.h b/webrtc/common_audio/include/audio_util.h
index 2c0028c..55dfc06 100644
--- a/webrtc/common_audio/include/audio_util.h
+++ b/webrtc/common_audio/include/audio_util.h
@@ -87,11 +87,11 @@
template <typename T>
void Deinterleave(const T* interleaved,
size_t samples_per_channel,
- int num_channels,
+ size_t num_channels,
T* const* deinterleaved) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
T* channel = deinterleaved[i];
- int interleaved_idx = i;
+ size_t interleaved_idx = i;
for (size_t j = 0; j < samples_per_channel; ++j) {
channel[j] = interleaved[interleaved_idx];
interleaved_idx += num_channels;
@@ -105,11 +105,11 @@
template <typename T>
void Interleave(const T* const* deinterleaved,
size_t samples_per_channel,
- int num_channels,
+ size_t num_channels,
T* interleaved) {
- for (int i = 0; i < num_channels; ++i) {
+ for (size_t i = 0; i < num_channels; ++i) {
const T* channel = deinterleaved[i];
- int interleaved_idx = i;
+ size_t interleaved_idx = i;
for (size_t j = 0; j < samples_per_channel; ++j) {
interleaved[interleaved_idx] = channel[j];
interleaved_idx += num_channels;
diff --git a/webrtc/common_audio/lapped_transform.cc b/webrtc/common_audio/lapped_transform.cc
index c01f1d9..5ab1db1 100644
--- a/webrtc/common_audio/lapped_transform.cc
+++ b/webrtc/common_audio/lapped_transform.cc
@@ -21,14 +21,14 @@
void LappedTransform::BlockThunk::ProcessBlock(const float* const* input,
size_t num_frames,
- int num_input_channels,
- int num_output_channels,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output) {
RTC_CHECK_EQ(num_input_channels, parent_->num_in_channels_);
RTC_CHECK_EQ(num_output_channels, parent_->num_out_channels_);
RTC_CHECK_EQ(parent_->block_length_, num_frames);
- for (int i = 0; i < num_input_channels; ++i) {
+ for (size_t i = 0; i < num_input_channels; ++i) {
memcpy(parent_->real_buf_.Row(i), input[i],
num_frames * sizeof(*input[0]));
parent_->fft_->Forward(parent_->real_buf_.Row(i),
@@ -44,7 +44,7 @@
num_output_channels,
parent_->cplx_post_.Array());
- for (int i = 0; i < num_output_channels; ++i) {
+ for (size_t i = 0; i < num_output_channels; ++i) {
parent_->fft_->Inverse(parent_->cplx_post_.Row(i),
parent_->real_buf_.Row(i));
memcpy(output[i], parent_->real_buf_.Row(i),
@@ -52,8 +52,8 @@
}
}
-LappedTransform::LappedTransform(int num_in_channels,
- int num_out_channels,
+LappedTransform::LappedTransform(size_t num_in_channels,
+ size_t num_out_channels,
size_t chunk_length,
const float* window,
size_t block_length,
diff --git a/webrtc/common_audio/lapped_transform.h b/webrtc/common_audio/lapped_transform.h
index 21e10e3..1373ca1 100644
--- a/webrtc/common_audio/lapped_transform.h
+++ b/webrtc/common_audio/lapped_transform.h
@@ -35,8 +35,8 @@
virtual ~Callback() {}
virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
- int num_in_channels, size_t frames,
- int num_out_channels,
+ size_t num_in_channels, size_t frames,
+ size_t num_out_channels,
std::complex<float>* const* out_block) = 0;
};
@@ -46,8 +46,8 @@
// |block_length| defines the length of a block, in samples.
// |shift_amount| is in samples. |callback| is the caller-owned audio
// processing function called for each block of the input chunk.
- LappedTransform(int num_in_channels,
- int num_out_channels,
+ LappedTransform(size_t num_in_channels,
+ size_t num_out_channels,
size_t chunk_length,
const float* window,
size_t block_length,
@@ -75,7 +75,7 @@
// in_chunk.
//
// Returns the same num_in_channels passed to the LappedTransform constructor.
- int num_in_channels() const { return num_in_channels_; }
+ size_t num_in_channels() const { return num_in_channels_; }
// Get the number of output channels.
//
@@ -84,7 +84,7 @@
//
// Returns the same num_out_channels passed to the LappedTransform
// constructor.
- int num_out_channels() const { return num_out_channels_; }
+ size_t num_out_channels() const { return num_out_channels_; }
private:
// Internal middleware callback, given to the blocker. Transforms each block
@@ -93,16 +93,18 @@
public:
explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
- virtual void ProcessBlock(const float* const* input, size_t num_frames,
- int num_input_channels, int num_output_channels,
+ virtual void ProcessBlock(const float* const* input,
+ size_t num_frames,
+ size_t num_input_channels,
+ size_t num_output_channels,
float* const* output);
private:
LappedTransform* const parent_;
} blocker_callback_;
- const int num_in_channels_;
- const int num_out_channels_;
+ const size_t num_in_channels_;
+ const size_t num_out_channels_;
const size_t block_length_;
const size_t chunk_length_;
diff --git a/webrtc/common_audio/lapped_transform_unittest.cc b/webrtc/common_audio/lapped_transform_unittest.cc
index f688cc2..a78488e 100644
--- a/webrtc/common_audio/lapped_transform_unittest.cc
+++ b/webrtc/common_audio/lapped_transform_unittest.cc
@@ -25,23 +25,23 @@
NoopCallback() : block_num_(0) {}
virtual void ProcessAudioBlock(const complex<float>* const* in_block,
- int in_channels,
+ size_t in_channels,
size_t frames,
- int out_channels,
+ size_t out_channels,
complex<float>* const* out_block) {
RTC_CHECK_EQ(in_channels, out_channels);
- for (int i = 0; i < out_channels; ++i) {
+ for (size_t i = 0; i < out_channels; ++i) {
memcpy(out_block[i], in_block[i], sizeof(**in_block) * frames);
}
++block_num_;
}
- int block_num() {
+ size_t block_num() {
return block_num_;
}
private:
- int block_num_;
+ size_t block_num_;
};
class FftCheckerCallback : public webrtc::LappedTransform::Callback {
@@ -49,9 +49,9 @@
FftCheckerCallback() : block_num_(0) {}
virtual void ProcessAudioBlock(const complex<float>* const* in_block,
- int in_channels,
+ size_t in_channels,
size_t frames,
- int out_channels,
+ size_t out_channels,
complex<float>* const* out_block) {
RTC_CHECK_EQ(in_channels, out_channels);
@@ -69,12 +69,12 @@
}
}
- int block_num() {
+ size_t block_num() {
return block_num_;
}
private:
- int block_num_;
+ size_t block_num_;
};
void SetFloatArray(float value, int rows, int cols, float* const* array) {
@@ -90,10 +90,10 @@
namespace webrtc {
TEST(LappedTransformTest, Windowless) {
- const int kChannels = 3;
- const int kChunkLength = 512;
- const int kBlockLength = 64;
- const int kShiftAmount = 64;
+ const size_t kChannels = 3;
+ const size_t kChunkLength = 512;
+ const size_t kBlockLength = 64;
+ const size_t kShiftAmount = 64;
NoopCallback noop;
// Rectangular window.
@@ -118,8 +118,8 @@
trans.ProcessChunk(in_chunk, out_chunk);
- for (int i = 0; i < kChannels; ++i) {
- for (int j = 0; j < kChunkLength; ++j) {
+ for (size_t i = 0; i < kChannels; ++i) {
+ for (size_t j = 0; j < kChunkLength; ++j) {
ASSERT_NEAR(out_chunk[i][j], 2.0f, 1e-5f);
}
}
@@ -128,9 +128,9 @@
}
TEST(LappedTransformTest, IdentityProcessor) {
- const int kChunkLength = 512;
- const int kBlockLength = 64;
- const int kShiftAmount = 32;
+ const size_t kChunkLength = 512;
+ const size_t kBlockLength = 64;
+ const size_t kShiftAmount = 32;
NoopCallback noop;
// Identity window for |overlap = block_size / 2|.
@@ -149,7 +149,7 @@
trans.ProcessChunk(&in_chunk, &out_chunk);
- for (int i = 0; i < kChunkLength; ++i) {
+ for (size_t i = 0; i < kChunkLength; ++i) {
ASSERT_NEAR(out_chunk[i],
(i < kBlockLength - kShiftAmount) ? 0.0f : 2.0f,
1e-5f);
@@ -159,8 +159,8 @@
}
TEST(LappedTransformTest, Callbacks) {
- const int kChunkLength = 512;
- const int kBlockLength = 64;
+ const size_t kChunkLength = 512;
+ const size_t kBlockLength = 64;
FftCheckerCallback call;
// Rectangular window.
@@ -183,7 +183,7 @@
}
TEST(LappedTransformTest, chunk_length) {
- const int kBlockLength = 64;
+ const size_t kBlockLength = 64;
FftCheckerCallback call;
const float window[kBlockLength] = {};
diff --git a/webrtc/common_audio/real_fourier.cc b/webrtc/common_audio/real_fourier.cc
index fef3c60..55ec49c 100644
--- a/webrtc/common_audio/real_fourier.cc
+++ b/webrtc/common_audio/real_fourier.cc
@@ -19,7 +19,7 @@
using std::complex;
-const int RealFourier::kFftBufferAlignment = 32;
+const size_t RealFourier::kFftBufferAlignment = 32;
rtc::scoped_ptr<RealFourier> RealFourier::Create(int fft_order) {
#if defined(RTC_USE_OPENMAX_DL)
diff --git a/webrtc/common_audio/real_fourier.h b/webrtc/common_audio/real_fourier.h
index ce3bbff..0be56a5 100644
--- a/webrtc/common_audio/real_fourier.h
+++ b/webrtc/common_audio/real_fourier.h
@@ -30,7 +30,7 @@
fft_cplx_scoper;
// The alignment required for all input and output buffers, in bytes.
- static const int kFftBufferAlignment;
+ static const size_t kFftBufferAlignment;
// Construct a wrapper instance for the given input order, which must be
// between 1 and kMaxFftOrder, inclusively.
diff --git a/webrtc/common_audio/real_fourier_unittest.cc b/webrtc/common_audio/real_fourier_unittest.cc
index 5c85421..eb5880e 100644
--- a/webrtc/common_audio/real_fourier_unittest.cc
+++ b/webrtc/common_audio/real_fourier_unittest.cc
@@ -26,15 +26,15 @@
RealFourier::fft_real_scoper real;
real = RealFourier::AllocRealBuffer(3);
ASSERT_TRUE(real.get() != nullptr);
- int64_t ptr_value = reinterpret_cast<int64_t>(real.get());
- EXPECT_EQ(0, ptr_value % RealFourier::kFftBufferAlignment);
+ uintptr_t ptr_value = reinterpret_cast<uintptr_t>(real.get());
+ EXPECT_EQ(0u, ptr_value % RealFourier::kFftBufferAlignment);
}
{
RealFourier::fft_cplx_scoper cplx;
cplx = RealFourier::AllocCplxBuffer(3);
ASSERT_TRUE(cplx.get() != nullptr);
- int64_t ptr_value = reinterpret_cast<int64_t>(cplx.get());
- EXPECT_EQ(0, ptr_value % RealFourier::kFftBufferAlignment);
+ uintptr_t ptr_value = reinterpret_cast<uintptr_t>(cplx.get());
+ EXPECT_EQ(0u, ptr_value % RealFourier::kFftBufferAlignment);
}
}
diff --git a/webrtc/common_audio/resampler/include/push_resampler.h b/webrtc/common_audio/resampler/include/push_resampler.h
index b5c0003..eeda790 100644
--- a/webrtc/common_audio/resampler/include/push_resampler.h
+++ b/webrtc/common_audio/resampler/include/push_resampler.h
@@ -29,7 +29,7 @@
// Must be called whenever the parameters change. Free to be called at any
// time as it is a no-op if parameters have not changed since the last call.
int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
- int num_channels);
+ size_t num_channels);
// Returns the total number of samples provided in destination (e.g. 32 kHz,
// 2 channel audio gives 640 samples).
@@ -40,7 +40,7 @@
rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_;
int src_sample_rate_hz_;
int dst_sample_rate_hz_;
- int num_channels_;
+ size_t num_channels_;
rtc::scoped_ptr<T[]> src_left_;
rtc::scoped_ptr<T[]> src_right_;
rtc::scoped_ptr<T[]> dst_left_;
diff --git a/webrtc/common_audio/resampler/include/resampler.h b/webrtc/common_audio/resampler/include/resampler.h
index 0d4c1af..e26ac90 100644
--- a/webrtc/common_audio/resampler/include/resampler.h
+++ b/webrtc/common_audio/resampler/include/resampler.h
@@ -28,14 +28,14 @@
public:
Resampler();
- Resampler(int inFreq, int outFreq, int num_channels);
+ Resampler(int inFreq, int outFreq, size_t num_channels);
~Resampler();
// Reset all states
- int Reset(int inFreq, int outFreq, int num_channels);
+ int Reset(int inFreq, int outFreq, size_t num_channels);
// Reset all states if any parameter has changed
- int ResetIfNeeded(int inFreq, int outFreq, int num_channels);
+ int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels);
// Resample samplesIn to samplesOut.
int Push(const int16_t* samplesIn, size_t lengthIn, int16_t* samplesOut,
@@ -83,7 +83,7 @@
int my_in_frequency_khz_;
int my_out_frequency_khz_;
ResamplerMode my_mode_;
- int num_channels_;
+ size_t num_channels_;
// Extra instance for stereo
Resampler* slave_left_;
diff --git a/webrtc/common_audio/resampler/push_resampler.cc b/webrtc/common_audio/resampler/push_resampler.cc
index 566acde..f654e9a 100644
--- a/webrtc/common_audio/resampler/push_resampler.cc
+++ b/webrtc/common_audio/resampler/push_resampler.cc
@@ -32,7 +32,7 @@
template <typename T>
int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
int dst_sample_rate_hz,
- int num_channels) {
+ size_t num_channels) {
if (src_sample_rate_hz == src_sample_rate_hz_ &&
dst_sample_rate_hz == dst_sample_rate_hz_ &&
num_channels == num_channels_)
@@ -68,10 +68,8 @@
template <typename T>
int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
size_t dst_capacity) {
- const size_t src_size_10ms =
- static_cast<size_t>(src_sample_rate_hz_ * num_channels_ / 100);
- const size_t dst_size_10ms =
- static_cast<size_t>(dst_sample_rate_hz_ * num_channels_ / 100);
+ const size_t src_size_10ms = src_sample_rate_hz_ * num_channels_ / 100;
+ const size_t dst_size_10ms = dst_sample_rate_hz_ * num_channels_ / 100;
if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
return -1;
diff --git a/webrtc/common_audio/resampler/resampler.cc b/webrtc/common_audio/resampler/resampler.cc
index c9e7a1f..7c690fc 100644
--- a/webrtc/common_audio/resampler/resampler.cc
+++ b/webrtc/common_audio/resampler/resampler.cc
@@ -39,7 +39,7 @@
slave_right_(nullptr) {
}
-Resampler::Resampler(int inFreq, int outFreq, int num_channels)
+Resampler::Resampler(int inFreq, int outFreq, size_t num_channels)
: Resampler() {
Reset(inFreq, outFreq, num_channels);
}
@@ -76,7 +76,7 @@
}
}
-int Resampler::ResetIfNeeded(int inFreq, int outFreq, int num_channels)
+int Resampler::ResetIfNeeded(int inFreq, int outFreq, size_t num_channels)
{
int tmpInFreq_kHz = inFreq / 1000;
int tmpOutFreq_kHz = outFreq / 1000;
@@ -91,7 +91,7 @@
}
}
-int Resampler::Reset(int inFreq, int outFreq, int num_channels)
+int Resampler::Reset(int inFreq, int outFreq, size_t num_channels)
{
if (num_channels != 1 && num_channels != 2) {
return -1;
diff --git a/webrtc/common_audio/signal_processing/real_fft_unittest.cc b/webrtc/common_audio/signal_processing/real_fft_unittest.cc
index 9bd35cd..fa98836 100644
--- a/webrtc/common_audio/signal_processing/real_fft_unittest.cc
+++ b/webrtc/common_audio/signal_processing/real_fft_unittest.cc
@@ -10,7 +10,6 @@
#include "webrtc/common_audio/signal_processing/include/real_fft.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"
#include "testing/gtest/include/gtest/gtest.h"
diff --git a/webrtc/common_audio/swap_queue.h b/webrtc/common_audio/swap_queue.h
new file mode 100644
index 0000000..d8bb5c0
--- /dev/null
+++ b/webrtc/common_audio/swap_queue.h
@@ -0,0 +1,210 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_AUDIO_SWAP_QUEUE_H_
+#define WEBRTC_COMMON_AUDIO_SWAP_QUEUE_H_
+
+#include <algorithm>
+#include <utility>
+#include <vector>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/criticalsection.h"
+
+namespace webrtc {
+
+namespace internal {
+
+// (Internal; please don't use outside this file.)
+template <typename T>
+bool NoopSwapQueueItemVerifierFunction(const T&) {
+ return true;
+}
+
+} // namespace internal
+
+// Functor to use when supplying a verifier function for the queue.
+template <typename T,
+ bool (*QueueItemVerifierFunction)(const T&) =
+ internal::NoopSwapQueueItemVerifierFunction>
+class SwapQueueItemVerifier {
+ public:
+ bool operator()(const T& t) const { return QueueItemVerifierFunction(t); }
+};
+
+// This class is a fixed-size queue. A producer calls Insert() to insert
+// an element of type T at the back of the queue, and a consumer calls
+// Remove() to remove an element from the front of the queue. It's safe
+// for the producer(s) and the consumer(s) to access the queue
+// concurrently, from different threads.
+//
+// To avoid the construction, copying, and destruction of Ts that a naive
+// queue implementation would require, for each "full" T passed from
+// producer to consumer, SwapQueue<T> passes an "empty" T in the other
+// direction (an "empty" T is one that contains nothing of value for the
+// consumer). This bidirectional movement is implemented with swap().
+//
+// // Create queue:
+// Bottle proto(568); // Prepare an empty Bottle. Heap allocates space for
+// // 568 ml.
+// SwapQueue<Bottle> q(N, proto); // Init queue with N copies of proto.
+// // Each copy allocates on the heap.
+// // Producer pseudo-code:
+// Bottle b(568); // Prepare an empty Bottle. Heap allocates space for 568 ml.
+// loop {
+// b.Fill(amount); // Where amount <= 568 ml.
+// q.Insert(&b); // Swap our full Bottle for an empty one from q.
+// }
+//
+// // Consumer pseudo-code:
+// Bottle b(568); // Prepare an empty Bottle. Heap allocates space for 568 ml.
+// loop {
+// q.Remove(&b); // Swap our empty Bottle for the next-in-line full Bottle.
+// Drink(&b);
+// }
+//
+// For a well-behaved Bottle class, there are no allocations in the
+// producer, since it just fills an empty Bottle that's already large
+// enough; no deallocations in the consumer, since it returns each empty
+// Bottle to the queue after having drunk it; and no copies along the
+// way, since the queue uses swap() everywhere to move full Bottles in
+// one direction and empty ones in the other.
+template <typename T, typename QueueItemVerifier = SwapQueueItemVerifier<T>>
+class SwapQueue {
+ public:
+ // Creates a queue of size size and fills it with default constructed Ts.
+ explicit SwapQueue(size_t size) : queue_(size) {
+ RTC_DCHECK(VerifyQueueSlots());
+ }
+
+ // Same as above and accepts an item verification functor.
+ SwapQueue(size_t size, const QueueItemVerifier& queue_item_verifier)
+ : queue_item_verifier_(queue_item_verifier), queue_(size) {
+ RTC_DCHECK(VerifyQueueSlots());
+ }
+
+ // Creates a queue of size size and fills it with copies of prototype.
+ SwapQueue(size_t size, const T& prototype) : queue_(size, prototype) {
+ RTC_DCHECK(VerifyQueueSlots());
+ }
+
+ // Same as above and accepts an item verification functor.
+ SwapQueue(size_t size,
+ const T& prototype,
+ const QueueItemVerifier& queue_item_verifier)
+ : queue_item_verifier_(queue_item_verifier), queue_(size, prototype) {
+ RTC_DCHECK(VerifyQueueSlots());
+ }
+
+ // Resets the queue to have zero content wile maintaining the queue size.
+ void Clear() {
+ rtc::CritScope cs(&crit_queue_);
+ next_write_index_ = 0;
+ next_read_index_ = 0;
+ num_elements_ = 0;
+ }
+
+ // Inserts a "full" T at the back of the queue by swapping *input with an
+ // "empty" T from the queue.
+ // Returns true if the item was inserted or false if not (the queue was full).
+ // When specified, the T given in *input must pass the ItemVerifier() test.
+ // The contents of *input after the call are then also guaranteed to pass the
+ // ItemVerifier() test.
+ bool Insert(T* input) WARN_UNUSED_RESULT {
+ RTC_DCHECK(input);
+
+ rtc::CritScope cs(&crit_queue_);
+
+ RTC_DCHECK(queue_item_verifier_(*input));
+
+ if (num_elements_ == queue_.size()) {
+ return false;
+ }
+
+ using std::swap;
+ swap(*input, queue_[next_write_index_]);
+
+ ++next_write_index_;
+ if (next_write_index_ == queue_.size()) {
+ next_write_index_ = 0;
+ }
+
+ ++num_elements_;
+
+ RTC_DCHECK_LT(next_write_index_, queue_.size());
+ RTC_DCHECK_LE(num_elements_, queue_.size());
+
+ return true;
+ }
+
+ // Removes the frontmost "full" T from the queue by swapping it with
+ // the "empty" T in *output.
+ // Returns true if an item could be removed or false if not (the queue was
+ // empty). When specified, The T given in *output must pass the ItemVerifier()
+ // test and the contents of *output after the call are then also guaranteed to
+ // pass the ItemVerifier() test.
+ bool Remove(T* output) WARN_UNUSED_RESULT {
+ RTC_DCHECK(output);
+
+ rtc::CritScope cs(&crit_queue_);
+
+ RTC_DCHECK(queue_item_verifier_(*output));
+
+ if (num_elements_ == 0) {
+ return false;
+ }
+
+ using std::swap;
+ swap(*output, queue_[next_read_index_]);
+
+ ++next_read_index_;
+ if (next_read_index_ == queue_.size()) {
+ next_read_index_ = 0;
+ }
+
+ --num_elements_;
+
+ RTC_DCHECK_LT(next_read_index_, queue_.size());
+ RTC_DCHECK_LE(num_elements_, queue_.size());
+
+ return true;
+ }
+
+ private:
+ // Verify that the queue slots complies with the ItemVerifier test.
+ bool VerifyQueueSlots() {
+ rtc::CritScope cs(&crit_queue_);
+ for (const auto& v : queue_) {
+ RTC_DCHECK(queue_item_verifier_(v));
+ }
+ return true;
+ }
+
+ rtc::CriticalSection crit_queue_;
+
+ // TODO(peah): Change this to use std::function() once we can use C++11 std
+ // lib.
+ QueueItemVerifier queue_item_verifier_ GUARDED_BY(crit_queue_);
+
+ // (next_read_index_ + num_elements_) % queue_.size() =
+ // next_write_index_
+ size_t next_write_index_ GUARDED_BY(crit_queue_) = 0;
+ size_t next_read_index_ GUARDED_BY(crit_queue_) = 0;
+ size_t num_elements_ GUARDED_BY(crit_queue_) = 0;
+
+ // queue_.size() is constant.
+ std::vector<T> queue_ GUARDED_BY(crit_queue_);
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(SwapQueue);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_COMMON_AUDIO_SWAP_QUEUE_H_
diff --git a/webrtc/common_audio/swap_queue_unittest.cc b/webrtc/common_audio/swap_queue_unittest.cc
new file mode 100644
index 0000000..104e494
--- /dev/null
+++ b/webrtc/common_audio/swap_queue_unittest.cc
@@ -0,0 +1,225 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/common_audio/swap_queue.h"
+
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+// Test parameter for the basic sample based SwapQueue Tests.
+const size_t kChunkSize = 3;
+
+// Queue item verification function for the vector test.
+bool LengthVerifierFunction(const std::vector<int>& v) {
+ return v.size() == kChunkSize;
+}
+
+// Queue item verifier for the vector test.
+class LengthVerifierFunctor {
+ public:
+ explicit LengthVerifierFunctor(size_t length) : length_(length) {}
+
+ bool operator()(const std::vector<int>& v) const {
+ return v.size() == length_;
+ }
+
+ private:
+ size_t length_;
+};
+
+} // anonymous namespace
+
+TEST(SwapQueueTest, BasicOperation) {
+ std::vector<int> i(kChunkSize, 0);
+ SwapQueue<std::vector<int>> queue(2, i);
+
+ EXPECT_TRUE(queue.Insert(&i));
+ EXPECT_EQ(i.size(), kChunkSize);
+ EXPECT_TRUE(queue.Insert(&i));
+ EXPECT_EQ(i.size(), kChunkSize);
+ EXPECT_TRUE(queue.Remove(&i));
+ EXPECT_EQ(i.size(), kChunkSize);
+ EXPECT_TRUE(queue.Remove(&i));
+ EXPECT_EQ(i.size(), kChunkSize);
+}
+
+TEST(SwapQueueTest, FullQueue) {
+ SwapQueue<int> queue(2);
+
+ // Fill the queue.
+ int i = 0;
+ EXPECT_TRUE(queue.Insert(&i));
+ i = 1;
+ EXPECT_TRUE(queue.Insert(&i));
+
+ // Ensure that the value is not swapped when doing an Insert
+ // on a full queue.
+ i = 2;
+ EXPECT_FALSE(queue.Insert(&i));
+ EXPECT_EQ(i, 2);
+
+ // Ensure that the Insert didn't overwrite anything in the queue.
+ EXPECT_TRUE(queue.Remove(&i));
+ EXPECT_EQ(i, 0);
+ EXPECT_TRUE(queue.Remove(&i));
+ EXPECT_EQ(i, 1);
+}
+
+TEST(SwapQueueTest, EmptyQueue) {
+ SwapQueue<int> queue(2);
+ int i = 0;
+ EXPECT_FALSE(queue.Remove(&i));
+ EXPECT_TRUE(queue.Insert(&i));
+ EXPECT_TRUE(queue.Remove(&i));
+ EXPECT_FALSE(queue.Remove(&i));
+}
+
+TEST(SwapQueueTest, Clear) {
+ SwapQueue<int> queue(2);
+ int i = 0;
+
+ // Fill the queue.
+ EXPECT_TRUE(queue.Insert(&i));
+ EXPECT_TRUE(queue.Insert(&i));
+
+ // Ensure full queue.
+ EXPECT_FALSE(queue.Insert(&i));
+
+ // Empty the queue.
+ queue.Clear();
+
+ // Ensure that the queue is empty
+ EXPECT_FALSE(queue.Remove(&i));
+
+ // Ensure that the queue is no longer full.
+ EXPECT_TRUE(queue.Insert(&i));
+}
+
+TEST(SwapQueueTest, SuccessfulItemVerifyFunction) {
+ std::vector<int> template_element(kChunkSize);
+ SwapQueue<std::vector<int>,
+ SwapQueueItemVerifier<std::vector<int>, LengthVerifierFunction>>
+ queue(2, template_element);
+ std::vector<int> valid_chunk(kChunkSize, 0);
+
+ EXPECT_TRUE(queue.Insert(&valid_chunk));
+ EXPECT_EQ(valid_chunk.size(), kChunkSize);
+ EXPECT_TRUE(queue.Remove(&valid_chunk));
+ EXPECT_EQ(valid_chunk.size(), kChunkSize);
+}
+
+TEST(SwapQueueTest, SuccessfulItemVerifyFunctor) {
+ std::vector<int> template_element(kChunkSize);
+ LengthVerifierFunctor verifier(kChunkSize);
+ SwapQueue<std::vector<int>, LengthVerifierFunctor> queue(2, template_element,
+ verifier);
+ std::vector<int> valid_chunk(kChunkSize, 0);
+
+ EXPECT_TRUE(queue.Insert(&valid_chunk));
+ EXPECT_EQ(valid_chunk.size(), kChunkSize);
+ EXPECT_TRUE(queue.Remove(&valid_chunk));
+ EXPECT_EQ(valid_chunk.size(), kChunkSize);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(SwapQueueTest, UnsuccessfulItemVerifyFunctor) {
+ // Queue item verifier for the test.
+ auto minus_2_verifier = [](const int& i) { return i > -2; };
+ SwapQueue<int, decltype(minus_2_verifier)> queue(2, minus_2_verifier);
+
+ int valid_value = 1;
+ int invalid_value = -4;
+ EXPECT_TRUE(queue.Insert(&valid_value));
+ EXPECT_TRUE(queue.Remove(&valid_value));
+ bool result;
+ EXPECT_DEATH(result = queue.Insert(&invalid_value), "");
+}
+
+TEST(SwapQueueTest, UnSuccessfulItemVerifyInsert) {
+ std::vector<int> template_element(kChunkSize);
+ SwapQueue<std::vector<int>,
+ SwapQueueItemVerifier<std::vector<int>, &LengthVerifierFunction>>
+ queue(2, template_element);
+ std::vector<int> invalid_chunk(kChunkSize - 1, 0);
+ bool result;
+ EXPECT_DEATH(result = queue.Insert(&invalid_chunk), "");
+}
+
+TEST(SwapQueueTest, UnSuccessfulItemVerifyRemove) {
+ std::vector<int> template_element(kChunkSize);
+ SwapQueue<std::vector<int>,
+ SwapQueueItemVerifier<std::vector<int>, &LengthVerifierFunction>>
+ queue(2, template_element);
+ std::vector<int> invalid_chunk(kChunkSize - 1, 0);
+ std::vector<int> valid_chunk(kChunkSize, 0);
+ EXPECT_TRUE(queue.Insert(&valid_chunk));
+ EXPECT_EQ(valid_chunk.size(), kChunkSize);
+ bool result;
+ EXPECT_DEATH(result = queue.Remove(&invalid_chunk), "");
+}
+#endif
+
+TEST(SwapQueueTest, VectorContentTest) {
+ const size_t kQueueSize = 10;
+ const size_t kFrameLength = 160;
+ const size_t kDataLength = kQueueSize * kFrameLength;
+ std::vector<int16_t> buffer_reader(kFrameLength, 0);
+ std::vector<int16_t> buffer_writer(kFrameLength, 0);
+ SwapQueue<std::vector<int16_t>> queue(kQueueSize,
+ std::vector<int16_t>(kFrameLength));
+ std::vector<int16_t> samples(kDataLength);
+
+ for (size_t k = 0; k < kDataLength; k++) {
+ samples[k] = k % 9;
+ }
+
+ for (size_t k = 0; k < kQueueSize; k++) {
+ buffer_writer.clear();
+ buffer_writer.insert(buffer_writer.end(), &samples[0] + k * kFrameLength,
+ &samples[0] + (k + 1) * kFrameLength);
+
+ EXPECT_TRUE(queue.Insert(&buffer_writer));
+ }
+
+ for (size_t k = 0; k < kQueueSize; k++) {
+ EXPECT_TRUE(queue.Remove(&buffer_reader));
+
+ for (size_t j = 0; j < buffer_reader.size(); j++) {
+ EXPECT_EQ(buffer_reader[j], samples[k * kFrameLength + j]);
+ }
+ }
+}
+
+TEST(SwapQueueTest, ZeroSlotQueue) {
+ SwapQueue<int> queue(0);
+ int i = 42;
+ EXPECT_FALSE(queue.Insert(&i));
+ EXPECT_FALSE(queue.Remove(&i));
+ EXPECT_EQ(i, 42);
+}
+
+TEST(SwapQueueTest, OneSlotQueue) {
+ SwapQueue<int> queue(1);
+ int i = 42;
+ EXPECT_TRUE(queue.Insert(&i));
+ i = 43;
+ EXPECT_FALSE(queue.Insert(&i));
+ EXPECT_EQ(i, 43);
+ EXPECT_TRUE(queue.Remove(&i));
+ EXPECT_EQ(i, 42);
+ EXPECT_FALSE(queue.Remove(&i));
+}
+
+} // namespace webrtc
diff --git a/webrtc/common_audio/wav_file.cc b/webrtc/common_audio/wav_file.cc
index 8dae7d6..94b7a3c 100644
--- a/webrtc/common_audio/wav_file.cc
+++ b/webrtc/common_audio/wav_file.cc
@@ -13,6 +13,7 @@
#include <algorithm>
#include <cstdio>
#include <limits>
+#include <sstream>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
@@ -23,7 +24,7 @@
// We write 16-bit PCM WAV files.
static const WavFormat kWavFormat = kWavFormatPcm;
-static const int kBytesPerSample = 2;
+static const size_t kBytesPerSample = 2;
// Doesn't take ownership of the file handle and won't close it.
class ReadableWavFile : public ReadableWav {
@@ -37,13 +38,21 @@
FILE* file_;
};
+std::string WavFile::FormatAsString() const {
+ std::ostringstream s;
+ s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
+ << ", Duration: "
+ << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
+ return s.str();
+}
+
WavReader::WavReader(const std::string& filename)
: file_handle_(fopen(filename.c_str(), "rb")) {
- RTC_CHECK(file_handle_ && "Could not open wav file for reading.");
+ RTC_CHECK(file_handle_) << "Could not open wav file for reading.";
ReadableWavFile readable(file_handle_);
WavFormat format;
- int bytes_per_sample;
+ size_t bytes_per_sample;
RTC_CHECK(ReadWavHeader(&readable, &num_channels_, &sample_rate_, &format,
&bytes_per_sample, &num_samples_));
num_samples_remaining_ = num_samples_;
@@ -60,14 +69,13 @@
#error "Need to convert samples to big-endian when reading from WAV file"
#endif
// There could be metadata after the audio; ensure we don't read it.
- num_samples = std::min(rtc::checked_cast<uint32_t>(num_samples),
- num_samples_remaining_);
+ num_samples = std::min(num_samples, num_samples_remaining_);
const size_t read =
fread(samples, sizeof(*samples), num_samples, file_handle_);
// If we didn't read what was requested, ensure we've reached the EOF.
RTC_CHECK(read == num_samples || feof(file_handle_));
RTC_CHECK_LE(read, num_samples_remaining_);
- num_samples_remaining_ -= rtc::checked_cast<uint32_t>(read);
+ num_samples_remaining_ -= read;
return read;
}
@@ -91,12 +99,12 @@
}
WavWriter::WavWriter(const std::string& filename, int sample_rate,
- int num_channels)
+ size_t num_channels)
: sample_rate_(sample_rate),
num_channels_(num_channels),
num_samples_(0),
file_handle_(fopen(filename.c_str(), "wb")) {
- RTC_CHECK(file_handle_ && "Could not open wav file for writing.");
+ RTC_CHECK(file_handle_) << "Could not open wav file for writing.";
RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, kWavFormat,
kBytesPerSample, num_samples_));
@@ -117,9 +125,8 @@
const size_t written =
fwrite(samples, sizeof(*samples), num_samples, file_handle_);
RTC_CHECK_EQ(num_samples, written);
- num_samples_ += static_cast<uint32_t>(written);
- RTC_CHECK(written <= std::numeric_limits<uint32_t>::max() ||
- num_samples_ >= written); // detect uint32_t overflow
+ num_samples_ += written;
+ RTC_CHECK(num_samples_ >= written); // detect size_t overflow
}
void WavWriter::WriteSamples(const float* samples, size_t num_samples) {
@@ -146,7 +153,7 @@
rtc_WavWriter* rtc_WavOpen(const char* filename,
int sample_rate,
- int num_channels) {
+ size_t num_channels) {
return reinterpret_cast<rtc_WavWriter*>(
new webrtc::WavWriter(filename, sample_rate, num_channels));
}
@@ -165,10 +172,10 @@
return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate();
}
-int rtc_WavNumChannels(const rtc_WavWriter* wf) {
+size_t rtc_WavNumChannels(const rtc_WavWriter* wf) {
return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_channels();
}
-uint32_t rtc_WavNumSamples(const rtc_WavWriter* wf) {
+size_t rtc_WavNumSamples(const rtc_WavWriter* wf) {
return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_samples();
}
diff --git a/webrtc/common_audio/wav_file.h b/webrtc/common_audio/wav_file.h
index 2eadd3f..e656eb8 100644
--- a/webrtc/common_audio/wav_file.h
+++ b/webrtc/common_audio/wav_file.h
@@ -27,8 +27,11 @@
virtual ~WavFile() {}
virtual int sample_rate() const = 0;
- virtual int num_channels() const = 0;
- virtual uint32_t num_samples() const = 0;
+ virtual size_t num_channels() const = 0;
+ virtual size_t num_samples() const = 0;
+
+ // Returns a human-readable string containing the audio format.
+ std::string FormatAsString() const;
};
// Simple C++ class for writing 16-bit PCM WAV files. All error handling is
@@ -36,7 +39,7 @@
class WavWriter final : public WavFile {
public:
// Open a new WAV file for writing.
- WavWriter(const std::string& filename, int sample_rate, int num_channels);
+ WavWriter(const std::string& filename, int sample_rate, size_t num_channels);
// Close the WAV file, after writing its header.
~WavWriter();
@@ -48,14 +51,14 @@
void WriteSamples(const int16_t* samples, size_t num_samples);
int sample_rate() const override { return sample_rate_; }
- int num_channels() const override { return num_channels_; }
- uint32_t num_samples() const override { return num_samples_; }
+ size_t num_channels() const override { return num_channels_; }
+ size_t num_samples() const override { return num_samples_; }
private:
void Close();
const int sample_rate_;
- const int num_channels_;
- uint32_t num_samples_; // Total number of samples written to file.
+ const size_t num_channels_;
+ size_t num_samples_; // Total number of samples written to file.
FILE* file_handle_; // Output file, owned by this class
RTC_DISALLOW_COPY_AND_ASSIGN(WavWriter);
@@ -76,15 +79,15 @@
size_t ReadSamples(size_t num_samples, int16_t* samples);
int sample_rate() const override { return sample_rate_; }
- int num_channels() const override { return num_channels_; }
- uint32_t num_samples() const override { return num_samples_; }
+ size_t num_channels() const override { return num_channels_; }
+ size_t num_samples() const override { return num_samples_; }
private:
void Close();
int sample_rate_;
- int num_channels_;
- uint32_t num_samples_; // Total number of samples in the file.
- uint32_t num_samples_remaining_;
+ size_t num_channels_;
+ size_t num_samples_; // Total number of samples in the file.
+ size_t num_samples_remaining_;
FILE* file_handle_; // Input file, owned by this class.
RTC_DISALLOW_COPY_AND_ASSIGN(WavReader);
@@ -99,14 +102,14 @@
typedef struct rtc_WavWriter rtc_WavWriter;
rtc_WavWriter* rtc_WavOpen(const char* filename,
int sample_rate,
- int num_channels);
+ size_t num_channels);
void rtc_WavClose(rtc_WavWriter* wf);
void rtc_WavWriteSamples(rtc_WavWriter* wf,
const float* samples,
size_t num_samples);
int rtc_WavSampleRate(const rtc_WavWriter* wf);
-int rtc_WavNumChannels(const rtc_WavWriter* wf);
-uint32_t rtc_WavNumSamples(const rtc_WavWriter* wf);
+size_t rtc_WavNumChannels(const rtc_WavWriter* wf);
+size_t rtc_WavNumSamples(const rtc_WavWriter* wf);
#ifdef __cplusplus
} // extern "C"
diff --git a/webrtc/common_audio/wav_file_unittest.cc b/webrtc/common_audio/wav_file_unittest.cc
index 78b0a34..ba1db1c 100644
--- a/webrtc/common_audio/wav_file_unittest.cc
+++ b/webrtc/common_audio/wav_file_unittest.cc
@@ -26,11 +26,11 @@
// Write a tiny WAV file with the C++ interface and verify the result.
TEST(WavWriterTest, CPP) {
const std::string outfile = test::OutputPath() + "wavtest1.wav";
- static const uint32_t kNumSamples = 3;
+ static const size_t kNumSamples = 3;
{
WavWriter w(outfile, 14099, 1);
EXPECT_EQ(14099, w.sample_rate());
- EXPECT_EQ(1, w.num_channels());
+ EXPECT_EQ(1u, w.num_channels());
EXPECT_EQ(0u, w.num_samples());
w.WriteSamples(kSamples, kNumSamples);
EXPECT_EQ(kNumSamples, w.num_samples());
@@ -64,10 +64,10 @@
0xff, 0x7f, // third sample: 4e4 (saturated)
kMetadata[0], kMetadata[1],
};
- static const int kContentSize =
+ static const size_t kContentSize =
kWavHeaderSize + kNumSamples * sizeof(int16_t) + sizeof(kMetadata);
static_assert(sizeof(kExpectedContents) == kContentSize, "content size");
- EXPECT_EQ(size_t(kContentSize), test::GetFileSize(outfile));
+ EXPECT_EQ(kContentSize, test::GetFileSize(outfile));
FILE* f = fopen(outfile.c_str(), "rb");
ASSERT_TRUE(f);
uint8_t contents[kContentSize];
@@ -78,7 +78,7 @@
{
WavReader r(outfile);
EXPECT_EQ(14099, r.sample_rate());
- EXPECT_EQ(1, r.num_channels());
+ EXPECT_EQ(1u, r.num_channels());
EXPECT_EQ(kNumSamples, r.num_samples());
static const float kTruncatedSamples[] = {0.0, 10.0, 32767.0};
float samples[kNumSamples];
@@ -93,9 +93,9 @@
const std::string outfile = test::OutputPath() + "wavtest2.wav";
rtc_WavWriter* w = rtc_WavOpen(outfile.c_str(), 11904, 2);
EXPECT_EQ(11904, rtc_WavSampleRate(w));
- EXPECT_EQ(2, rtc_WavNumChannels(w));
+ EXPECT_EQ(2u, rtc_WavNumChannels(w));
EXPECT_EQ(0u, rtc_WavNumSamples(w));
- static const uint32_t kNumSamples = 4;
+ static const size_t kNumSamples = 4;
rtc_WavWriteSamples(w, &kSamples[0], 2);
EXPECT_EQ(2u, rtc_WavNumSamples(w));
rtc_WavWriteSamples(w, &kSamples[2], kNumSamples - 2);
@@ -120,10 +120,10 @@
0xff, 0x7f, // third sample: 4e4 (saturated)
0, 0x80, // fourth sample: -1e9 (saturated)
};
- static const int kContentSize =
+ static const size_t kContentSize =
kWavHeaderSize + kNumSamples * sizeof(int16_t);
static_assert(sizeof(kExpectedContents) == kContentSize, "content size");
- EXPECT_EQ(size_t(kContentSize), test::GetFileSize(outfile));
+ EXPECT_EQ(kContentSize, test::GetFileSize(outfile));
FILE* f = fopen(outfile.c_str(), "rb");
ASSERT_TRUE(f);
uint8_t contents[kContentSize];
@@ -136,10 +136,10 @@
TEST(WavWriterTest, LargeFile) {
std::string outfile = test::OutputPath() + "wavtest3.wav";
static const int kSampleRate = 8000;
- static const int kNumChannels = 2;
- static const uint32_t kNumSamples = 3 * kSampleRate * kNumChannels;
+ static const size_t kNumChannels = 2;
+ static const size_t kNumSamples = 3 * kSampleRate * kNumChannels;
float samples[kNumSamples];
- for (uint32_t i = 0; i < kNumSamples; i += kNumChannels) {
+ for (size_t i = 0; i < kNumSamples; i += kNumChannels) {
// A nice periodic beeping sound.
static const double kToneHz = 440;
const double t = static_cast<double>(i) / (kNumChannels * kSampleRate);
diff --git a/webrtc/common_audio/wav_header.cc b/webrtc/common_audio/wav_header.cc
index 61cfffe..402ea17 100644
--- a/webrtc/common_audio/wav_header.cc
+++ b/webrtc/common_audio/wav_header.cc
@@ -59,20 +59,19 @@
} // namespace
-bool CheckWavParameters(int num_channels,
+bool CheckWavParameters(size_t num_channels,
int sample_rate,
WavFormat format,
- int bytes_per_sample,
- uint32_t num_samples) {
+ size_t bytes_per_sample,
+ size_t num_samples) {
// num_channels, sample_rate, and bytes_per_sample must be positive, must fit
// in their respective fields, and their product must fit in the 32-bit
// ByteRate field.
- if (num_channels <= 0 || sample_rate <= 0 || bytes_per_sample <= 0)
+ if (num_channels == 0 || sample_rate <= 0 || bytes_per_sample == 0)
return false;
if (static_cast<uint64_t>(sample_rate) > std::numeric_limits<uint32_t>::max())
return false;
- if (static_cast<uint64_t>(num_channels) >
- std::numeric_limits<uint16_t>::max())
+ if (num_channels > std::numeric_limits<uint16_t>::max())
return false;
if (static_cast<uint64_t>(bytes_per_sample) * 8 >
std::numeric_limits<uint16_t>::max())
@@ -99,10 +98,9 @@
// The number of bytes in the file, not counting the first ChunkHeader, must
// be less than 2^32; otherwise, the ChunkSize field overflows.
- const uint32_t max_samples =
- (std::numeric_limits<uint32_t>::max()
- - (kWavHeaderSize - sizeof(ChunkHeader))) /
- bytes_per_sample;
+ const size_t header_size = kWavHeaderSize - sizeof(ChunkHeader);
+ const size_t max_samples =
+ (std::numeric_limits<uint32_t>::max() - header_size) / bytes_per_sample;
if (num_samples > max_samples)
return false;
@@ -132,30 +130,32 @@
#error "Write be-to-le conversion functions"
#endif
-static inline uint32_t RiffChunkSize(uint32_t bytes_in_payload) {
- return bytes_in_payload + kWavHeaderSize - sizeof(ChunkHeader);
+static inline uint32_t RiffChunkSize(size_t bytes_in_payload) {
+ return static_cast<uint32_t>(
+ bytes_in_payload + kWavHeaderSize - sizeof(ChunkHeader));
}
-static inline uint32_t ByteRate(int num_channels, int sample_rate,
- int bytes_per_sample) {
- return static_cast<uint32_t>(num_channels) * sample_rate * bytes_per_sample;
+static inline uint32_t ByteRate(size_t num_channels, int sample_rate,
+ size_t bytes_per_sample) {
+ return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
}
-static inline uint16_t BlockAlign(int num_channels, int bytes_per_sample) {
- return num_channels * bytes_per_sample;
+static inline uint16_t BlockAlign(size_t num_channels,
+ size_t bytes_per_sample) {
+ return static_cast<uint16_t>(num_channels * bytes_per_sample);
}
void WriteWavHeader(uint8_t* buf,
- int num_channels,
+ size_t num_channels,
int sample_rate,
WavFormat format,
- int bytes_per_sample,
- uint32_t num_samples) {
+ size_t bytes_per_sample,
+ size_t num_samples) {
RTC_CHECK(CheckWavParameters(num_channels, sample_rate, format,
bytes_per_sample, num_samples));
WavHeader header;
- const uint32_t bytes_in_payload = bytes_per_sample * num_samples;
+ const size_t bytes_in_payload = bytes_per_sample * num_samples;
WriteFourCC(&header.riff.header.ID, 'R', 'I', 'F', 'F');
WriteLE32(&header.riff.header.Size, RiffChunkSize(bytes_in_payload));
@@ -164,15 +164,16 @@
WriteFourCC(&header.fmt.header.ID, 'f', 'm', 't', ' ');
WriteLE32(&header.fmt.header.Size, kFmtSubchunkSize);
WriteLE16(&header.fmt.AudioFormat, format);
- WriteLE16(&header.fmt.NumChannels, num_channels);
+ WriteLE16(&header.fmt.NumChannels, static_cast<uint16_t>(num_channels));
WriteLE32(&header.fmt.SampleRate, sample_rate);
WriteLE32(&header.fmt.ByteRate, ByteRate(num_channels, sample_rate,
bytes_per_sample));
WriteLE16(&header.fmt.BlockAlign, BlockAlign(num_channels, bytes_per_sample));
- WriteLE16(&header.fmt.BitsPerSample, 8 * bytes_per_sample);
+ WriteLE16(&header.fmt.BitsPerSample,
+ static_cast<uint16_t>(8 * bytes_per_sample));
WriteFourCC(&header.data.header.ID, 'd', 'a', 't', 'a');
- WriteLE32(&header.data.header.Size, bytes_in_payload);
+ WriteLE32(&header.data.header.Size, static_cast<uint32_t>(bytes_in_payload));
// Do an extra copy rather than writing everything to buf directly, since buf
// might not be correctly aligned.
@@ -180,11 +181,11 @@
}
bool ReadWavHeader(ReadableWav* readable,
- int* num_channels,
+ size_t* num_channels,
int* sample_rate,
WavFormat* format,
- int* bytes_per_sample,
- uint32_t* num_samples) {
+ size_t* bytes_per_sample,
+ size_t* num_samples) {
WavHeader header;
if (readable->Read(&header, kWavHeaderSize - sizeof(header.data)) !=
kWavHeaderSize - sizeof(header.data))
@@ -210,8 +211,8 @@
*num_channels = ReadLE16(header.fmt.NumChannels);
*sample_rate = ReadLE32(header.fmt.SampleRate);
*bytes_per_sample = ReadLE16(header.fmt.BitsPerSample) / 8;
- const uint32_t bytes_in_payload = ReadLE32(header.data.header.Size);
- if (*bytes_per_sample <= 0)
+ const size_t bytes_in_payload = ReadLE32(header.data.header.Size);
+ if (*bytes_per_sample == 0)
return false;
*num_samples = bytes_in_payload / *bytes_per_sample;
diff --git a/webrtc/common_audio/wav_header.h b/webrtc/common_audio/wav_header.h
index 1a0fd7c..6844306 100644
--- a/webrtc/common_audio/wav_header.h
+++ b/webrtc/common_audio/wav_header.h
@@ -32,32 +32,32 @@
};
// Return true if the given parameters will make a well-formed WAV header.
-bool CheckWavParameters(int num_channels,
+bool CheckWavParameters(size_t num_channels,
int sample_rate,
WavFormat format,
- int bytes_per_sample,
- uint32_t num_samples);
+ size_t bytes_per_sample,
+ size_t num_samples);
// Write a kWavHeaderSize bytes long WAV header to buf. The payload that
// follows the header is supposed to have the specified number of interleaved
// channels and contain the specified total number of samples of the specified
// type. CHECKs the input parameters for validity.
void WriteWavHeader(uint8_t* buf,
- int num_channels,
+ size_t num_channels,
int sample_rate,
WavFormat format,
- int bytes_per_sample,
- uint32_t num_samples);
+ size_t bytes_per_sample,
+ size_t num_samples);
// Read a WAV header from an implemented ReadableWav and parse the values into
// the provided output parameters. ReadableWav is used because the header can
// be variably sized. Returns false if the header is invalid.
bool ReadWavHeader(ReadableWav* readable,
- int* num_channels,
+ size_t* num_channels,
int* sample_rate,
WavFormat* format,
- int* bytes_per_sample,
- uint32_t* num_samples);
+ size_t* bytes_per_sample,
+ size_t* num_samples);
} // namespace webrtc
diff --git a/webrtc/common_audio/wav_header_unittest.cc b/webrtc/common_audio/wav_header_unittest.cc
index e03cb30..8527939 100644
--- a/webrtc/common_audio/wav_header_unittest.cc
+++ b/webrtc/common_audio/wav_header_unittest.cc
@@ -70,7 +70,7 @@
// Try some really stupid values for one parameter at a time.
EXPECT_TRUE(CheckWavParameters(1, 8000, kWavFormatPcm, 1, 0));
EXPECT_FALSE(CheckWavParameters(0, 8000, kWavFormatPcm, 1, 0));
- EXPECT_FALSE(CheckWavParameters(-1, 8000, kWavFormatPcm, 1, 0));
+ EXPECT_FALSE(CheckWavParameters(0x10000, 8000, kWavFormatPcm, 1, 0));
EXPECT_FALSE(CheckWavParameters(1, 0, kWavFormatPcm, 1, 0));
EXPECT_FALSE(CheckWavParameters(1, 8000, WavFormat(0), 1, 0));
EXPECT_FALSE(CheckWavParameters(1, 8000, kWavFormatPcm, 0, 0));
@@ -91,11 +91,11 @@
}
TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
- int num_channels = 0;
+ size_t num_channels = 0;
int sample_rate = 0;
WavFormat format = kWavFormatPcm;
- int bytes_per_sample = 0;
- uint32_t num_samples = 0;
+ size_t bytes_per_sample = 0;
+ size_t num_samples = 0;
// Test a few ways the header can be invalid. We start with the valid header
// used in WriteAndReadWavHeader, and invalidate one field per test. The
@@ -268,19 +268,19 @@
static_assert(sizeof(kExpectedBuf) == kSize, "buffer size");
EXPECT_EQ(0, memcmp(kExpectedBuf, buf, kSize));
- int num_channels = 0;
+ size_t num_channels = 0;
int sample_rate = 0;
WavFormat format = kWavFormatPcm;
- int bytes_per_sample = 0;
- uint32_t num_samples = 0;
+ size_t bytes_per_sample = 0;
+ size_t num_samples = 0;
ReadableWavBuffer r(buf + 4, sizeof(buf) - 8);
EXPECT_TRUE(
ReadWavHeader(&r, &num_channels, &sample_rate, &format,
&bytes_per_sample, &num_samples));
- EXPECT_EQ(17, num_channels);
+ EXPECT_EQ(17u, num_channels);
EXPECT_EQ(12345, sample_rate);
EXPECT_EQ(kWavFormatALaw, format);
- EXPECT_EQ(1, bytes_per_sample);
+ EXPECT_EQ(1u, bytes_per_sample);
EXPECT_EQ(123457689u, num_samples);
}
@@ -304,19 +304,19 @@
0x99, 0xd0, 0x5b, 0x07, // size of payload: 123457689
};
- int num_channels = 0;
+ size_t num_channels = 0;
int sample_rate = 0;
WavFormat format = kWavFormatPcm;
- int bytes_per_sample = 0;
- uint32_t num_samples = 0;
+ size_t bytes_per_sample = 0;
+ size_t num_samples = 0;
ReadableWavBuffer r(kBuf, sizeof(kBuf));
EXPECT_TRUE(
ReadWavHeader(&r, &num_channels, &sample_rate, &format,
&bytes_per_sample, &num_samples));
- EXPECT_EQ(17, num_channels);
+ EXPECT_EQ(17u, num_channels);
EXPECT_EQ(12345, sample_rate);
EXPECT_EQ(kWavFormatALaw, format);
- EXPECT_EQ(1, bytes_per_sample);
+ EXPECT_EQ(1u, bytes_per_sample);
EXPECT_EQ(123457689u, num_samples);
}
diff --git a/webrtc/common_types.h b/webrtc/common_types.h
index 07faf6a..444ef92 100644
--- a/webrtc/common_types.h
+++ b/webrtc/common_types.h
@@ -291,7 +291,7 @@
char plname[RTP_PAYLOAD_NAME_SIZE];
int plfreq;
int pacsize;
- int channels;
+ size_t channels;
int rate; // bits/sec unlike {start,min,max}Bitrate elsewhere in this file!
bool operator==(const CodecInst& other) const {
@@ -311,12 +311,6 @@
// RTP
enum {kRtpCsrcSize = 15}; // RFC 3550 page 13
-enum RTPDirections
-{
- kRtpIncoming = 0,
- kRtpOutgoing
-};
-
enum PayloadFrequencies
{
kFreq8000Hz = 8000,
@@ -547,6 +541,7 @@
enum { kConfigParameterSize = 128};
enum { kPayloadNameSize = 32};
enum { kMaxSimulcastStreams = 4};
+enum { kMaxSpatialLayers = 5 };
enum { kMaxTemporalStreams = 4};
enum VideoCodecComplexity
@@ -676,6 +671,13 @@
}
};
+struct SpatialLayer {
+ int scaling_factor_num;
+ int scaling_factor_den;
+ int target_bitrate_bps;
+ // TODO(ivica): Add max_quantizer and min_quantizer?
+};
+
enum VideoCodecMode {
kRealtimeVideo,
kScreensharing
@@ -702,6 +704,7 @@
unsigned int qpMax;
unsigned char numberOfSimulcastStreams;
SimulcastStream simulcastStream[kMaxSimulcastStreams];
+ SpatialLayer spatialLayers[kMaxSpatialLayers];
VideoCodecMode mode;
diff --git a/webrtc/common_video/BUILD.gn b/webrtc/common_video/BUILD.gn
index 473c757..4ef968d 100644
--- a/webrtc/common_video/BUILD.gn
+++ b/webrtc/common_video/BUILD.gn
@@ -10,7 +10,7 @@
config("common_video_config") {
include_dirs = [
- "interface",
+ "include",
"libyuv/include",
]
}
@@ -18,10 +18,10 @@
source_set("common_video") {
sources = [
"i420_buffer_pool.cc",
+ "include/i420_buffer_pool.h",
+ "include/incoming_video_stream.h",
+ "include/video_frame_buffer.h",
"incoming_video_stream.cc",
- "interface/i420_buffer_pool.h",
- "interface/incoming_video_stream.h",
- "interface/video_frame_buffer.h",
"libyuv/include/scaler.h",
"libyuv/include/webrtc_libyuv.h",
"libyuv/scaler.cc",
diff --git a/webrtc/common_video/common_video.gyp b/webrtc/common_video/common_video.gyp
index 5c0ecb8..fe14da1 100644
--- a/webrtc/common_video/common_video.gyp
+++ b/webrtc/common_video/common_video.gyp
@@ -14,7 +14,7 @@
'type': 'static_library',
'include_dirs': [
'<(webrtc_root)/modules/interface/',
- 'interface',
+ 'include',
'libyuv/include',
],
'dependencies': [
@@ -23,7 +23,7 @@
],
'direct_dependent_settings': {
'include_dirs': [
- 'interface',
+ 'include',
'libyuv/include',
],
},
@@ -42,9 +42,9 @@
'i420_buffer_pool.cc',
'video_frame.cc',
'incoming_video_stream.cc',
- 'interface/i420_buffer_pool.h',
- 'interface/incoming_video_stream.h',
- 'interface/video_frame_buffer.h',
+ 'include/i420_buffer_pool.h',
+ 'include/incoming_video_stream.h',
+ 'include/video_frame_buffer.h',
'libyuv/include/scaler.h',
'libyuv/include/webrtc_libyuv.h',
'libyuv/scaler.cc',
diff --git a/webrtc/common_video/common_video_unittests.gyp b/webrtc/common_video/common_video_unittests.gyp
index beeab5d..b5e892c 100644
--- a/webrtc/common_video/common_video_unittests.gyp
+++ b/webrtc/common_video/common_video_unittests.gyp
@@ -17,6 +17,7 @@
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support_main',
+ '<(webrtc_root)/test/test.gyp:fake_video_frames',
],
'sources': [
'i420_buffer_pool_unittest.cc',
diff --git a/webrtc/common_video/i420_buffer_pool.cc b/webrtc/common_video/i420_buffer_pool.cc
index c746666..98daec9 100644
--- a/webrtc/common_video/i420_buffer_pool.cc
+++ b/webrtc/common_video/i420_buffer_pool.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/common_video/interface/i420_buffer_pool.h"
+#include "webrtc/common_video/include/i420_buffer_pool.h"
#include "webrtc/base/checks.h"
diff --git a/webrtc/common_video/i420_buffer_pool_unittest.cc b/webrtc/common_video/i420_buffer_pool_unittest.cc
index a1596eb..b030ee7 100644
--- a/webrtc/common_video/i420_buffer_pool_unittest.cc
+++ b/webrtc/common_video/i420_buffer_pool_unittest.cc
@@ -11,7 +11,7 @@
#include <string>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common_video/interface/i420_buffer_pool.h"
+#include "webrtc/common_video/include/i420_buffer_pool.h"
namespace webrtc {
diff --git a/webrtc/common_video/i420_video_frame_unittest.cc b/webrtc/common_video/i420_video_frame_unittest.cc
index da3996b..1ec451c 100644
--- a/webrtc/common_video/i420_video_frame_unittest.cc
+++ b/webrtc/common_video/i420_video_frame_unittest.cc
@@ -8,8 +8,6 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/video_frame.h"
-
#include <math.h>
#include <string.h>
@@ -17,6 +15,7 @@
#include "webrtc/base/bind.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/test/fake_texture_frame.h"
+#include "webrtc/video_frame.h"
namespace webrtc {
@@ -25,7 +24,6 @@
int stride,
int width,
int height);
-bool EqualFrames(const VideoFrame& frame1, const VideoFrame& frame2);
int ExpectedSize(int plane_stride, int image_height, PlaneType type);
TEST(TestVideoFrame, InitialValues) {
@@ -103,7 +101,7 @@
stride_u, stride_v, kRotation));
// Frame of smaller dimensions.
EXPECT_EQ(0, small_frame.CopyFrame(big_frame));
- EXPECT_TRUE(EqualFrames(small_frame, big_frame));
+ EXPECT_TRUE(small_frame.EqualsFrame(big_frame));
EXPECT_EQ(kRotation, small_frame.rotation());
// Frame of larger dimensions.
@@ -113,7 +111,7 @@
memset(small_frame.buffer(kUPlane), 2, small_frame.allocated_size(kUPlane));
memset(small_frame.buffer(kVPlane), 3, small_frame.allocated_size(kVPlane));
EXPECT_EQ(0, big_frame.CopyFrame(small_frame));
- EXPECT_TRUE(EqualFrames(small_frame, big_frame));
+ EXPECT_TRUE(small_frame.EqualsFrame(big_frame));
}
TEST(TestVideoFrame, ShallowCopy) {
@@ -174,7 +172,7 @@
TEST(TestVideoFrame, Reset) {
VideoFrame frame;
- ASSERT_TRUE(frame.CreateEmptyFrame(5, 5, 5, 5, 5) == 0);
+ ASSERT_EQ(frame.CreateEmptyFrame(5, 5, 5, 5, 5), 0);
frame.set_ntp_time_ms(1);
frame.set_timestamp(2);
frame.set_render_time_ms(3);
@@ -244,7 +242,7 @@
TEST(TestVideoFrame, TextureInitialValues) {
test::FakeNativeHandle* handle = new test::FakeNativeHandle();
- VideoFrame frame = test::CreateFakeNativeHandleFrame(
+ VideoFrame frame = test::FakeNativeHandle::CreateFrame(
handle, 640, 480, 100, 10, webrtc::kVideoRotation_0);
EXPECT_EQ(640, frame.width());
EXPECT_EQ(480, frame.height());
@@ -258,48 +256,4 @@
EXPECT_EQ(20, frame.render_time_ms());
}
-bool EqualPlane(const uint8_t* data1,
- const uint8_t* data2,
- int stride,
- int width,
- int height) {
- for (int y = 0; y < height; ++y) {
- if (memcmp(data1, data2, width) != 0)
- return false;
- data1 += stride;
- data2 += stride;
- }
- return true;
-}
-
-bool EqualFrames(const VideoFrame& frame1, const VideoFrame& frame2) {
- if ((frame1.width() != frame2.width()) ||
- (frame1.height() != frame2.height()) ||
- (frame1.stride(kYPlane) != frame2.stride(kYPlane)) ||
- (frame1.stride(kUPlane) != frame2.stride(kUPlane)) ||
- (frame1.stride(kVPlane) != frame2.stride(kVPlane)) ||
- (frame1.timestamp() != frame2.timestamp()) ||
- (frame1.ntp_time_ms() != frame2.ntp_time_ms()) ||
- (frame1.render_time_ms() != frame2.render_time_ms())) {
- return false;
- }
- const int half_width = (frame1.width() + 1) / 2;
- const int half_height = (frame1.height() + 1) / 2;
- return EqualPlane(frame1.buffer(kYPlane), frame2.buffer(kYPlane),
- frame1.stride(kYPlane), frame1.width(), frame1.height()) &&
- EqualPlane(frame1.buffer(kUPlane), frame2.buffer(kUPlane),
- frame1.stride(kUPlane), half_width, half_height) &&
- EqualPlane(frame1.buffer(kVPlane), frame2.buffer(kVPlane),
- frame1.stride(kVPlane), half_width, half_height);
-}
-
-int ExpectedSize(int plane_stride, int image_height, PlaneType type) {
- if (type == kYPlane) {
- return (plane_stride * image_height);
- } else {
- int half_height = (image_height + 1) / 2;
- return (plane_stride * half_height);
- }
-}
-
} // namespace webrtc
diff --git a/webrtc/common_video/include/i420_buffer_pool.h b/webrtc/common_video/include/i420_buffer_pool.h
new file mode 100644
index 0000000..5ab1510
--- /dev/null
+++ b/webrtc/common_video/include/i420_buffer_pool.h
@@ -0,0 +1,43 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
+#define WEBRTC_COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
+
+#include <list>
+
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
+
+namespace webrtc {
+
+// Simple buffer pool to avoid unnecessary allocations of I420Buffer objects.
+// The pool manages the memory of the I420Buffer returned from CreateBuffer.
+// When the I420Buffer is destructed, the memory is returned to the pool for use
+// by subsequent calls to CreateBuffer. If the resolution passed to CreateBuffer
+// changes, old buffers will be purged from the pool.
+class I420BufferPool {
+ public:
+ I420BufferPool();
+ // Returns a buffer from the pool, or creates a new buffer if no suitable
+ // buffer exists in the pool.
+ rtc::scoped_refptr<VideoFrameBuffer> CreateBuffer(int width, int height);
+ // Clears buffers_ and detaches the thread checker so that it can be reused
+ // later from another thread.
+ void Release();
+
+ private:
+ rtc::ThreadChecker thread_checker_;
+ std::list<rtc::scoped_refptr<I420Buffer>> buffers_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
diff --git a/webrtc/common_video/include/incoming_video_stream.h b/webrtc/common_video/include/incoming_video_stream.h
new file mode 100644
index 0000000..e3147eb
--- /dev/null
+++ b/webrtc/common_video/include/incoming_video_stream.h
@@ -0,0 +1,107 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
+#define WEBRTC_COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
+
+#include "webrtc/base/platform_thread.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_annotations.h"
+#include "webrtc/common_video/video_render_frames.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class EventTimerWrapper;
+
+class VideoRenderCallback {
+ public:
+ virtual int32_t RenderFrame(const uint32_t streamId,
+ const VideoFrame& videoFrame) = 0;
+
+ protected:
+ virtual ~VideoRenderCallback() {}
+};
+
+class IncomingVideoStream : public VideoRenderCallback {
+ public:
+ IncomingVideoStream(uint32_t stream_id, bool disable_prerenderer_smoothing);
+ ~IncomingVideoStream();
+
+ // Get callback to deliver frames to the module.
+ VideoRenderCallback* ModuleCallback();
+ virtual int32_t RenderFrame(const uint32_t stream_id,
+ const VideoFrame& video_frame);
+
+ // Set callback to the platform dependent code.
+ void SetRenderCallback(VideoRenderCallback* render_callback);
+
+ // Callback for file recording, snapshot, ...
+ void SetExternalCallback(VideoRenderCallback* render_object);
+
+ // Start/Stop.
+ int32_t Start();
+ int32_t Stop();
+
+ // Clear all buffers.
+ int32_t Reset();
+
+ // Properties.
+ uint32_t StreamId() const;
+ uint32_t IncomingRate() const;
+
+ int32_t SetStartImage(const VideoFrame& video_frame);
+
+ int32_t SetTimeoutImage(const VideoFrame& video_frame,
+ const uint32_t timeout);
+
+ int32_t SetExpectedRenderDelay(int32_t delay_ms);
+
+ protected:
+ static bool IncomingVideoStreamThreadFun(void* obj);
+ bool IncomingVideoStreamProcess();
+
+ private:
+ enum { kEventStartupTimeMs = 10 };
+ enum { kEventMaxWaitTimeMs = 100 };
+ enum { kFrameRatePeriodMs = 1000 };
+
+ void DeliverFrame(const VideoFrame& video_frame);
+
+ uint32_t const stream_id_;
+ const bool disable_prerenderer_smoothing_;
+ // Critsects in allowed to enter order.
+ const rtc::scoped_ptr<CriticalSectionWrapper> stream_critsect_;
+ const rtc::scoped_ptr<CriticalSectionWrapper> thread_critsect_;
+ const rtc::scoped_ptr<CriticalSectionWrapper> buffer_critsect_;
+ // TODO(pbos): Make plain member and stop resetting this thread, just
+ // start/stoping it is enough.
+ rtc::scoped_ptr<rtc::PlatformThread> incoming_render_thread_
+ GUARDED_BY(thread_critsect_);
+ rtc::scoped_ptr<EventTimerWrapper> deliver_buffer_event_;
+
+ bool running_ GUARDED_BY(stream_critsect_);
+ VideoRenderCallback* external_callback_ GUARDED_BY(thread_critsect_);
+ VideoRenderCallback* render_callback_ GUARDED_BY(thread_critsect_);
+ const rtc::scoped_ptr<VideoRenderFrames> render_buffers_
+ GUARDED_BY(buffer_critsect_);
+
+ uint32_t incoming_rate_ GUARDED_BY(stream_critsect_);
+ int64_t last_rate_calculation_time_ms_ GUARDED_BY(stream_critsect_);
+ uint16_t num_frames_since_last_calculation_ GUARDED_BY(stream_critsect_);
+ int64_t last_render_time_ms_ GUARDED_BY(thread_critsect_);
+ VideoFrame temp_frame_ GUARDED_BY(thread_critsect_);
+ VideoFrame start_image_ GUARDED_BY(thread_critsect_);
+ VideoFrame timeout_image_ GUARDED_BY(thread_critsect_);
+ uint32_t timeout_time_ GUARDED_BY(thread_critsect_);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
diff --git a/webrtc/common_video/include/video_frame_buffer.h b/webrtc/common_video/include/video_frame_buffer.h
new file mode 100644
index 0000000..710d286
--- /dev/null
+++ b/webrtc/common_video/include/video_frame_buffer.h
@@ -0,0 +1,157 @@
+/*
+ * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
+#define WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
+
+#include "webrtc/base/callback.h"
+#include "webrtc/base/refcount.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/scoped_ref_ptr.h"
+#include "webrtc/system_wrappers/include/aligned_malloc.h"
+
+namespace webrtc {
+
+enum PlaneType {
+ kYPlane = 0,
+ kUPlane = 1,
+ kVPlane = 2,
+ kNumOfPlanes = 3,
+};
+
+// Interface of a simple frame buffer containing pixel data. This interface does
+// not contain any frame metadata such as rotation, timestamp, pixel_width, etc.
+class VideoFrameBuffer : public rtc::RefCountInterface {
+ public:
+ // Returns true if this buffer has a single exclusive owner.
+ virtual bool HasOneRef() const = 0;
+
+ // The resolution of the frame in pixels. For formats where some planes are
+ // subsampled, this is the highest-resolution plane.
+ virtual int width() const = 0;
+ virtual int height() const = 0;
+
+ // Returns pointer to the pixel data for a given plane. The memory is owned by
+ // the VideoFrameBuffer object and must not be freed by the caller.
+ virtual const uint8_t* data(PlaneType type) const = 0;
+
+ // Non-const data access is disallowed by default. You need to make sure you
+ // have exclusive access and a writable buffer before calling this function.
+ virtual uint8_t* MutableData(PlaneType type);
+
+ // Returns the number of bytes between successive rows for a given plane.
+ virtual int stride(PlaneType type) const = 0;
+
+ // Return the handle of the underlying video frame. This is used when the
+ // frame is backed by a texture.
+ virtual void* native_handle() const = 0;
+
+ // Returns a new memory-backed frame buffer converted from this buffer's
+ // native handle.
+ virtual rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() = 0;
+
+ protected:
+ virtual ~VideoFrameBuffer();
+};
+
+// Plain I420 buffer in standard memory.
+class I420Buffer : public VideoFrameBuffer {
+ public:
+ I420Buffer(int width, int height);
+ I420Buffer(int width, int height, int stride_y, int stride_u, int stride_v);
+
+ int width() const override;
+ int height() const override;
+ const uint8_t* data(PlaneType type) const override;
+ // Non-const data access is only allowed if HasOneRef() is true to protect
+ // against unexpected overwrites.
+ uint8_t* MutableData(PlaneType type) override;
+ int stride(PlaneType type) const override;
+ void* native_handle() const override;
+ rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override;
+
+ protected:
+ ~I420Buffer() override;
+
+ private:
+ const int width_;
+ const int height_;
+ const int stride_y_;
+ const int stride_u_;
+ const int stride_v_;
+ const rtc::scoped_ptr<uint8_t, AlignedFreeDeleter> data_;
+};
+
+// Base class for native-handle buffer is a wrapper around a |native_handle|.
+// This is used for convenience as most native-handle implementations can share
+// many VideoFrame implementations, but need to implement a few others (such
+// as their own destructors or conversion methods back to software I420).
+class NativeHandleBuffer : public VideoFrameBuffer {
+ public:
+ NativeHandleBuffer(void* native_handle, int width, int height);
+
+ int width() const override;
+ int height() const override;
+ const uint8_t* data(PlaneType type) const override;
+ int stride(PlaneType type) const override;
+ void* native_handle() const override;
+
+ protected:
+ void* native_handle_;
+ const int width_;
+ const int height_;
+};
+
+class WrappedI420Buffer : public webrtc::VideoFrameBuffer {
+ public:
+ WrappedI420Buffer(int width,
+ int height,
+ const uint8_t* y_plane,
+ int y_stride,
+ const uint8_t* u_plane,
+ int u_stride,
+ const uint8_t* v_plane,
+ int v_stride,
+ const rtc::Callback0<void>& no_longer_used);
+ int width() const override;
+ int height() const override;
+
+ const uint8_t* data(PlaneType type) const override;
+
+ int stride(PlaneType type) const override;
+ void* native_handle() const override;
+
+ rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override;
+
+ private:
+ friend class rtc::RefCountedObject<WrappedI420Buffer>;
+ ~WrappedI420Buffer() override;
+
+ const int width_;
+ const int height_;
+ const uint8_t* const y_plane_;
+ const uint8_t* const u_plane_;
+ const uint8_t* const v_plane_;
+ const int y_stride_;
+ const int u_stride_;
+ const int v_stride_;
+ rtc::Callback0<void> no_longer_used_cb_;
+};
+
+// Helper function to crop |buffer| without making a deep copy. May only be used
+// for non-native frames.
+rtc::scoped_refptr<VideoFrameBuffer> ShallowCenterCrop(
+ const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
+ int cropped_width,
+ int cropped_height);
+
+} // namespace webrtc
+
+#endif // WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
diff --git a/webrtc/common_video/include/video_image.h b/webrtc/common_video/include/video_image.h
new file mode 100644
index 0000000..4a6e451
--- /dev/null
+++ b/webrtc/common_video/include/video_image.h
@@ -0,0 +1,17 @@
+/*
+ * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_IMAGE_H_
+#define WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_IMAGE_H_
+
+// TODO(pbos): Remove this file and include webrtc/video_frame.h instead.
+#include "webrtc/video_frame.h"
+
+#endif // WEBRTC_COMMON_VIDEO_INCLUDE_VIDEO_IMAGE_H_
diff --git a/webrtc/common_video/incoming_video_stream.cc b/webrtc/common_video/incoming_video_stream.cc
index 79bbb8a..1272ecc 100644
--- a/webrtc/common_video/incoming_video_stream.cc
+++ b/webrtc/common_video/incoming_video_stream.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/common_video/interface/incoming_video_stream.h"
+#include "webrtc/common_video/include/incoming_video_stream.h"
#include <assert.h>
@@ -21,18 +21,21 @@
#include <sys/time.h>
#endif
+#include "webrtc/base/platform_thread.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/common_video/video_render_frames.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/thread_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"
+#include "webrtc/video_renderer.h"
namespace webrtc {
-IncomingVideoStream::IncomingVideoStream(uint32_t stream_id)
+IncomingVideoStream::IncomingVideoStream(uint32_t stream_id,
+ bool disable_prerenderer_smoothing)
: stream_id_(stream_id),
+ disable_prerenderer_smoothing_(disable_prerenderer_smoothing),
stream_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
thread_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
buffer_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
@@ -49,8 +52,7 @@
temp_frame_(),
start_image_(),
timeout_image_(),
- timeout_time_() {
-}
+ timeout_time_() {}
IncomingVideoStream::~IncomingVideoStream() {
Stop();
@@ -80,11 +82,15 @@
last_rate_calculation_time_ms_ = now_ms;
}
- // Insert frame.
- CriticalSectionScoped csB(buffer_critsect_.get());
- if (render_buffers_->AddFrame(video_frame) == 1)
- deliver_buffer_event_->Set();
-
+ // Hand over or insert frame.
+ if (disable_prerenderer_smoothing_) {
+ DeliverFrame(video_frame);
+ } else {
+ CriticalSectionScoped csB(buffer_critsect_.get());
+ if (render_buffers_->AddFrame(video_frame) == 1) {
+ deliver_buffer_event_->Set();
+ }
+ }
return 0;
}
@@ -128,21 +134,20 @@
return 0;
}
- CriticalSectionScoped csT(thread_critsect_.get());
- assert(incoming_render_thread_ == NULL);
+ if (!disable_prerenderer_smoothing_) {
+ CriticalSectionScoped csT(thread_critsect_.get());
+ assert(incoming_render_thread_ == NULL);
- incoming_render_thread_ = ThreadWrapper::CreateThread(
- IncomingVideoStreamThreadFun, this, "IncomingVideoStreamThread");
- if (!incoming_render_thread_) {
- return -1;
- }
+ incoming_render_thread_.reset(new rtc::PlatformThread(
+ IncomingVideoStreamThreadFun, this, "IncomingVideoStreamThread"));
+ if (!incoming_render_thread_) {
+ return -1;
+ }
- if (incoming_render_thread_->Start()) {
- } else {
- return -1;
+ incoming_render_thread_->Start();
+ incoming_render_thread_->SetPriority(rtc::kRealtimePriority);
+ deliver_buffer_event_->StartTimer(false, kEventStartupTimeMs);
}
- incoming_render_thread_->SetPriority(kRealtimePriority);
- deliver_buffer_event_->StartTimer(false, kEventStartupTimeMs);
running_ = true;
return 0;
@@ -155,7 +160,7 @@
return 0;
}
- ThreadWrapper* thread = NULL;
+ rtc::PlatformThread* thread = NULL;
{
CriticalSectionScoped cs_thread(thread_critsect_.get());
if (incoming_render_thread_) {
@@ -169,11 +174,8 @@
}
}
if (thread) {
- if (thread->Stop()) {
- delete thread;
- } else {
- assert(false);
- }
+ thread->Stop();
+ delete thread;
}
running_ = false;
return 0;
@@ -205,6 +207,7 @@
// Terminating
return false;
}
+
// Get a new frame to render and the time for the frame after this one.
VideoFrame frame_to_render;
uint32_t wait_time;
@@ -220,37 +223,41 @@
}
deliver_buffer_event_->StartTimer(false, wait_time);
- if (frame_to_render.IsZeroSize()) {
- if (render_callback_) {
- if (last_render_time_ms_ == 0 && !start_image_.IsZeroSize()) {
- // We have not rendered anything and have a start image.
- temp_frame_.CopyFrame(start_image_);
- render_callback_->RenderFrame(stream_id_, temp_frame_);
- } else if (!timeout_image_.IsZeroSize() &&
- last_render_time_ms_ + timeout_time_ <
- TickTime::MillisecondTimestamp()) {
- // Render a timeout image.
- temp_frame_.CopyFrame(timeout_image_);
- render_callback_->RenderFrame(stream_id_, temp_frame_);
- }
- }
-
- // No frame.
- return true;
- }
-
- // Send frame for rendering.
- if (external_callback_) {
- external_callback_->RenderFrame(stream_id_, frame_to_render);
- } else if (render_callback_) {
- render_callback_->RenderFrame(stream_id_, frame_to_render);
- }
-
- // We're done with this frame.
- if (!frame_to_render.IsZeroSize())
- last_render_time_ms_ = frame_to_render.render_time_ms();
+ DeliverFrame(frame_to_render);
}
return true;
}
+void IncomingVideoStream::DeliverFrame(const VideoFrame& video_frame) {
+ CriticalSectionScoped cs(thread_critsect_.get());
+ if (video_frame.IsZeroSize()) {
+ if (render_callback_) {
+ if (last_render_time_ms_ == 0 && !start_image_.IsZeroSize()) {
+ // We have not rendered anything and have a start image.
+ temp_frame_.CopyFrame(start_image_);
+ render_callback_->RenderFrame(stream_id_, temp_frame_);
+ } else if (!timeout_image_.IsZeroSize() &&
+ last_render_time_ms_ + timeout_time_ <
+ TickTime::MillisecondTimestamp()) {
+ // Render a timeout image.
+ temp_frame_.CopyFrame(timeout_image_);
+ render_callback_->RenderFrame(stream_id_, temp_frame_);
+ }
+ }
+
+ // No frame.
+ return;
+ }
+
+ // Send frame for rendering.
+ if (external_callback_) {
+ external_callback_->RenderFrame(stream_id_, video_frame);
+ } else if (render_callback_) {
+ render_callback_->RenderFrame(stream_id_, video_frame);
+ }
+
+ // We're done with this frame.
+ last_render_time_ms_ = video_frame.render_time_ms();
+}
+
} // namespace webrtc
diff --git a/webrtc/common_video/interface/i420_buffer_pool.h b/webrtc/common_video/interface/i420_buffer_pool.h
deleted file mode 100644
index df862cd..0000000
--- a/webrtc/common_video/interface/i420_buffer_pool.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_COMMON_VIDEO_INTERFACE_I420_BUFFER_POOL_H_
-#define WEBRTC_COMMON_VIDEO_INTERFACE_I420_BUFFER_POOL_H_
-
-#include <list>
-
-#include "webrtc/base/thread_checker.h"
-#include "webrtc/common_video/interface/video_frame_buffer.h"
-
-namespace webrtc {
-
-// Simple buffer pool to avoid unnecessary allocations of I420Buffer objects.
-// The pool manages the memory of the I420Buffer returned from CreateBuffer.
-// When the I420Buffer is destructed, the memory is returned to the pool for use
-// by subsequent calls to CreateBuffer. If the resolution passed to CreateBuffer
-// changes, old buffers will be purged from the pool.
-class I420BufferPool {
- public:
- I420BufferPool();
- // Returns a buffer from the pool, or creates a new buffer if no suitable
- // buffer exists in the pool.
- rtc::scoped_refptr<VideoFrameBuffer> CreateBuffer(int width, int height);
- // Clears buffers_ and detaches the thread checker so that it can be reused
- // later from another thread.
- void Release();
-
- private:
- rtc::ThreadChecker thread_checker_;
- std::list<rtc::scoped_refptr<I420Buffer>> buffers_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_COMMON_VIDEO_INTERFACE_I420_BUFFER_POOL_H_
diff --git a/webrtc/common_video/interface/incoming_video_stream.h b/webrtc/common_video/interface/incoming_video_stream.h
deleted file mode 100644
index 74ecc4e..0000000
--- a/webrtc/common_video/interface/incoming_video_stream.h
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_COMMON_VIDEO_INTERFACE_INCOMING_VIDEO_STREAM_H_
-#define WEBRTC_COMMON_VIDEO_INTERFACE_INCOMING_VIDEO_STREAM_H_
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_annotations.h"
-#include "webrtc/common_video/video_render_frames.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventTimerWrapper;
-class ThreadWrapper;
-
-class VideoRenderCallback {
- public:
- virtual int32_t RenderFrame(const uint32_t streamId,
- const VideoFrame& videoFrame) = 0;
-
- protected:
- virtual ~VideoRenderCallback() {}
-};
-
-class IncomingVideoStream : public VideoRenderCallback {
- public:
- explicit IncomingVideoStream(uint32_t stream_id);
- ~IncomingVideoStream();
-
- // Get callback to deliver frames to the module.
- VideoRenderCallback* ModuleCallback();
- virtual int32_t RenderFrame(const uint32_t stream_id,
- const VideoFrame& video_frame);
-
- // Set callback to the platform dependent code.
- void SetRenderCallback(VideoRenderCallback* render_callback);
-
- // Callback for file recording, snapshot, ...
- void SetExternalCallback(VideoRenderCallback* render_object);
-
- // Start/Stop.
- int32_t Start();
- int32_t Stop();
-
- // Clear all buffers.
- int32_t Reset();
-
- // Properties.
- uint32_t StreamId() const;
- uint32_t IncomingRate() const;
-
- int32_t SetStartImage(const VideoFrame& video_frame);
-
- int32_t SetTimeoutImage(const VideoFrame& video_frame,
- const uint32_t timeout);
-
- int32_t SetExpectedRenderDelay(int32_t delay_ms);
-
- protected:
- static bool IncomingVideoStreamThreadFun(void* obj);
- bool IncomingVideoStreamProcess();
-
- private:
- enum { kEventStartupTimeMs = 10 };
- enum { kEventMaxWaitTimeMs = 100 };
- enum { kFrameRatePeriodMs = 1000 };
-
- uint32_t const stream_id_;
- // Critsects in allowed to enter order.
- const rtc::scoped_ptr<CriticalSectionWrapper> stream_critsect_;
- const rtc::scoped_ptr<CriticalSectionWrapper> thread_critsect_;
- const rtc::scoped_ptr<CriticalSectionWrapper> buffer_critsect_;
- rtc::scoped_ptr<ThreadWrapper> incoming_render_thread_
- GUARDED_BY(thread_critsect_);
- rtc::scoped_ptr<EventTimerWrapper> deliver_buffer_event_;
-
- bool running_ GUARDED_BY(stream_critsect_);
- VideoRenderCallback* external_callback_ GUARDED_BY(thread_critsect_);
- VideoRenderCallback* render_callback_ GUARDED_BY(thread_critsect_);
- const rtc::scoped_ptr<VideoRenderFrames> render_buffers_
- GUARDED_BY(buffer_critsect_);
-
- uint32_t incoming_rate_ GUARDED_BY(stream_critsect_);
- int64_t last_rate_calculation_time_ms_ GUARDED_BY(stream_critsect_);
- uint16_t num_frames_since_last_calculation_ GUARDED_BY(stream_critsect_);
- int64_t last_render_time_ms_ GUARDED_BY(thread_critsect_);
- VideoFrame temp_frame_ GUARDED_BY(thread_critsect_);
- VideoFrame start_image_ GUARDED_BY(thread_critsect_);
- VideoFrame timeout_image_ GUARDED_BY(thread_critsect_);
- uint32_t timeout_time_ GUARDED_BY(thread_critsect_);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_COMMON_VIDEO_INTERFACE_INCOMING_VIDEO_STREAM_H_
diff --git a/webrtc/common_video/interface/video_frame_buffer.h b/webrtc/common_video/interface/video_frame_buffer.h
deleted file mode 100644
index 1062165..0000000
--- a/webrtc/common_video/interface/video_frame_buffer.h
+++ /dev/null
@@ -1,157 +0,0 @@
-/*
- * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_FRAME_BUFFER_H_
-#define WEBRTC_VIDEO_FRAME_BUFFER_H_
-
-#include "webrtc/base/callback.h"
-#include "webrtc/base/refcount.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/system_wrappers/include/aligned_malloc.h"
-
-namespace webrtc {
-
-enum PlaneType {
- kYPlane = 0,
- kUPlane = 1,
- kVPlane = 2,
- kNumOfPlanes = 3,
-};
-
-// Interface of a simple frame buffer containing pixel data. This interface does
-// not contain any frame metadata such as rotation, timestamp, pixel_width, etc.
-class VideoFrameBuffer : public rtc::RefCountInterface {
- public:
- // Returns true if this buffer has a single exclusive owner.
- virtual bool HasOneRef() const = 0;
-
- // The resolution of the frame in pixels. For formats where some planes are
- // subsampled, this is the highest-resolution plane.
- virtual int width() const = 0;
- virtual int height() const = 0;
-
- // Returns pointer to the pixel data for a given plane. The memory is owned by
- // the VideoFrameBuffer object and must not be freed by the caller.
- virtual const uint8_t* data(PlaneType type) const = 0;
-
- // Non-const data access is disallowed by default. You need to make sure you
- // have exclusive access and a writable buffer before calling this function.
- virtual uint8_t* MutableData(PlaneType type);
-
- // Returns the number of bytes between successive rows for a given plane.
- virtual int stride(PlaneType type) const = 0;
-
- // Return the handle of the underlying video frame. This is used when the
- // frame is backed by a texture.
- virtual void* native_handle() const = 0;
-
- // Returns a new memory-backed frame buffer converted from this buffer's
- // native handle.
- virtual rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() = 0;
-
- protected:
- virtual ~VideoFrameBuffer();
-};
-
-// Plain I420 buffer in standard memory.
-class I420Buffer : public VideoFrameBuffer {
- public:
- I420Buffer(int width, int height);
- I420Buffer(int width, int height, int stride_y, int stride_u, int stride_v);
-
- int width() const override;
- int height() const override;
- const uint8_t* data(PlaneType type) const override;
- // Non-const data access is only allowed if HasOneRef() is true to protect
- // against unexpected overwrites.
- uint8_t* MutableData(PlaneType type) override;
- int stride(PlaneType type) const override;
- void* native_handle() const override;
- rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override;
-
- protected:
- ~I420Buffer() override;
-
- private:
- const int width_;
- const int height_;
- const int stride_y_;
- const int stride_u_;
- const int stride_v_;
- const rtc::scoped_ptr<uint8_t, AlignedFreeDeleter> data_;
-};
-
-// Base class for native-handle buffer is a wrapper around a |native_handle|.
-// This is used for convenience as most native-handle implementations can share
-// many VideoFrame implementations, but need to implement a few others (such
-// as their own destructors or conversion methods back to software I420).
-class NativeHandleBuffer : public VideoFrameBuffer {
- public:
- NativeHandleBuffer(void* native_handle, int width, int height);
-
- int width() const override;
- int height() const override;
- const uint8_t* data(PlaneType type) const override;
- int stride(PlaneType type) const override;
- void* native_handle() const override;
-
- protected:
- void* native_handle_;
- const int width_;
- const int height_;
-};
-
-class WrappedI420Buffer : public webrtc::VideoFrameBuffer {
- public:
- WrappedI420Buffer(int width,
- int height,
- const uint8_t* y_plane,
- int y_stride,
- const uint8_t* u_plane,
- int u_stride,
- const uint8_t* v_plane,
- int v_stride,
- const rtc::Callback0<void>& no_longer_used);
- int width() const override;
- int height() const override;
-
- const uint8_t* data(PlaneType type) const override;
-
- int stride(PlaneType type) const override;
- void* native_handle() const override;
-
- rtc::scoped_refptr<VideoFrameBuffer> NativeToI420Buffer() override;
-
- private:
- friend class rtc::RefCountedObject<WrappedI420Buffer>;
- ~WrappedI420Buffer() override;
-
- const int width_;
- const int height_;
- const uint8_t* const y_plane_;
- const uint8_t* const u_plane_;
- const uint8_t* const v_plane_;
- const int y_stride_;
- const int u_stride_;
- const int v_stride_;
- rtc::Callback0<void> no_longer_used_cb_;
-};
-
-// Helper function to crop |buffer| without making a deep copy. May only be used
-// for non-native frames.
-rtc::scoped_refptr<VideoFrameBuffer> ShallowCenterCrop(
- const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
- int cropped_width,
- int cropped_height);
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_FRAME_BUFFER_H_
diff --git a/webrtc/common_video/interface/video_image.h b/webrtc/common_video/interface/video_image.h
deleted file mode 100644
index 4cbf23f..0000000
--- a/webrtc/common_video/interface/video_image.h
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
-#define COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
-
-// TODO(pbos): Remove this file and include webrtc/video_frame.h instead.
-#include "webrtc/video_frame.h"
-
-#endif // COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
diff --git a/webrtc/common_video/libyuv/include/scaler.h b/webrtc/common_video/libyuv/include/scaler.h
index c04d01f..2b92f81 100644
--- a/webrtc/common_video/libyuv/include/scaler.h
+++ b/webrtc/common_video/libyuv/include/scaler.h
@@ -15,7 +15,7 @@
#ifndef WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_SCALER_H_
#define WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_SCALER_H_
-#include "webrtc/common_video/interface/i420_buffer_pool.h"
+#include "webrtc/common_video/include/i420_buffer_pool.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_frame.h"
diff --git a/webrtc/common_video/libyuv/include/webrtc_libyuv.h b/webrtc/common_video/libyuv/include/webrtc_libyuv.h
index baf8c17..d66736f 100644
--- a/webrtc/common_video/libyuv/include/webrtc_libyuv.h
+++ b/webrtc/common_video/libyuv/include/webrtc_libyuv.h
@@ -152,6 +152,7 @@
double I420PSNR(const VideoFrame* ref_frame, const VideoFrame* test_frame);
// Compute SSIM for an I420 frame (all planes).
double I420SSIM(const VideoFrame* ref_frame, const VideoFrame* test_frame);
-}
+
+} // namespace webrtc
#endif // WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_WEBRTC_LIBYUV_H_
diff --git a/webrtc/common_video/libyuv/libyuv_unittest.cc b/webrtc/common_video/libyuv/libyuv_unittest.cc
index b674b51..e7cf075 100644
--- a/webrtc/common_video/libyuv/libyuv_unittest.cc
+++ b/webrtc/common_video/libyuv/libyuv_unittest.cc
@@ -97,7 +97,7 @@
width_(352),
height_(288),
size_y_(width_ * height_),
- size_uv_(((width_ + 1 ) / 2) * ((height_ + 1) / 2)),
+ size_uv_(((width_ + 1) / 2) * ((height_ + 1) / 2)),
frame_length_(CalcBufferSize(kI420, 352, 288)) {
orig_buffer_.reset(new uint8_t[frame_length_]);
}
@@ -142,9 +142,9 @@
double psnr = 0.0;
VideoFrame res_i420_frame;
- EXPECT_EQ(0,res_i420_frame.CreateEmptyFrame(width_, height_, width_,
- (width_ + 1) / 2,
- (width_ + 1) / 2));
+ EXPECT_EQ(0, res_i420_frame.CreateEmptyFrame(width_, height_, width_,
+ (width_ + 1) / 2,
+ (width_ + 1) / 2));
printf("\nConvert #%d I420 <-> I420 \n", j);
rtc::scoped_ptr<uint8_t[]> out_i420_buffer(new uint8_t[frame_length_]);
EXPECT_EQ(0, ConvertFromI420(orig_frame_, kI420, 0,
@@ -281,8 +281,8 @@
int stride_y = 0;
int stride_uv = 0;
Calc16ByteAlignedStride(width_, &stride_y, &stride_uv);
- EXPECT_EQ(0,res_i420_frame.CreateEmptyFrame(width_, height_,
- stride_y, stride_uv, stride_uv));
+ EXPECT_EQ(0, res_i420_frame.CreateEmptyFrame(width_, height_,
+ stride_y, stride_uv, stride_uv));
rtc::scoped_ptr<uint8_t[]> out_i420_buffer(new uint8_t[frame_length_]);
EXPECT_EQ(0, ConvertFromI420(orig_frame_, kI420, 0,
out_i420_buffer.get()));
@@ -303,27 +303,27 @@
VideoFrame rotated_res_i420_frame;
int rotated_width = height_;
int rotated_height = width_;
- int stride_y ;
+ int stride_y;
int stride_uv;
Calc16ByteAlignedStride(rotated_width, &stride_y, &stride_uv);
- EXPECT_EQ(0,rotated_res_i420_frame.CreateEmptyFrame(rotated_width,
- rotated_height,
- stride_y,
- stride_uv,
- stride_uv));
+ EXPECT_EQ(0, rotated_res_i420_frame.CreateEmptyFrame(rotated_width,
+ rotated_height,
+ stride_y,
+ stride_uv,
+ stride_uv));
EXPECT_EQ(0, ConvertToI420(kI420, orig_buffer_.get(), 0, 0, width_, height_,
0, kVideoRotation_90, &rotated_res_i420_frame));
EXPECT_EQ(0, ConvertToI420(kI420, orig_buffer_.get(), 0, 0, width_, height_,
0, kVideoRotation_270, &rotated_res_i420_frame));
- EXPECT_EQ(0,rotated_res_i420_frame.CreateEmptyFrame(width_, height_,
- width_, (width_ + 1) / 2,
- (width_ + 1) / 2));
+ EXPECT_EQ(0, rotated_res_i420_frame.CreateEmptyFrame(width_, height_,
+ width_, (width_ + 1) / 2,
+ (width_ + 1) / 2));
EXPECT_EQ(0, ConvertToI420(kI420, orig_buffer_.get(), 0, 0, width_, height_,
0, kVideoRotation_180, &rotated_res_i420_frame));
}
TEST_F(TestLibYuv, alignment) {
- int value = 0x3FF; // 1023
+ int value = 0x3FF; // 1023
EXPECT_EQ(0x400, AlignInt(value, 128)); // Low 7 bits are zero.
EXPECT_EQ(0x400, AlignInt(value, 64)); // Low 6 bits are zero.
EXPECT_EQ(0x400, AlignInt(value, 32)); // Low 5 bits are zero.
@@ -346,4 +346,4 @@
EXPECT_EQ(64, stride_uv);
}
-} // namespace
+} // namespace webrtc
diff --git a/webrtc/common_video/libyuv/scaler_unittest.cc b/webrtc/common_video/libyuv/scaler_unittest.cc
index 568311b..6d02638 100644
--- a/webrtc/common_video/libyuv/scaler_unittest.cc
+++ b/webrtc/common_video/libyuv/scaler_unittest.cc
@@ -15,7 +15,6 @@
#include "webrtc/common_video/libyuv/include/scaler.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
namespace webrtc {
@@ -114,8 +113,13 @@
EXPECT_EQ(half_height_, test_frame2.height());
}
-//TODO (mikhal): Converge the test into one function that accepts the method.
-TEST_F(TestScaler, DISABLED_ON_ANDROID(PointScaleTest)) {
+// TODO(mikhal): Converge the test into one function that accepts the method.
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_PointScaleTest DISABLED_PointScaleTest
+#else
+#define MAYBE_PointScaleTest PointScaleTest
+#endif
+TEST_F(TestScaler, MAYBE_PointScaleTest) {
double avg_psnr;
FILE* source_file2;
ScaleMethod method = kScalePoint;
@@ -182,7 +186,12 @@
ASSERT_EQ(0, fclose(source_file2));
}
-TEST_F(TestScaler, DISABLED_ON_ANDROID(BiLinearScaleTest)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_BilinearScaleTest DISABLED_BiLinearScaleTest
+#else
+#define MAYBE_BilinearScaleTest BiLinearScaleTest
+#endif
+TEST_F(TestScaler, MAYBE_BiLinearScaleTest) {
double avg_psnr;
FILE* source_file2;
ScaleMethod method = kScaleBilinear;
@@ -234,7 +243,12 @@
400, 300);
}
-TEST_F(TestScaler, DISABLED_ON_ANDROID(BoxScaleTest)) {
+#if defined(WEBRTC_ANDROID)
+#define MAYBE_BoxScaleTest DISABLED_BoxScaleTest
+#else
+#define MAYBE_BoxScaleTest BoxScaleTest
+#endif
+TEST_F(TestScaler, MAYBE_BoxScaleTest) {
double avg_psnr;
FILE* source_file2;
ScaleMethod method = kScaleBox;
@@ -322,7 +336,7 @@
return avg_psnr;
}
-// TODO (mikhal): Move part to a separate scale test.
+// TODO(mikhal): Move part to a separate scale test.
void TestScaler::ScaleSequence(ScaleMethod method,
FILE* source_file, std::string out_name,
int src_width, int src_height,
diff --git a/webrtc/common_video/libyuv/webrtc_libyuv.cc b/webrtc/common_video/libyuv/webrtc_libyuv.cc
index bf95624..48f5c20 100644
--- a/webrtc/common_video/libyuv/webrtc_libyuv.cc
+++ b/webrtc/common_video/libyuv/webrtc_libyuv.cc
@@ -58,7 +58,7 @@
int AlignInt(int value, int alignment) {
assert(!((alignment - 1) & alignment));
- return ((value + alignment - 1) & ~ (alignment - 1));
+ return ((value + alignment - 1) & ~(alignment - 1));
}
void Calc16ByteAlignedStride(int width, int* stride_y, int* stride_uv) {
@@ -119,8 +119,8 @@
}
plane_buffer += frame.stride(plane_type);
}
- }
- return 0;
+ }
+ return 0;
}
int ExtractBuffer(const VideoFrame& input_frame, size_t size, uint8_t* buffer) {
@@ -176,7 +176,7 @@
}
libyuv::RotationMode ConvertRotationMode(VideoRotation rotation) {
- switch(rotation) {
+ switch (rotation) {
case kVideoRotation_0:
return libyuv::kRotate0;
case kVideoRotation_90:
@@ -191,7 +191,7 @@
}
int ConvertVideoType(VideoType video_type) {
- switch(video_type) {
+ switch (video_type) {
case kUnknown:
return libyuv::FOURCC_ANY;
case kI420:
@@ -243,7 +243,7 @@
// Stride values should correspond to the destination values.
if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) {
dst_width = dst_frame->height();
- dst_height =dst_frame->width();
+ dst_height = dst_frame->width();
}
return libyuv::ConvertToI420(src_frame, sample_size,
dst_frame->buffer(kYPlane),
diff --git a/webrtc/common_video/plane.cc b/webrtc/common_video/plane.cc
deleted file mode 100644
index e0bbba1..0000000
--- a/webrtc/common_video/plane.cc
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/common_video/plane.h"
-
-#include <string.h> // memcpy
-
-#include <algorithm> // swap
-
-namespace webrtc {
-
-// Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
-static const int kBufferAlignment = 64;
-
-Plane::Plane()
- : allocated_size_(0),
- plane_size_(0),
- stride_(0) {}
-
-Plane::~Plane() {}
-
-int Plane::CreateEmptyPlane(int allocated_size, int stride, int plane_size) {
- if (allocated_size < 1 || stride < 1 || plane_size < 1)
- return -1;
- stride_ = stride;
- if (MaybeResize(allocated_size) < 0)
- return -1;
- plane_size_ = plane_size;
- return 0;
-}
-
-int Plane::MaybeResize(int new_size) {
- if (new_size <= 0)
- return -1;
- if (new_size <= allocated_size_)
- return 0;
- rtc::scoped_ptr<uint8_t, AlignedFreeDeleter> new_buffer(
- static_cast<uint8_t*>(AlignedMalloc(new_size, kBufferAlignment)));
- if (buffer_.get()) {
- memcpy(new_buffer.get(), buffer_.get(), plane_size_);
- }
- buffer_.reset(new_buffer.release());
- allocated_size_ = new_size;
- return 0;
-}
-
-int Plane::Copy(const Plane& plane) {
- if (MaybeResize(plane.allocated_size_) < 0)
- return -1;
- if (plane.buffer_.get())
- memcpy(buffer_.get(), plane.buffer_.get(), plane.plane_size_);
- stride_ = plane.stride_;
- plane_size_ = plane.plane_size_;
- return 0;
-}
-
-int Plane::Copy(int size, int stride, const uint8_t* buffer) {
- if (MaybeResize(size) < 0)
- return -1;
- memcpy(buffer_.get(), buffer, size);
- plane_size_ = size;
- stride_ = stride;
- return 0;
-}
-
-void Plane::Swap(Plane& plane) {
- std::swap(stride_, plane.stride_);
- std::swap(allocated_size_, plane.allocated_size_);
- std::swap(plane_size_, plane.plane_size_);
- buffer_.swap(plane.buffer_);
-}
-
-} // namespace webrtc
diff --git a/webrtc/common_video/plane.h b/webrtc/common_video/plane.h
deleted file mode 100644
index 3ef949a..0000000
--- a/webrtc/common_video/plane.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef COMMON_VIDEO_PLANE_H
-#define COMMON_VIDEO_PLANE_H
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/include/aligned_malloc.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// Helper class for VideoFrame: Store plane data and perform basic plane
-// operations.
-class Plane {
- public:
- Plane();
- ~Plane();
- // CreateEmptyPlane - set allocated size, actual plane size and stride:
- // If current size is smaller than current size, then a buffer of sufficient
- // size will be allocated.
- // Return value: 0 on success ,-1 on error.
- int CreateEmptyPlane(int allocated_size, int stride, int plane_size);
-
- // Copy the entire plane data.
- // Return value: 0 on success ,-1 on error.
- int Copy(const Plane& plane);
-
- // Copy buffer: If current size is smaller
- // than current size, then a buffer of sufficient size will be allocated.
- // Return value: 0 on success ,-1 on error.
- int Copy(int size, int stride, const uint8_t* buffer);
-
- // Swap plane data.
- void Swap(Plane& plane);
-
- // Get allocated size.
- int allocated_size() const {return allocated_size_;}
-
- // Set actual size.
- void ResetSize() {plane_size_ = 0;}
-
- // Return true is plane size is zero, false if not.
- bool IsZeroSize() const {return plane_size_ == 0;}
-
- // Get stride value.
- int stride() const {return stride_;}
-
- // Return data pointer.
- const uint8_t* buffer() const {return buffer_.get();}
- // Overloading with non-const.
- uint8_t* buffer() {return buffer_.get();}
-
- private:
- // Resize when needed: If current allocated size is less than new_size, buffer
- // will be updated. Old data will be copied to new buffer.
- // Return value: 0 on success ,-1 on error.
- int MaybeResize(int new_size);
-
- rtc::scoped_ptr<uint8_t, AlignedFreeDeleter> buffer_;
- int allocated_size_;
- int plane_size_;
- int stride_;
-}; // Plane
-
-} // namespace webrtc
-
-#endif // COMMON_VIDEO_PLANE_H
diff --git a/webrtc/common_video/video_frame.cc b/webrtc/common_video/video_frame.cc
index 7cdbd53..8ccd821 100644
--- a/webrtc/common_video/video_frame.cc
+++ b/webrtc/common_video/video_frame.cc
@@ -19,6 +19,26 @@
namespace webrtc {
+bool EqualPlane(const uint8_t* data1,
+ const uint8_t* data2,
+ int stride,
+ int width,
+ int height) {
+ for (int y = 0; y < height; ++y) {
+ if (memcmp(data1, data2, width) != 0)
+ return false;
+ data1 += stride;
+ data2 += stride;
+ }
+ return true;
+}
+
+int ExpectedSize(int plane_stride, int image_height, PlaneType type) {
+ if (type == kYPlane)
+ return plane_stride * image_height;
+ return plane_stride * ((image_height + 1) / 2);
+}
+
VideoFrame::VideoFrame() {
// Intentionally using Reset instead of initializer list so that any missed
// fields in Reset will be caught by memory checkers.
@@ -202,4 +222,24 @@
return frame;
}
+bool VideoFrame::EqualsFrame(const VideoFrame& frame) const {
+ if (width() != frame.width() || height() != frame.height() ||
+ stride(kYPlane) != frame.stride(kYPlane) ||
+ stride(kUPlane) != frame.stride(kUPlane) ||
+ stride(kVPlane) != frame.stride(kVPlane) ||
+ timestamp() != frame.timestamp() ||
+ ntp_time_ms() != frame.ntp_time_ms() ||
+ render_time_ms() != frame.render_time_ms()) {
+ return false;
+ }
+ const int half_width = (width() + 1) / 2;
+ const int half_height = (height() + 1) / 2;
+ return EqualPlane(buffer(kYPlane), frame.buffer(kYPlane),
+ stride(kYPlane), width(), height()) &&
+ EqualPlane(buffer(kUPlane), frame.buffer(kUPlane),
+ stride(kUPlane), half_width, half_height) &&
+ EqualPlane(buffer(kVPlane), frame.buffer(kVPlane),
+ stride(kVPlane), half_width, half_height);
+}
+
} // namespace webrtc
diff --git a/webrtc/common_video/video_frame_buffer.cc b/webrtc/common_video/video_frame_buffer.cc
index 36ee14a..492bc49 100644
--- a/webrtc/common_video/video_frame_buffer.cc
+++ b/webrtc/common_video/video_frame_buffer.cc
@@ -8,21 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/common_video/interface/video_frame_buffer.h"
+#include "webrtc/common_video/include/video_frame_buffer.h"
-#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/keep_ref_until_done.h"
// Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
static const int kBufferAlignment = 64;
namespace webrtc {
-namespace {
-
-// Used in rtc::Bind to keep a buffer alive until destructor is called.
-static void NoLongerUsedCallback(rtc::scoped_refptr<VideoFrameBuffer> dummy) {}
-
-} // anonymous namespace
uint8_t* VideoFrameBuffer::MutableData(PlaneType type) {
RTC_NOTREACHED();
@@ -238,7 +232,7 @@
y_plane, buffer->stride(kYPlane),
u_plane, buffer->stride(kUPlane),
v_plane, buffer->stride(kVPlane),
- rtc::Bind(&NoLongerUsedCallback, buffer));
+ rtc::KeepRefUntilDone(buffer));
}
} // namespace webrtc
diff --git a/webrtc/common_video/video_render_frames.cc b/webrtc/common_video/video_render_frames.cc
index f4ece5e..8b447cb 100644
--- a/webrtc/common_video/video_render_frames.cc
+++ b/webrtc/common_video/video_render_frames.cc
@@ -12,7 +12,7 @@
#include <assert.h>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"
@@ -20,7 +20,7 @@
const uint32_t KEventMaxWaitTimeMs = 200;
const uint32_t kMinRenderDelayMs = 10;
-const uint32_t kMaxRenderDelayMs= 500;
+const uint32_t kMaxRenderDelayMs = 500;
VideoRenderFrames::VideoRenderFrames()
: render_delay_ms_(10) {
diff --git a/webrtc/config.h b/webrtc/config.h
index 5271163..06460ae 100644
--- a/webrtc/config.h
+++ b/webrtc/config.h
@@ -16,6 +16,7 @@
#include <string>
#include <vector>
+#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/typedefs.h"
@@ -49,10 +50,13 @@
int red_rtx_payload_type;
};
-// RTP header extension to use for the video stream, see RFC 5285.
+// RTP header extension, see RFC 5285.
struct RtpExtension {
RtpExtension(const std::string& name, int id) : name(name), id(id) {}
std::string ToString() const;
+ bool operator==(const RtpExtension& rhs) const {
+ return name == rhs.name && id == rhs.id;
+ }
static bool IsSupportedForAudio(const std::string& name);
static bool IsSupportedForVideo(const std::string& name);
@@ -104,6 +108,7 @@
std::string ToString() const;
std::vector<VideoStream> streams;
+ std::vector<SpatialLayer> spatial_layers;
ContentType content_type;
void* encoder_specific_settings;
@@ -124,6 +129,7 @@
struct NetEqCapacityConfig {
NetEqCapacityConfig() : enabled(false), capacity(0) {}
explicit NetEqCapacityConfig(int value) : enabled(true), capacity(value) {}
+ static const ConfigOptionID identifier = ConfigOptionID::kNetEqCapacityConfig;
bool enabled;
int capacity;
};
@@ -131,6 +137,14 @@
struct NetEqFastAccelerate {
NetEqFastAccelerate() : enabled(false) {}
explicit NetEqFastAccelerate(bool value) : enabled(value) {}
+ static const ConfigOptionID identifier = ConfigOptionID::kNetEqFastAccelerate;
+ bool enabled;
+};
+
+struct VoicePacing {
+ VoicePacing() : enabled(false) {}
+ explicit VoicePacing(bool value) : enabled(value) {}
+ static const ConfigOptionID identifier = ConfigOptionID::kVoicePacing;
bool enabled;
};
diff --git a/webrtc/engine_configurations.h b/webrtc/engine_configurations.h
index c832d9a..42b1816 100644
--- a/webrtc/engine_configurations.h
+++ b/webrtc/engine_configurations.h
@@ -14,19 +14,6 @@
#include "webrtc/typedefs.h"
// ============================================================================
-// Voice and Video
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// [Video] Codec settings
-// ----------------------------------------------------------------------------
-
-#define VIDEOCODEC_I420
-#define VIDEOCODEC_VP8
-#define VIDEOCODEC_VP9
-#define VIDEOCODEC_H264
-
-// ============================================================================
// VoiceEngine
// ============================================================================
diff --git a/webrtc/examples/android/media_demo/AndroidManifest.xml b/webrtc/examples/android/media_demo/AndroidManifest.xml
deleted file mode 100644
index 62bf460..0000000
--- a/webrtc/examples/android/media_demo/AndroidManifest.xml
+++ /dev/null
@@ -1,29 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
- android:versionCode="1" package="org.webrtc.webrtcdemo" android:versionName="1.07">
- <application android:icon="@drawable/logo"
- android:label="@string/appName"
- android:debuggable="true">
- <activity android:name=".WebRTCDemo"
- android:theme="@android:style/Theme.Holo"
- android:label="@string/appName"
- android:screenOrientation="landscape"
- >
- <intent-filter>
- <action android:name="android.intent.action.MAIN" />
- <category android:name="android.intent.category.LAUNCHER" />
- <action android:name="android.intent.action.HEADSET_PLUG"/>
- </intent-filter>
- </activity>
- </application>
-
- <uses-sdk android:minSdkVersion="14" />
- <uses-permission android:name="android.permission.CAMERA"></uses-permission>
- <uses-feature android:name="android.hardware.camera" />
- <uses-feature android:name="android.hardware.camera.autofocus" />
- <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
- <uses-permission android:name="android.permission.RECORD_AUDIO" />
- <uses-permission android:name="android.permission.INTERNET" />
- <uses-permission android:name="android.permission.WAKE_LOCK" />
- <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
-</manifest>
\ No newline at end of file
diff --git a/webrtc/examples/android/media_demo/README b/webrtc/examples/android/media_demo/README
deleted file mode 100644
index af71f38..0000000
--- a/webrtc/examples/android/media_demo/README
+++ /dev/null
@@ -1,24 +0,0 @@
-This directory contains a sample app for sending and receiving audio
-on Android. It further lets you enable and disable some call quality
-enhancements such as echo cancellation, noise suppression etc.
-
-Prerequisites:
-- Make sure gclient is checking out tools necessary to target Android: your
- .gclient file should contain a line like:
- target_os = ['android']
- Make sure to re-run gclient sync after adding this to download the tools.
-- Env vars need to be set up to target Android; easiest way to do this is to run
- (from the libjingle trunk directory):
- . ./build/android/envsetup.sh
- Note that this clobbers any previously-set $GYP_DEFINES so it must be done
- before the next item.
-- Set up webrtc-related GYP variables:
- export GYP_DEFINES="$GYP_DEFINES java_home=</path/to/JDK>"
-- Finally, run "gclient runhooks" to generate Android-targeting .ninja files.
-
-Example of building the app:
-cd <path/to/repository>/trunk
-ninja -C out/Debug WebRTCDemo
-
-It can then be installed and run on the device:
-adb install -r out/Debug/WebRTCDemo-debug.apk
diff --git a/webrtc/examples/android/media_demo/build.xml b/webrtc/examples/android/media_demo/build.xml
deleted file mode 100644
index 1773488..0000000
--- a/webrtc/examples/android/media_demo/build.xml
+++ /dev/null
@@ -1,92 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<project name="WebRTCDemo" default="help">
-
- <!-- The local.properties file is created and updated by the 'android' tool.
- It contains the path to the SDK. It should *NOT* be checked into
- Version Control Systems. -->
- <property file="local.properties" />
-
- <!-- The ant.properties file can be created by you. It is only edited by the
- 'android' tool to add properties to it.
- This is the place to change some Ant specific build properties.
- Here are some properties you may want to change/update:
-
- source.dir
- The name of the source directory. Default is 'src'.
- out.dir
- The name of the output directory. Default is 'bin'.
-
- For other overridable properties, look at the beginning of the rules
- files in the SDK, at tools/ant/build.xml
-
- Properties related to the SDK location or the project target should
- be updated using the 'android' tool with the 'update' action.
-
- This file is an integral part of the build system for your
- application and should be checked into Version Control Systems.
-
- -->
- <property file="ant.properties" />
-
- <!-- if sdk.dir was not set from one of the property file, then
- get it from the ANDROID_HOME env var.
- This must be done before we load project.properties since
- the proguard config can use sdk.dir -->
- <property environment="env" />
- <condition property="sdk.dir" value="${env.ANDROID_SDK_ROOT}">
- <isset property="env.ANDROID_SDK_ROOT" />
- </condition>
-
- <!-- The project.properties file is created and updated by the 'android'
- tool, as well as ADT.
-
- This contains project specific properties such as project target, and library
- dependencies. Lower level build properties are stored in ant.properties
- (or in .classpath for Eclipse projects).
-
- This file is an integral part of the build system for your
- application and should be checked into Version Control Systems. -->
- <loadproperties srcFile="project.properties" />
-
- <!-- quick check on sdk.dir -->
- <fail
- message="sdk.dir is missing. Make sure to generate local.properties using 'android update project' or to inject it through the ANDROID_SDK_ROOT environment variable."
- unless="sdk.dir"
- />
-
- <!--
- Import per project custom build rules if present at the root of the project.
- This is the place to put custom intermediary targets such as:
- -pre-build
- -pre-compile
- -post-compile (This is typically used for code obfuscation.
- Compiled code location: ${out.classes.absolute.dir}
- If this is not done in place, override ${out.dex.input.absolute.dir})
- -post-package
- -post-build
- -pre-clean
- -->
- <import file="custom_rules.xml" optional="true" />
-
- <!-- Import the actual build file.
-
- To customize existing targets, there are two options:
- - Customize only one target:
- - copy/paste the target into this file, *before* the
- <import> task.
- - customize it to your needs.
- - Customize the whole content of build.xml
- - copy/paste the content of the rules files (minus the top node)
- into this file, replacing the <import> task.
- - customize to your needs.
-
- ***********************
- ****** IMPORTANT ******
- ***********************
- In all cases you must update the value of version-tag below to read 'custom' instead of an integer,
- in order to avoid having your file be overridden by tools such as "android update project"
- -->
- <!-- version-tag: 1 -->
- <import file="${sdk.dir}/tools/ant/build.xml" />
-
-</project>
diff --git a/webrtc/examples/android/media_demo/jni/jni_helpers.cc b/webrtc/examples/android/media_demo/jni/jni_helpers.cc
deleted file mode 100644
index b0d1a74..0000000
--- a/webrtc/examples/android/media_demo/jni/jni_helpers.cc
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/examples/android/media_demo/jni/jni_helpers.h"
-
-#include <limits>
-
-#include "unicode/unistr.h"
-
-using icu::UnicodeString;
-
-jmethodID GetMethodID(JNIEnv* jni, jclass c, const std::string& name,
- const char* signature) {
- jmethodID m = jni->GetMethodID(c, name.c_str(), signature);
- CHECK_JNI_EXCEPTION(jni, "error during GetMethodID");
- return m;
-}
-
-jlong jlongFromPointer(void* ptr) {
- CHECK(sizeof(intptr_t) <= sizeof(jlong), "Time to rethink the use of jlongs");
- // Going through intptr_t to be obvious about the definedness of the
- // conversion from pointer to integral type. intptr_t to jlong is a standard
- // widening by the COMPILE_ASSERT above.
- jlong ret = reinterpret_cast<intptr_t>(ptr);
- CHECK(reinterpret_cast<void*>(ret) == ptr,
- "jlong does not convert back to pointer");
- return ret;
-}
-
-// Given a (UTF-16) jstring return a new UTF-8 native string.
-std::string JavaToStdString(JNIEnv* jni, const jstring& j_string) {
- const jchar* jchars = jni->GetStringChars(j_string, NULL);
- CHECK_JNI_EXCEPTION(jni, "Error during GetStringChars");
- UnicodeString ustr(jchars, jni->GetStringLength(j_string));
- CHECK_JNI_EXCEPTION(jni, "Error during GetStringLength");
- jni->ReleaseStringChars(j_string, jchars);
- CHECK_JNI_EXCEPTION(jni, "Error during ReleaseStringChars");
- std::string ret;
- return ustr.toUTF8String(ret);
-}
-
-ClassReferenceHolder::ClassReferenceHolder(JNIEnv* jni, const char** classes,
- int size) {
- for (int i = 0; i < size; ++i) {
- LoadClass(jni, classes[i]);
- }
-}
-ClassReferenceHolder::~ClassReferenceHolder() {
- CHECK(classes_.empty(), "Must call FreeReferences() before dtor!");
-}
-
-void ClassReferenceHolder::FreeReferences(JNIEnv* jni) {
- for (std::map<std::string, jclass>::const_iterator it = classes_.begin();
- it != classes_.end(); ++it) {
- jni->DeleteGlobalRef(it->second);
- }
- classes_.clear();
-}
-
-jclass ClassReferenceHolder::GetClass(const std::string& name) {
- std::map<std::string, jclass>::iterator it = classes_.find(name);
- CHECK(it != classes_.end(), "Could not find class");
- return it->second;
-}
-
-void ClassReferenceHolder::LoadClass(JNIEnv* jni, const std::string& name) {
- jclass localRef = jni->FindClass(name.c_str());
- CHECK_JNI_EXCEPTION(jni, "Could not load class");
- CHECK(localRef, name.c_str());
- jclass globalRef = reinterpret_cast<jclass>(jni->NewGlobalRef(localRef));
- CHECK_JNI_EXCEPTION(jni, "error during NewGlobalRef");
- CHECK(globalRef, name.c_str());
- bool inserted = classes_.insert(std::make_pair(name, globalRef)).second;
- CHECK(inserted, "Duplicate class name");
-}
diff --git a/webrtc/examples/android/media_demo/jni/jni_helpers.h b/webrtc/examples/android/media_demo/jni/jni_helpers.h
deleted file mode 100644
index 3d8ff48..0000000
--- a/webrtc/examples/android/media_demo/jni/jni_helpers.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_EXAMPLES_ANDROID_MEDIA_DEMO_JNI_JNI_HELPERS_H_
-#define WEBRTC_EXAMPLES_ANDROID_MEDIA_DEMO_JNI_JNI_HELPERS_H_
-
-// TODO(henrike): this file contains duplication with regards to
-// talk/app/webrtc/java/jni/peerconnection_jni.cc. When/if code can be shared
-// between trunk/talk and trunk/webrtc remove the duplication.
-
-#include <android/log.h>
-#include <jni.h>
-
-#include <assert.h>
-#include <map>
-#include <string>
-
-#define TAG "WEBRTC-NATIVE"
-
-// Abort the process if |x| is false, emitting |msg| to logcat.
-#define CHECK(x, msg) \
- if (x) { \
- } else { \
- __android_log_print(ANDROID_LOG_ERROR, TAG, "%s:%d: %s", __FILE__, \
- __LINE__, msg); \
- assert(false); \
- }
-
-// Abort the process if |jni| has a Java exception pending, emitting |msg| to
-// logcat.
-#define CHECK_JNI_EXCEPTION(jni, msg) \
- if (0) { \
- } else { \
- if (jni->ExceptionCheck()) { \
- jni->ExceptionDescribe(); \
- jni->ExceptionClear(); \
- CHECK(0, msg); \
- } \
- }
-
-// JNIEnv-helper methods that CHECK success: no Java exception thrown and found
-// object/class/method/field is non-null.
-jmethodID GetMethodID(JNIEnv* jni, jclass c, const std::string& name,
- const char* signature);
-
-// Return a |jlong| that will automatically convert back to |ptr| when assigned
-// to a |uint64_t|
-jlong jlongFromPointer(void* ptr);
-
-// Given a (UTF-16) jstring return a new UTF-8 native string.
-std::string JavaToStdString(JNIEnv* jni, const jstring& j_string);
-
-// Android's FindClass() is trickier than usual because the app-specific
-// ClassLoader is not consulted when there is no app-specific frame on the
-// stack. Consequently, we only look up classes once in JNI_OnLoad.
-// http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
-class ClassReferenceHolder {
- public:
- ClassReferenceHolder(JNIEnv* jni, const char** classes, int size);
- ~ClassReferenceHolder();
-
- void FreeReferences(JNIEnv* jni);
-
- jclass GetClass(const std::string& name);
-
- private:
- void LoadClass(JNIEnv* jni, const std::string& name);
-
- std::map<std::string, jclass> classes_;
-};
-
-#endif // WEBRTC_EXAMPLES_ANDROID_MEDIA_DEMO_JNI_JNI_HELPERS_H_
diff --git a/webrtc/examples/android/media_demo/jni/on_load.cc b/webrtc/examples/android/media_demo/jni/on_load.cc
deleted file mode 100644
index 5827ee8..0000000
--- a/webrtc/examples/android/media_demo/jni/on_load.cc
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <jni.h>
-
-#include <assert.h>
-
-#include "webrtc/examples/android/media_demo/jni/jni_helpers.h"
-#include "webrtc/examples/android/media_demo/jni/voice_engine_jni.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-
-// Macro for native functions that can be found by way of jni-auto discovery.
-// Note extern "C" is needed for "discovery" of native methods to work.
-#define JOWW(rettype, name) \
- extern "C" rettype JNIEXPORT JNICALL Java_org_webrtc_webrtcdemo_##name
-
-static JavaVM* g_vm = NULL;
-
-extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* vm, void* reserved) {
- // Only called once.
- CHECK(!g_vm, "OnLoad called more than once");
- g_vm = vm;
- return JNI_VERSION_1_4;
-}
-
-JOWW(void, NativeWebRtcContextRegistry_register)(
- JNIEnv* jni,
- jclass,
- jobject context) {
- webrtc_examples::SetVoeDeviceObjects(g_vm);
- CHECK(webrtc::VoiceEngine::SetAndroidObjects(g_vm, context) == 0,
- "Failed to register android objects to voice engine");
-}
-
-JOWW(void, NativeWebRtcContextRegistry_unRegister)(
- JNIEnv* jni,
- jclass) {
- CHECK(webrtc::VoiceEngine::SetAndroidObjects(NULL, NULL) == 0,
- "Failed to unregister android objects from voice engine");
- webrtc_examples::ClearVoeDeviceObjects();
-}
diff --git a/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc b/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc
deleted file mode 100644
index 79d6cbc..0000000
--- a/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc
+++ /dev/null
@@ -1,423 +0,0 @@
-/*
- * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file contains JNI for the voice engine interfaces.
-// The native functions are found using jni's auto discovery.
-
-#include "webrtc/examples/android/media_demo/jni/voice_engine_jni.h"
-
-#include <map>
-#include <string>
-
-#include "webrtc/base/arraysize.h"
-#include "webrtc/examples/android/media_demo/jni/jni_helpers.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
-#include "webrtc/voice_engine/include/voe_audio_processing.h"
-#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_codec.h"
-#include "webrtc/voice_engine/include/voe_file.h"
-#include "webrtc/voice_engine/include/voe_hardware.h"
-#include "webrtc/voice_engine/include/voe_network.h"
-#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
-
-// Macro for native functions that can be found by way of jni-auto discovery.
-// Note extern "C" is needed for "discovery" of native methods to work.
-#define JOWW(rettype, name) \
- extern "C" rettype JNIEXPORT JNICALL Java_org_webrtc_webrtcdemo_##name
-
-namespace {
-
-static JavaVM* g_vm = NULL;
-static ClassReferenceHolder* g_class_reference_holder = NULL;
-
-jclass GetClass(JNIEnv* jni, const char* name) {
- CHECK(g_class_reference_holder, "Class reference holder NULL");
- return g_class_reference_holder->GetClass(name);
-}
-
-static const char* g_classes[] = {"org/webrtc/webrtcdemo/CodecInst"};
-
-template<typename T>
-void ReleaseSubApi(T instance) {
- CHECK(instance->Release() >= 0, "failed to release instance")
-}
-
-class VoiceEngineData {
- public:
- VoiceEngineData()
- : ve(webrtc::VoiceEngine::Create()),
- base(webrtc::VoEBase::GetInterface(ve)),
- codec(webrtc::VoECodec::GetInterface(ve)),
- file(webrtc::VoEFile::GetInterface(ve)),
- netw(webrtc::VoENetwork::GetInterface(ve)),
- apm(webrtc::VoEAudioProcessing::GetInterface(ve)),
- volume(webrtc::VoEVolumeControl::GetInterface(ve)),
- hardware(webrtc::VoEHardware::GetInterface(ve)),
- rtp(webrtc::VoERTP_RTCP::GetInterface(ve)) {
- CHECK(ve != NULL, "Voice engine instance failed to be created");
- CHECK(base != NULL, "Failed to acquire base interface");
- CHECK(codec != NULL, "Failed to acquire codec interface");
- CHECK(file != NULL, "Failed to acquire file interface");
- CHECK(netw != NULL, "Failed to acquire netw interface");
- CHECK(apm != NULL, "Failed to acquire apm interface");
- CHECK(volume != NULL, "Failed to acquire volume interface");
- CHECK(hardware != NULL, "Failed to acquire hardware interface");
- CHECK(rtp != NULL, "Failed to acquire rtp interface");
- }
-
- ~VoiceEngineData() {
- CHECK(channel_transports_.empty(),
- "VoE transports must be deleted before terminating");
- CHECK(base->Terminate() == 0, "VoE failed to terminate");
- ReleaseSubApi(base);
- ReleaseSubApi(codec);
- ReleaseSubApi(file);
- ReleaseSubApi(netw);
- ReleaseSubApi(apm);
- ReleaseSubApi(volume);
- ReleaseSubApi(hardware);
- ReleaseSubApi(rtp);
- webrtc::VoiceEngine* ve_instance = ve;
- CHECK(webrtc::VoiceEngine::Delete(ve_instance), "VoE failed to be deleted