modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 5d02b0e..bc4981b 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -14,7 +14,7 @@
#include "webrtc/audio_receive_stream.h"
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/voice_engine/include/voe_base.h"
namespace webrtc {
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 594ddf5..f83d320 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -25,9 +25,9 @@
#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/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"
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index c37b83b..95fd32e 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -19,7 +19,7 @@
#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/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"
diff --git a/webrtc/call/congestion_controller.cc b/webrtc/call/congestion_controller.cc
index 1ec361e..5270415 100644
--- a/webrtc/call/congestion_controller.cc
+++ b/webrtc/call/congestion_controller.cc
@@ -20,8 +20,8 @@
#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"
diff --git a/webrtc/common_video/video_render_frames.cc b/webrtc/common_video/video_render_frames.cc
index f4ece5e..a7db15c 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"
diff --git a/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc b/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc
index 79d6cbc..7d5b99b 100644
--- a/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc
+++ b/webrtc/examples/android/media_demo/jni/voice_engine_jni.cc
@@ -18,7 +18,7 @@
#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/modules/utility/include/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"
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 839a143..13f9a5d 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -45,7 +45,7 @@
config("audio_coding_config") {
include_dirs = [
"main/include",
- "../interface",
+ "../include",
]
}
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
index 9e812b2..4b080ba 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
@@ -24,7 +24,7 @@
#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
index dbbf8f8..3aee344 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
@@ -33,7 +33,7 @@
#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
diff --git a/webrtc/modules/audio_coding/main/acm2/call_statistics.h b/webrtc/modules/audio_coding/main/acm2/call_statistics.h
index 2aece0f..e2df921 100644
--- a/webrtc/modules/audio_coding/main/acm2/call_statistics.h
+++ b/webrtc/modules/audio_coding/main/acm2/call_statistics.h
@@ -12,7 +12,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_CALL_STATISTICS_H_
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
//
// This class is for book keeping of calls to ACM. It is not useful to log API
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
index c6942ec..6b50dd0 100644
--- a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
+++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h
@@ -12,7 +12,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/audio_coding_module.gypi
index 6fb37d2..088a319 100644
--- a/webrtc/modules/audio_coding/main/audio_coding_module.gypi
+++ b/webrtc/modules/audio_coding/main/audio_coding_module.gypi
@@ -81,13 +81,13 @@
],
'include_dirs': [
'include',
- '../../interface',
+ '../../include',
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'include',
- '../../interface',
+ '../../include',
'<(webrtc_root)',
],
},
diff --git a/webrtc/modules/audio_coding/main/include/audio_coding_module.h b/webrtc/modules/audio_coding/main/include/audio_coding_module.h
index 1792a25..00f8d54 100644
--- a/webrtc/modules/audio_coding/main/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/main/include/audio_coding_module.h
@@ -18,7 +18,7 @@
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h b/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h
index 489df40..1ca6f9d 100644
--- a/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h
+++ b/webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h
@@ -13,7 +13,7 @@
#include <map>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/test/Channel.h b/webrtc/modules/audio_coding/main/test/Channel.h
index 39d4dab..ff6937e 100644
--- a/webrtc/modules/audio_coding/main/test/Channel.h
+++ b/webrtc/modules/audio_coding/main/test/Channel.h
@@ -14,7 +14,7 @@
#include <stdio.h>
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.cc b/webrtc/modules/audio_coding/main/test/PCMFile.cc
index d0ae783..0466e02 100644
--- a/webrtc/modules/audio_coding/main/test/PCMFile.cc
+++ b/webrtc/modules/audio_coding/main/test/PCMFile.cc
@@ -15,7 +15,7 @@
#include <string.h>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/test/PCMFile.h b/webrtc/modules/audio_coding/main/test/PCMFile.h
index 8353898..785ed66 100644
--- a/webrtc/modules/audio_coding/main/test/PCMFile.h
+++ b/webrtc/modules/audio_coding/main/test/PCMFile.h
@@ -16,7 +16,7 @@
#include <string>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/main/test/RTPFile.h b/webrtc/modules/audio_coding/main/test/RTPFile.h
index c79b63e..6bad755 100644
--- a/webrtc/modules/audio_coding/main/test/RTPFile.h
+++ b/webrtc/modules/audio_coding/main/test/RTPFile.h
@@ -15,7 +15,7 @@
#include <queue>
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
index 5b1e07e..857381d 100644
--- a/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
+++ b/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
@@ -17,7 +17,7 @@
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
index 20b10a3..9d0986a 100644
--- a/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
@@ -14,7 +14,7 @@
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/utility.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
index d3f6fa6..0252d1c 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
@@ -20,7 +20,7 @@
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/delay_manager.cc b/webrtc/modules/audio_coding/neteq/delay_manager.cc
index 5140c06..806d02b 100644
--- a/webrtc/modules/audio_coding/neteq/delay_manager.cc
+++ b/webrtc/modules/audio_coding/neteq/delay_manager.cc
@@ -17,7 +17,7 @@
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/nack.cc b/webrtc/modules/audio_coding/neteq/nack.cc
index fd3d762..011914b 100644
--- a/webrtc/modules/audio_coding/neteq/nack.cc
+++ b/webrtc/modules/audio_coding/neteq/nack.cc
@@ -15,7 +15,7 @@
#include <algorithm> // For std::max.
#include "webrtc/base/checks.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 9ed0fc8..d9ae15e 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -42,7 +42,7 @@
#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
// Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
diff --git a/webrtc/modules/audio_coding/neteq/packet.h b/webrtc/modules/audio_coding/neteq/packet.h
index 723ed8b..64b325e 100644
--- a/webrtc/modules/audio_coding/neteq/packet.h
+++ b/webrtc/modules/audio_coding/neteq/packet.h
@@ -13,7 +13,7 @@
#include <list>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/rtcp.cc b/webrtc/modules/audio_coding/neteq/rtcp.cc
index cf8e028..7ef40bc 100644
--- a/webrtc/modules/audio_coding/neteq/rtcp.cc
+++ b/webrtc/modules/audio_coding/neteq/rtcp.cc
@@ -15,7 +15,7 @@
#include <algorithm>
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
index 3fbce8b..56ed72f 100644
--- a/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
+++ b/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
@@ -14,7 +14,7 @@
#include <map>
#include <stdio.h>
#include "webrtc/typedefs.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
enum stereoModes {
stereoModeMono,
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_sink.h b/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
index 3bd2df5..489a8b2 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
+++ b/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
@@ -12,7 +12,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
index 0a41c6e..c9fe11f 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
@@ -14,7 +14,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
namespace test {
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index adce1cf..7a52408 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -34,7 +34,7 @@
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/rtp_file_reader.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/webrtc/modules/audio_coding/neteq/tools/packet.cc b/webrtc/modules/audio_coding/neteq/tools/packet.cc
index b8b27af..2b2fcc2 100644
--- a/webrtc/modules/audio_coding/neteq/tools/packet.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/packet.cc
@@ -12,8 +12,8 @@
#include <string.h>
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
namespace webrtc {
namespace test {
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
index 9b17ba8..dad72ea 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -18,7 +18,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
// Files generated at build-time by the protobuf compiler.
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
index 7150bcf..90d5931 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -16,7 +16,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
index e3a829b..b7a3109 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -20,7 +20,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/test/rtp_file_reader.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
index cd7d7e8..2febf68 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -18,7 +18,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h b/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
index 6c16192..53371be 100644
--- a/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
+++ b/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -12,7 +12,7 @@
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_conference_mixer/BUILD.gn b/webrtc/modules/audio_conference_mixer/BUILD.gn
index 3b9e276..36391c7 100644
--- a/webrtc/modules/audio_conference_mixer/BUILD.gn
+++ b/webrtc/modules/audio_conference_mixer/BUILD.gn
@@ -9,15 +9,15 @@
config("audio_conference_mixer_config") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
include_dirs = [
- "interface",
- "../interface",
+ "include",
+ "../include",
]
}
source_set("audio_conference_mixer") {
sources = [
- "interface/audio_conference_mixer.h",
- "interface/audio_conference_mixer_defines.h",
+ "include/audio_conference_mixer.h",
+ "include/audio_conference_mixer_defines.h",
"source/audio_conference_mixer_impl.cc",
"source/audio_conference_mixer_impl.h",
"source/audio_frame_manipulator.cc",
diff --git a/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
index 5aa3cc4..9d71795 100644
--- a/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
+++ b/webrtc/modules/audio_conference_mixer/audio_conference_mixer.gypi
@@ -17,8 +17,8 @@
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
- 'interface/audio_conference_mixer.h',
- 'interface/audio_conference_mixer_defines.h',
+ 'include/audio_conference_mixer.h',
+ 'include/audio_conference_mixer_defines.h',
'source/audio_frame_manipulator.cc',
'source/audio_frame_manipulator.h',
'source/memory_pool.h',
diff --git a/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
new file mode 100644
index 0000000..7370442
--- /dev/null
+++ b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2011 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_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
+
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+class AudioMixerOutputReceiver;
+class MixerParticipant;
+class Trace;
+
+class AudioConferenceMixer : public Module
+{
+public:
+ enum {kMaximumAmountOfMixedParticipants = 3};
+ enum Frequency
+ {
+ kNbInHz = 8000,
+ kWbInHz = 16000,
+ kSwbInHz = 32000,
+ kFbInHz = 48000,
+ kLowestPossible = -1,
+ kDefaultFrequency = kWbInHz
+ };
+
+ // Factory method. Constructor disabled.
+ static AudioConferenceMixer* Create(int id);
+ virtual ~AudioConferenceMixer() {}
+
+ // Module functions
+ int64_t TimeUntilNextProcess() override = 0;
+ int32_t Process() override = 0;
+
+ // Register/unregister a callback class for receiving the mixed audio.
+ virtual int32_t RegisterMixedStreamCallback(
+ AudioMixerOutputReceiver* receiver) = 0;
+ virtual int32_t UnRegisterMixedStreamCallback() = 0;
+
+ // Add/remove participants as candidates for mixing.
+ virtual int32_t SetMixabilityStatus(MixerParticipant* participant,
+ bool mixable) = 0;
+ // Returns true if a participant is a candidate for mixing.
+ virtual bool MixabilityStatus(
+ const MixerParticipant& participant) const = 0;
+
+ // Inform the mixer that the participant should always be mixed and not
+ // count toward the number of mixed participants. Note that a participant
+ // must have been added to the mixer (by calling SetMixabilityStatus())
+ // before this function can be successfully called.
+ virtual int32_t SetAnonymousMixabilityStatus(
+ MixerParticipant* participant, bool mixable) = 0;
+ // Returns true if the participant is mixed anonymously.
+ virtual bool AnonymousMixabilityStatus(
+ const MixerParticipant& participant) const = 0;
+
+ // Set the minimum sampling frequency at which to mix. The mixing algorithm
+ // may still choose to mix at a higher samling frequency to avoid
+ // downsampling of audio contributing to the mixed audio.
+ virtual int32_t SetMinimumMixingFrequency(Frequency freq) = 0;
+
+protected:
+ AudioConferenceMixer() {}
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
diff --git a/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
new file mode 100644
index 0000000..5d58f42
--- /dev/null
+++ b/webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h
@@ -0,0 +1,60 @@
+/*
+ * Copyright (c) 2011 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_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+class MixHistory;
+
+// A callback class that all mixer participants must inherit from/implement.
+class MixerParticipant
+{
+public:
+ // The implementation of this function should update audioFrame with new
+ // audio every time it's called.
+ //
+ // If it returns -1, the frame will not be added to the mix.
+ virtual int32_t GetAudioFrame(int32_t id,
+ AudioFrame* audioFrame) = 0;
+
+ // Returns true if the participant was mixed this mix iteration.
+ bool IsMixed() const;
+
+ // This function specifies the sampling frequency needed for the AudioFrame
+ // for future GetAudioFrame(..) calls.
+ virtual int32_t NeededFrequency(int32_t id) const = 0;
+
+ MixHistory* _mixHistory;
+protected:
+ MixerParticipant();
+ virtual ~MixerParticipant();
+};
+
+class AudioMixerOutputReceiver
+{
+public:
+ // This callback function provides the mixed audio for this mix iteration.
+ // Note that uniqueAudioFrames is an array of AudioFrame pointers with the
+ // size according to the size parameter.
+ virtual void NewMixedAudio(const int32_t id,
+ const AudioFrame& generalAudioFrame,
+ const AudioFrame** uniqueAudioFrames,
+ const uint32_t size) = 0;
+protected:
+ AudioMixerOutputReceiver() {}
+ virtual ~AudioMixerOutputReceiver() {}
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
index 7ff3957..7536ce9 100644
--- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
+++ b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
@@ -8,12 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#pragma message("WARNING: audio_conference_mixer/interface is DEPRECATED; use include")
+
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
class AudioMixerOutputReceiver;
@@ -74,4 +76,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
index d15b7fc..057c37f 100644
--- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
+++ b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
@@ -8,10 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-#include "webrtc/modules/interface/module_common_types.h"
+#pragma message("WARNING: audio_conference_mixer/interface is DEPRECATED; use include")
+
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -57,4 +59,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
+#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
index 2d2cf9d..0ac9eae 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h"
#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
index bc9a27e..2466112 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
+++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
@@ -16,10 +16,10 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
#include "webrtc/modules/audio_conference_mixer/source/memory_pool.h"
#include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
class AudioProcessing;
diff --git a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
index 636698e..9c5d3b9 100644
--- a/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
+++ b/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
@@ -9,7 +9,7 @@
*/
#include "webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace {
diff --git a/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc b/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
index d4fbd20..293bfa0 100644
--- a/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
+++ b/webrtc/modules/audio_conference_mixer/test/audio_conference_mixer_unittest.cc
@@ -10,8 +10,8 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn
index 8875178..b5e503a 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -10,7 +10,7 @@
config("audio_device_config") {
include_dirs = [
- "../interface",
+ "../include",
"include",
"dummy", # Contains dummy audio device implementations.
]
diff --git a/webrtc/modules/audio_device/android/audio_manager.cc b/webrtc/modules/audio_device/android/audio_manager.cc
index 260e793..522010e 100644
--- a/webrtc/modules/audio_device/android/audio_manager.cc
+++ b/webrtc/modules/audio_device/android/audio_manager.cc
@@ -16,7 +16,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_device/android/audio_common.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
#define TAG "AudioManager"
#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
diff --git a/webrtc/modules/audio_device/android/audio_manager.h b/webrtc/modules/audio_device/android/audio_manager.h
index 9cceaac..26caf61 100644
--- a/webrtc/modules/audio_device/android/audio_manager.h
+++ b/webrtc/modules/audio_device/android/audio_manager.h
@@ -19,8 +19,8 @@
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_device/android/audio_record_jni.h b/webrtc/modules/audio_device/android/audio_record_jni.h
index efd5164..a847180 100644
--- a/webrtc/modules/audio_device/android/audio_record_jni.h
+++ b/webrtc/modules/audio_device/android/audio_record_jni.h
@@ -17,8 +17,8 @@
#include "webrtc/modules/audio_device/android/audio_manager.h"
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_device/android/audio_track_jni.h b/webrtc/modules/audio_device/android/audio_track_jni.h
index 43bfcad..067dc6c 100644
--- a/webrtc/modules/audio_device/android/audio_track_jni.h
+++ b/webrtc/modules/audio_device/android/audio_track_jni.h
@@ -18,8 +18,8 @@
#include "webrtc/modules/audio_device/android/audio_manager.h"
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_device/android/build_info.cc b/webrtc/modules/audio_device/android/build_info.cc
index cb5dc29..6289697 100644
--- a/webrtc/modules/audio_device/android/build_info.cc
+++ b/webrtc/modules/audio_device/android/build_info.cc
@@ -10,7 +10,7 @@
#include "webrtc/modules/audio_device/android/build_info.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_device/android/build_info.h b/webrtc/modules/audio_device/android/build_info.h
index d9b2871..1490fa0 100644
--- a/webrtc/modules/audio_device/android/build_info.h
+++ b/webrtc/modules/audio_device/android/build_info.h
@@ -14,7 +14,7 @@
#include <jni.h>
#include <string>
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_device/android/ensure_initialized.cc b/webrtc/modules/audio_device/android/ensure_initialized.cc
index e8197b7..11b2b25 100644
--- a/webrtc/modules/audio_device/android/ensure_initialized.cc
+++ b/webrtc/modules/audio_device/android/ensure_initialized.cc
@@ -18,7 +18,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_device/android/audio_record_jni.h"
#include "webrtc/modules/audio_device/android/audio_track_jni.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
namespace webrtc {
namespace audiodevicemodule {
diff --git a/webrtc/modules/audio_device/android/opensles_player.h b/webrtc/modules/audio_device/android/opensles_player.h
index d96388b..4c4d724 100644
--- a/webrtc/modules/audio_device/android/opensles_player.h
+++ b/webrtc/modules/audio_device/android/opensles_player.h
@@ -22,7 +22,7 @@
#include "webrtc/modules/audio_device/android/opensles_common.h"
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_device/audio_device.gypi b/webrtc/modules/audio_device/audio_device.gypi
index 0678d33..c3b6fba 100644
--- a/webrtc/modules/audio_device/audio_device.gypi
+++ b/webrtc/modules/audio_device/audio_device.gypi
@@ -20,13 +20,13 @@
],
'include_dirs': [
'.',
- '../interface',
+ '../include',
'include',
'dummy', # Contains dummy audio device implementations.
],
'direct_dependent_settings': {
'include_dirs': [
- '../interface',
+ '../include',
'include',
],
},
diff --git a/webrtc/modules/audio_device/include/audio_device.h b/webrtc/modules/audio_device/include/audio_device.h
index c2c2b88..15e0873 100644
--- a/webrtc/modules/audio_device/include/audio_device.h
+++ b/webrtc/modules/audio_device/include/audio_device.h
@@ -12,7 +12,7 @@
#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
#include "webrtc/modules/audio_device/include/audio_device_defines.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index f26e9f1..5bf6b3b 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -21,7 +21,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_device/fine_audio_buffer.h"
-#include "webrtc/modules/utility/interface/helpers_ios.h"
+#include "webrtc/modules/utility/include/helpers_ios.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_device/test/audio_device_test_defines.h b/webrtc/modules/audio_device/test/audio_device_test_defines.h
index 5068646b..cc8e3e3 100644
--- a/webrtc/modules/audio_device/test/audio_device_test_defines.h
+++ b/webrtc/modules/audio_device/test/audio_device_test_defines.h
@@ -13,7 +13,7 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/trace.h"
#ifdef _WIN32
diff --git a/webrtc/modules/audio_processing/agc/agc.cc b/webrtc/modules/audio_processing/agc/agc.cc
index 706b963..fc78f07 100644
--- a/webrtc/modules/audio_processing/agc/agc.cc
+++ b/webrtc/modules/audio_processing/agc/agc.cc
@@ -19,7 +19,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_processing/agc/histogram.h"
#include "webrtc/modules/audio_processing/agc/utility.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
namespace {
diff --git a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
index 867022d..e4b753b 100644
--- a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
@@ -19,7 +19,7 @@
#include "webrtc/modules/audio_processing/agc/gain_map_internal.h"
#include "webrtc/modules/audio_processing/gain_control_impl.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/agc/agc_unittest.cc b/webrtc/modules/audio_processing/agc/agc_unittest.cc
index 66a8a2b..25b99d8 100644
--- a/webrtc/modules/audio_processing/agc/agc_unittest.cc
+++ b/webrtc/modules/audio_processing/agc/agc_unittest.cc
@@ -13,7 +13,7 @@
#include "gmock/gmock.h"
#include "gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/tools/agc/test_utils.h"
diff --git a/webrtc/modules/audio_processing/agc/histogram.cc b/webrtc/modules/audio_processing/agc/histogram.cc
index 1d3035f..5c66727 100644
--- a/webrtc/modules/audio_processing/agc/histogram.cc
+++ b/webrtc/modules/audio_processing/agc/histogram.cc
@@ -13,7 +13,7 @@
#include <cmath>
#include <cstring>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/agc/mock_agc.h b/webrtc/modules/audio_processing/agc/mock_agc.h
index 13dbd2e..e362200 100644
--- a/webrtc/modules/audio_processing/agc/mock_agc.h
+++ b/webrtc/modules/audio_processing/agc/mock_agc.h
@@ -14,7 +14,7 @@
#include "webrtc/modules/audio_processing/agc/agc.h"
#include "gmock/gmock.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/audio_buffer.h b/webrtc/modules/audio_processing/audio_buffer.h
index 864633f..48c9488 100644
--- a/webrtc/modules/audio_processing/audio_buffer.h
+++ b/webrtc/modules/audio_processing/audio_buffer.h
@@ -15,7 +15,7 @@
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/splitting_filter.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/scoped_vector.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index c657415..668ec11 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -36,7 +36,7 @@
#include "webrtc/modules/audio_processing/processing_component.h"
#include "webrtc/modules/audio_processing/transient/transient_suppressor.h"
#include "webrtc/modules/audio_processing/voice_detection_impl.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
diff --git a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
index f4c36d0..ed20daa 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -14,7 +14,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/config.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
using ::testing::Invoke;
using ::testing::Return;
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
index eac165c..eff791d 100644
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -24,7 +24,7 @@
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/webrtc/modules/audio_processing/test/process_test.cc b/webrtc/modules/audio_processing/test/process_test.cc
index 555d9d6..1383bbe 100644
--- a/webrtc/modules/audio_processing/test/process_test.cc
+++ b/webrtc/modules/audio_processing/test/process_test.cc
@@ -22,7 +22,7 @@
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/test/testsupport/fileutils.h"
diff --git a/webrtc/modules/audio_processing/test/test_utils.h b/webrtc/modules/audio_processing/test/test_utils.h
index f53d8ac..93a0138 100644
--- a/webrtc/modules/audio_processing/test/test_utils.h
+++ b/webrtc/modules/audio_processing/test/test_utils.h
@@ -22,7 +22,7 @@
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
index 506abaf..b7b7595 100644
--- a/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
+++ b/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
@@ -19,7 +19,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/modules/audio_processing/agc/agc.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/audio_processing/typing_detection.h b/webrtc/modules/audio_processing/typing_detection.h
index 5fa6456..40608f8 100644
--- a/webrtc/modules/audio_processing/typing_detection.h
+++ b/webrtc/modules/audio_processing/typing_detection.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/vad/pitch_based_vad.cc b/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
index 39ec37e..fce144d 100644
--- a/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
+++ b/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
@@ -18,7 +18,7 @@
#include "webrtc/modules/audio_processing/vad/common.h"
#include "webrtc/modules/audio_processing/vad/noise_gmm_tables.h"
#include "webrtc/modules/audio_processing/vad/voice_gmm_tables.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/vad/standalone_vad.cc b/webrtc/modules/audio_processing/vad/standalone_vad.cc
index 468b8ff..1209526 100644
--- a/webrtc/modules/audio_processing/vad/standalone_vad.cc
+++ b/webrtc/modules/audio_processing/vad/standalone_vad.cc
@@ -12,8 +12,8 @@
#include <assert.h>
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc b/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
index 942008e..5462d05 100644
--- a/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
+++ b/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
@@ -14,7 +14,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
index 8535d1f..1a59559 100644
--- a/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
+++ b/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
@@ -23,7 +23,7 @@
#include "webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/source/structs.h"
}
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
index f509af4..a8a4ead 100644
--- a/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
+++ b/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
@@ -21,7 +21,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_processing/vad/common.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc b/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
index 8857ee4..d779562 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
@@ -14,7 +14,7 @@
#include <algorithm>
#include <utility>
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
index 72831c7..2b9e589 100644
--- a/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
+++ b/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
@@ -14,7 +14,7 @@
#include <vector>
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
using webrtc::RtcpBandwidthObserver;
using webrtc::BitrateObserver;
diff --git a/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
index bb53288..5b20287 100644
--- a/webrtc/modules/bitrate_controller/include/bitrate_controller.h
+++ b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
@@ -17,8 +17,8 @@
#include <map>
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
index 40061d3..baf8857 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
@@ -15,7 +15,7 @@
#include <deque>
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
diff --git a/webrtc/modules/include/module.h b/webrtc/modules/include/module.h
new file mode 100644
index 0000000..d02aa95
--- /dev/null
+++ b/webrtc/modules/include/module.h
@@ -0,0 +1,81 @@
+/*
+ * 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_MODULES_INCLUDE_MODULE_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class ProcessThread;
+
+class Module {
+ public:
+ // Returns the number of milliseconds until the module wants a worker
+ // thread to call Process.
+ // This method is called on the same worker thread as Process will
+ // be called on.
+ // TODO(tommi): Almost all implementations of this function, need to know
+ // the current tick count. Consider passing it as an argument. It could
+ // also improve the accuracy of when the next callback occurs since the
+ // thread that calls Process() will also have it's tick count reference
+ // which might not match with what the implementations use.
+ virtual int64_t TimeUntilNextProcess() = 0;
+
+ // Process any pending tasks such as timeouts.
+ // Called on a worker thread.
+ virtual int32_t Process() = 0;
+
+ // This method is called when the module is attached to a *running* process
+ // thread or detached from one. In the case of detaching, |process_thread|
+ // will be nullptr.
+ //
+ // This method will be called in the following cases:
+ //
+ // * Non-null process_thread:
+ // * ProcessThread::RegisterModule() is called while the thread is running.
+ // * ProcessThread::Start() is called and RegisterModule has previously
+ // been called. The thread will be started immediately after notifying
+ // all modules.
+ //
+ // * Null process_thread:
+ // * ProcessThread::DeRegisterModule() is called while the thread is
+ // running.
+ // * ProcessThread::Stop() was called and the thread has been stopped.
+ //
+ // NOTE: This method is not called from the worker thread itself, but from
+ // the thread that registers/deregisters the module or calls Start/Stop.
+ virtual void ProcessThreadAttached(ProcessThread* process_thread) {}
+
+ protected:
+ virtual ~Module() {}
+};
+
+// Reference counted version of the Module interface.
+class RefCountedModule : public Module {
+ public:
+ // Increase the reference count by one.
+ // Returns the incremented reference count.
+ virtual int32_t AddRef() const = 0;
+
+ // Decrease the reference count by one.
+ // Returns the decreased reference count.
+ // Returns 0 if the last reference was just released.
+ // When the reference count reaches 0 the object will self-destruct.
+ virtual int32_t Release() const = 0;
+
+ protected:
+ ~RefCountedModule() override = default;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_INCLUDE_MODULE_H_
diff --git a/webrtc/modules/include/module_common_types.h b/webrtc/modules/include/module_common_types.h
new file mode 100644
index 0000000..fb63581
--- /dev/null
+++ b/webrtc/modules/include/module_common_types.h
@@ -0,0 +1,810 @@
+/*
+ * 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_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+
+#include <assert.h>
+#include <string.h> // memcpy
+
+#include <algorithm>
+#include <limits>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_types.h"
+#include "webrtc/common_video/rotation.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct RTPAudioHeader {
+ uint8_t numEnergy; // number of valid entries in arrOfEnergy
+ uint8_t arrOfEnergy[kRtpCsrcSize]; // one energy byte (0-9) per channel
+ bool isCNG; // is this CNG
+ uint8_t channel; // number of channels 2 = stereo
+};
+
+const int16_t kNoPictureId = -1;
+const int16_t kMaxOneBytePictureId = 0x7F; // 7 bits
+const int16_t kMaxTwoBytePictureId = 0x7FFF; // 15 bits
+const int16_t kNoTl0PicIdx = -1;
+const uint8_t kNoTemporalIdx = 0xFF;
+const uint8_t kNoSpatialIdx = 0xFF;
+const uint8_t kNoGofIdx = 0xFF;
+const size_t kMaxVp9RefPics = 3;
+const size_t kMaxVp9FramesInGof = 0xFF; // 8 bits
+const size_t kMaxVp9NumberOfSpatialLayers = 8;
+const int kNoKeyIdx = -1;
+
+struct RTPVideoHeaderVP8 {
+ void InitRTPVideoHeaderVP8() {
+ nonReference = false;
+ pictureId = kNoPictureId;
+ tl0PicIdx = kNoTl0PicIdx;
+ temporalIdx = kNoTemporalIdx;
+ layerSync = false;
+ keyIdx = kNoKeyIdx;
+ partitionId = 0;
+ beginningOfPartition = false;
+ }
+
+ bool nonReference; // Frame is discardable.
+ int16_t pictureId; // Picture ID index, 15 bits;
+ // kNoPictureId if PictureID does not exist.
+ int16_t tl0PicIdx; // TL0PIC_IDX, 8 bits;
+ // kNoTl0PicIdx means no value provided.
+ uint8_t temporalIdx; // Temporal layer index, or kNoTemporalIdx.
+ bool layerSync; // This frame is a layer sync frame.
+ // Disabled if temporalIdx == kNoTemporalIdx.
+ int keyIdx; // 5 bits; kNoKeyIdx means not used.
+ int partitionId; // VP8 partition ID
+ bool beginningOfPartition; // True if this packet is the first
+ // in a VP8 partition. Otherwise false
+};
+
+enum TemporalStructureMode {
+ kTemporalStructureMode1, // 1 temporal layer structure - i.e., IPPP...
+ kTemporalStructureMode2, // 2 temporal layers 0-1-0-1...
+ kTemporalStructureMode3 // 3 temporal layers 0-2-1-2-0-2-1-2...
+};
+
+struct GofInfoVP9 {
+ void SetGofInfoVP9(TemporalStructureMode tm) {
+ switch (tm) {
+ case kTemporalStructureMode1:
+ num_frames_in_gof = 1;
+ temporal_idx[0] = 0;
+ temporal_up_switch[0] = false;
+ num_ref_pics[0] = 1;
+ pid_diff[0][0] = 1;
+ break;
+ case kTemporalStructureMode2:
+ num_frames_in_gof = 2;
+ temporal_idx[0] = 0;
+ temporal_up_switch[0] = false;
+ num_ref_pics[0] = 1;
+ pid_diff[0][0] = 2;
+
+ temporal_idx[1] = 1;
+ temporal_up_switch[1] = true;
+ num_ref_pics[1] = 1;
+ pid_diff[1][0] = 1;
+ break;
+ case kTemporalStructureMode3:
+ num_frames_in_gof = 4;
+ temporal_idx[0] = 0;
+ temporal_up_switch[0] = false;
+ num_ref_pics[0] = 1;
+ pid_diff[0][0] = 4;
+
+ temporal_idx[1] = 2;
+ temporal_up_switch[1] = true;
+ num_ref_pics[1] = 1;
+ pid_diff[1][0] = 1;
+
+ temporal_idx[2] = 1;
+ temporal_up_switch[2] = true;
+ num_ref_pics[2] = 1;
+ pid_diff[2][0] = 2;
+
+ temporal_idx[3] = 2;
+ temporal_up_switch[3] = false;
+ num_ref_pics[3] = 2;
+ pid_diff[3][0] = 1;
+ pid_diff[3][1] = 2;
+ break;
+ default:
+ assert(false);
+ }
+ }
+
+ void CopyGofInfoVP9(const GofInfoVP9& src) {
+ num_frames_in_gof = src.num_frames_in_gof;
+ for (size_t i = 0; i < num_frames_in_gof; ++i) {
+ temporal_idx[i] = src.temporal_idx[i];
+ temporal_up_switch[i] = src.temporal_up_switch[i];
+ num_ref_pics[i] = src.num_ref_pics[i];
+ for (size_t r = 0; r < num_ref_pics[i]; ++r) {
+ pid_diff[i][r] = src.pid_diff[i][r];
+ }
+ }
+ }
+
+ size_t num_frames_in_gof;
+ uint8_t temporal_idx[kMaxVp9FramesInGof];
+ bool temporal_up_switch[kMaxVp9FramesInGof];
+ size_t num_ref_pics[kMaxVp9FramesInGof];
+ int16_t pid_diff[kMaxVp9FramesInGof][kMaxVp9RefPics];
+};
+
+struct RTPVideoHeaderVP9 {
+ void InitRTPVideoHeaderVP9() {
+ inter_pic_predicted = false;
+ flexible_mode = false;
+ beginning_of_frame = false;
+ end_of_frame = false;
+ ss_data_available = false;
+ picture_id = kNoPictureId;
+ max_picture_id = kMaxTwoBytePictureId;
+ tl0_pic_idx = kNoTl0PicIdx;
+ temporal_idx = kNoTemporalIdx;
+ spatial_idx = kNoSpatialIdx;
+ temporal_up_switch = false;
+ inter_layer_predicted = false;
+ gof_idx = kNoGofIdx;
+ num_ref_pics = 0;
+ num_spatial_layers = 1;
+ }
+
+ bool inter_pic_predicted; // This layer frame is dependent on previously
+ // coded frame(s).
+ bool flexible_mode; // This frame is in flexible mode.
+ bool beginning_of_frame; // True if this packet is the first in a VP9 layer
+ // frame.
+ bool end_of_frame; // True if this packet is the last in a VP9 layer frame.
+ bool ss_data_available; // True if SS data is available in this payload
+ // descriptor.
+ int16_t picture_id; // PictureID index, 15 bits;
+ // kNoPictureId if PictureID does not exist.
+ int16_t max_picture_id; // Maximum picture ID index; either 0x7F or 0x7FFF;
+ int16_t tl0_pic_idx; // TL0PIC_IDX, 8 bits;
+ // kNoTl0PicIdx means no value provided.
+ uint8_t temporal_idx; // Temporal layer index, or kNoTemporalIdx.
+ uint8_t spatial_idx; // Spatial layer index, or kNoSpatialIdx.
+ bool temporal_up_switch; // True if upswitch to higher frame rate is possible
+ // starting from this frame.
+ bool inter_layer_predicted; // Frame is dependent on directly lower spatial
+ // layer frame.
+
+ uint8_t gof_idx; // Index to predefined temporal frame info in SS data.
+
+ size_t num_ref_pics; // Number of reference pictures used by this layer
+ // frame.
+ int16_t pid_diff[kMaxVp9RefPics]; // P_DIFF signaled to derive the PictureID
+ // of the reference pictures.
+ int16_t ref_picture_id[kMaxVp9RefPics]; // PictureID of reference pictures.
+
+ // SS data.
+ size_t num_spatial_layers; // Always populated.
+ bool spatial_layer_resolution_present;
+ uint16_t width[kMaxVp9NumberOfSpatialLayers];
+ uint16_t height[kMaxVp9NumberOfSpatialLayers];
+ GofInfoVP9 gof;
+};
+
+// The packetization types that we support: single, aggregated, and fragmented.
+enum H264PacketizationTypes {
+ kH264SingleNalu, // This packet contains a single NAL unit.
+ kH264StapA, // This packet contains STAP-A (single time
+ // aggregation) packets. If this packet has an
+ // associated NAL unit type, it'll be for the
+ // first such aggregated packet.
+ kH264FuA, // This packet contains a FU-A (fragmentation
+ // unit) packet, meaning it is a part of a frame
+ // that was too large to fit into a single packet.
+};
+
+struct RTPVideoHeaderH264 {
+ uint8_t nalu_type; // The NAL unit type. If this is a header for a
+ // fragmented packet, it's the NAL unit type of
+ // the original data. If this is the header for an
+ // aggregated packet, it's the NAL unit type of
+ // the first NAL unit in the packet.
+ H264PacketizationTypes packetization_type;
+};
+
+union RTPVideoTypeHeader {
+ RTPVideoHeaderVP8 VP8;
+ RTPVideoHeaderVP9 VP9;
+ RTPVideoHeaderH264 H264;
+};
+
+enum RtpVideoCodecTypes {
+ kRtpVideoNone,
+ kRtpVideoGeneric,
+ kRtpVideoVp8,
+ kRtpVideoVp9,
+ kRtpVideoH264
+};
+// Since RTPVideoHeader is used as a member of a union, it can't have a
+// non-trivial default constructor.
+struct RTPVideoHeader {
+ uint16_t width; // size
+ uint16_t height;
+ VideoRotation rotation;
+
+ bool isFirstPacket; // first packet in frame
+ uint8_t simulcastIdx; // Index if the simulcast encoder creating
+ // this frame, 0 if not using simulcast.
+ RtpVideoCodecTypes codec;
+ RTPVideoTypeHeader codecHeader;
+};
+union RTPTypeHeader {
+ RTPAudioHeader Audio;
+ RTPVideoHeader Video;
+};
+
+struct WebRtcRTPHeader {
+ RTPHeader header;
+ FrameType frameType;
+ RTPTypeHeader type;
+ // NTP time of the capture time in local timebase in milliseconds.
+ int64_t ntp_time_ms;
+};
+
+class RTPFragmentationHeader {
+ public:
+ RTPFragmentationHeader()
+ : fragmentationVectorSize(0),
+ fragmentationOffset(NULL),
+ fragmentationLength(NULL),
+ fragmentationTimeDiff(NULL),
+ fragmentationPlType(NULL) {};
+
+ ~RTPFragmentationHeader() {
+ delete[] fragmentationOffset;
+ delete[] fragmentationLength;
+ delete[] fragmentationTimeDiff;
+ delete[] fragmentationPlType;
+ }
+
+ void CopyFrom(const RTPFragmentationHeader& src) {
+ if (this == &src) {
+ return;
+ }
+
+ if (src.fragmentationVectorSize != fragmentationVectorSize) {
+ // new size of vectors
+
+ // delete old
+ delete[] fragmentationOffset;
+ fragmentationOffset = NULL;
+ delete[] fragmentationLength;
+ fragmentationLength = NULL;
+ delete[] fragmentationTimeDiff;
+ fragmentationTimeDiff = NULL;
+ delete[] fragmentationPlType;
+ fragmentationPlType = NULL;
+
+ if (src.fragmentationVectorSize > 0) {
+ // allocate new
+ if (src.fragmentationOffset) {
+ fragmentationOffset = new size_t[src.fragmentationVectorSize];
+ }
+ if (src.fragmentationLength) {
+ fragmentationLength = new size_t[src.fragmentationVectorSize];
+ }
+ if (src.fragmentationTimeDiff) {
+ fragmentationTimeDiff = new uint16_t[src.fragmentationVectorSize];
+ }
+ if (src.fragmentationPlType) {
+ fragmentationPlType = new uint8_t[src.fragmentationVectorSize];
+ }
+ }
+ // set new size
+ fragmentationVectorSize = src.fragmentationVectorSize;
+ }
+
+ if (src.fragmentationVectorSize > 0) {
+ // copy values
+ if (src.fragmentationOffset) {
+ memcpy(fragmentationOffset, src.fragmentationOffset,
+ src.fragmentationVectorSize * sizeof(size_t));
+ }
+ if (src.fragmentationLength) {
+ memcpy(fragmentationLength, src.fragmentationLength,
+ src.fragmentationVectorSize * sizeof(size_t));
+ }
+ if (src.fragmentationTimeDiff) {
+ memcpy(fragmentationTimeDiff, src.fragmentationTimeDiff,
+ src.fragmentationVectorSize * sizeof(uint16_t));
+ }
+ if (src.fragmentationPlType) {
+ memcpy(fragmentationPlType, src.fragmentationPlType,
+ src.fragmentationVectorSize * sizeof(uint8_t));
+ }
+ }
+ }
+
+ void VerifyAndAllocateFragmentationHeader(const size_t size) {
+ assert(size <= std::numeric_limits<uint16_t>::max());
+ const uint16_t size16 = static_cast<uint16_t>(size);
+ if (fragmentationVectorSize < size16) {
+ uint16_t oldVectorSize = fragmentationVectorSize;
+ {
+ // offset
+ size_t* oldOffsets = fragmentationOffset;
+ fragmentationOffset = new size_t[size16];
+ memset(fragmentationOffset + oldVectorSize, 0,
+ sizeof(size_t) * (size16 - oldVectorSize));
+ // copy old values
+ memcpy(fragmentationOffset, oldOffsets,
+ sizeof(size_t) * oldVectorSize);
+ delete[] oldOffsets;
+ }
+ // length
+ {
+ size_t* oldLengths = fragmentationLength;
+ fragmentationLength = new size_t[size16];
+ memset(fragmentationLength + oldVectorSize, 0,
+ sizeof(size_t) * (size16 - oldVectorSize));
+ memcpy(fragmentationLength, oldLengths,
+ sizeof(size_t) * oldVectorSize);
+ delete[] oldLengths;
+ }
+ // time diff
+ {
+ uint16_t* oldTimeDiffs = fragmentationTimeDiff;
+ fragmentationTimeDiff = new uint16_t[size16];
+ memset(fragmentationTimeDiff + oldVectorSize, 0,
+ sizeof(uint16_t) * (size16 - oldVectorSize));
+ memcpy(fragmentationTimeDiff, oldTimeDiffs,
+ sizeof(uint16_t) * oldVectorSize);
+ delete[] oldTimeDiffs;
+ }
+ // payload type
+ {
+ uint8_t* oldTimePlTypes = fragmentationPlType;
+ fragmentationPlType = new uint8_t[size16];
+ memset(fragmentationPlType + oldVectorSize, 0,
+ sizeof(uint8_t) * (size16 - oldVectorSize));
+ memcpy(fragmentationPlType, oldTimePlTypes,
+ sizeof(uint8_t) * oldVectorSize);
+ delete[] oldTimePlTypes;
+ }
+ fragmentationVectorSize = size16;
+ }
+ }
+
+ uint16_t fragmentationVectorSize; // Number of fragmentations
+ size_t* fragmentationOffset; // Offset of pointer to data for each
+ // fragmentation
+ size_t* fragmentationLength; // Data size for each fragmentation
+ uint16_t* fragmentationTimeDiff; // Timestamp difference relative "now" for
+ // each fragmentation
+ uint8_t* fragmentationPlType; // Payload type of each fragmentation
+
+ private:
+ RTC_DISALLOW_COPY_AND_ASSIGN(RTPFragmentationHeader);
+};
+
+struct RTCPVoIPMetric {
+ // RFC 3611 4.7
+ uint8_t lossRate;
+ uint8_t discardRate;
+ uint8_t burstDensity;
+ uint8_t gapDensity;
+ uint16_t burstDuration;
+ uint16_t gapDuration;
+ uint16_t roundTripDelay;
+ uint16_t endSystemDelay;
+ uint8_t signalLevel;
+ uint8_t noiseLevel;
+ uint8_t RERL;
+ uint8_t Gmin;
+ uint8_t Rfactor;
+ uint8_t extRfactor;
+ uint8_t MOSLQ;
+ uint8_t MOSCQ;
+ uint8_t RXconfig;
+ uint16_t JBnominal;
+ uint16_t JBmax;
+ uint16_t JBabsMax;
+};
+
+// Types for the FEC packet masks. The type |kFecMaskRandom| is based on a
+// random loss model. The type |kFecMaskBursty| is based on a bursty/consecutive
+// loss model. The packet masks are defined in
+// modules/rtp_rtcp/fec_private_tables_random(bursty).h
+enum FecMaskType {
+ kFecMaskRandom,
+ kFecMaskBursty,
+};
+
+// Struct containing forward error correction settings.
+struct FecProtectionParams {
+ int fec_rate;
+ bool use_uep_protection;
+ int max_fec_frames;
+ FecMaskType fec_mask_type;
+};
+
+// Interface used by the CallStats class to distribute call statistics.
+// Callbacks will be triggered as soon as the class has been registered to a
+// CallStats object using RegisterStatsObserver.
+class CallStatsObserver {
+ public:
+ virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) = 0;
+
+ virtual ~CallStatsObserver() {}
+};
+
+struct VideoContentMetrics {
+ VideoContentMetrics()
+ : motion_magnitude(0.0f),
+ spatial_pred_err(0.0f),
+ spatial_pred_err_h(0.0f),
+ spatial_pred_err_v(0.0f) {}
+
+ void Reset() {
+ motion_magnitude = 0.0f;
+ spatial_pred_err = 0.0f;
+ spatial_pred_err_h = 0.0f;
+ spatial_pred_err_v = 0.0f;
+ }
+ float motion_magnitude;
+ float spatial_pred_err;
+ float spatial_pred_err_h;
+ float spatial_pred_err_v;
+};
+
+/* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
+ * allows for adding and subtracting frames while keeping track of the resulting
+ * states.
+ *
+ * Notes
+ * - The total number of samples in |data_| is
+ * samples_per_channel_ * num_channels_
+ *
+ * - Stereo data is interleaved starting with the left channel.
+ *
+ * - The +operator assume that you would never add exactly opposite frames when
+ * deciding the resulting state. To do this use the -operator.
+ */
+class AudioFrame {
+ public:
+ // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
+ static const size_t kMaxDataSizeSamples = 3840;
+
+ enum VADActivity {
+ kVadActive = 0,
+ kVadPassive = 1,
+ kVadUnknown = 2
+ };
+ enum SpeechType {
+ kNormalSpeech = 0,
+ kPLC = 1,
+ kCNG = 2,
+ kPLCCNG = 3,
+ kUndefined = 4
+ };
+
+ AudioFrame();
+ virtual ~AudioFrame() {}
+
+ // Resets all members to their default state (except does not modify the
+ // contents of |data_|).
+ void Reset();
+
+ // |interleaved_| is not changed by this method.
+ void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
+ size_t samples_per_channel, int sample_rate_hz,
+ SpeechType speech_type, VADActivity vad_activity,
+ int num_channels = 1, uint32_t energy = -1);
+
+ AudioFrame& Append(const AudioFrame& rhs);
+
+ void CopyFrom(const AudioFrame& src);
+
+ void Mute();
+
+ AudioFrame& operator>>=(const int rhs);
+ AudioFrame& operator+=(const AudioFrame& rhs);
+ AudioFrame& operator-=(const AudioFrame& rhs);
+
+ int id_;
+ // RTP timestamp of the first sample in the AudioFrame.
+ uint32_t timestamp_;
+ // Time since the first frame in milliseconds.
+ // -1 represents an uninitialized value.
+ int64_t elapsed_time_ms_;
+ // NTP time of the estimated capture time in local timebase in milliseconds.
+ // -1 represents an uninitialized value.
+ int64_t ntp_time_ms_;
+ int16_t data_[kMaxDataSizeSamples];
+ size_t samples_per_channel_;
+ int sample_rate_hz_;
+ int num_channels_;
+ SpeechType speech_type_;
+ VADActivity vad_activity_;
+ // Note that there is no guarantee that |energy_| is correct. Any user of this
+ // member must verify that the value is correct.
+ // TODO(henrike) Remove |energy_|.
+ // See https://code.google.com/p/webrtc/issues/detail?id=3315.
+ uint32_t energy_;
+ bool interleaved_;
+
+ private:
+ RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
+};
+
+inline AudioFrame::AudioFrame()
+ : data_() {
+ Reset();
+}
+
+inline void AudioFrame::Reset() {
+ id_ = -1;
+ // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
+ // to an invalid value, or add a new member to indicate invalidity.
+ timestamp_ = 0;
+ elapsed_time_ms_ = -1;
+ ntp_time_ms_ = -1;
+ samples_per_channel_ = 0;
+ sample_rate_hz_ = 0;
+ num_channels_ = 0;
+ speech_type_ = kUndefined;
+ vad_activity_ = kVadUnknown;
+ energy_ = 0xffffffff;
+ interleaved_ = true;
+}
+
+inline void AudioFrame::UpdateFrame(int id,
+ uint32_t timestamp,
+ const int16_t* data,
+ size_t samples_per_channel,
+ int sample_rate_hz,
+ SpeechType speech_type,
+ VADActivity vad_activity,
+ int num_channels,
+ uint32_t energy) {
+ id_ = id;
+ timestamp_ = timestamp;
+ samples_per_channel_ = samples_per_channel;
+ sample_rate_hz_ = sample_rate_hz;
+ speech_type_ = speech_type;
+ vad_activity_ = vad_activity;
+ num_channels_ = num_channels;
+ energy_ = energy;
+
+ assert(num_channels >= 0);
+ const size_t length = samples_per_channel * num_channels;
+ assert(length <= kMaxDataSizeSamples);
+ if (data != NULL) {
+ memcpy(data_, data, sizeof(int16_t) * length);
+ } else {
+ memset(data_, 0, sizeof(int16_t) * length);
+ }
+}
+
+inline void AudioFrame::CopyFrom(const AudioFrame& src) {
+ if (this == &src) return;
+
+ id_ = src.id_;
+ timestamp_ = src.timestamp_;
+ elapsed_time_ms_ = src.elapsed_time_ms_;
+ ntp_time_ms_ = src.ntp_time_ms_;
+ samples_per_channel_ = src.samples_per_channel_;
+ sample_rate_hz_ = src.sample_rate_hz_;
+ speech_type_ = src.speech_type_;
+ vad_activity_ = src.vad_activity_;
+ num_channels_ = src.num_channels_;
+ energy_ = src.energy_;
+ interleaved_ = src.interleaved_;
+
+ assert(num_channels_ >= 0);
+ const size_t length = samples_per_channel_ * num_channels_;
+ assert(length <= kMaxDataSizeSamples);
+ memcpy(data_, src.data_, sizeof(int16_t) * length);
+}
+
+inline void AudioFrame::Mute() {
+ memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
+}
+
+inline AudioFrame& AudioFrame::operator>>=(const int rhs) {
+ assert((num_channels_ > 0) && (num_channels_ < 3));
+ if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
+
+ for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
+ data_[i] = static_cast<int16_t>(data_[i] >> rhs);
+ }
+ return *this;
+}
+
+inline AudioFrame& AudioFrame::Append(const AudioFrame& rhs) {
+ // Sanity check
+ assert((num_channels_ > 0) && (num_channels_ < 3));
+ assert(interleaved_ == rhs.interleaved_);
+ if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
+ if (num_channels_ != rhs.num_channels_) return *this;
+
+ if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
+ vad_activity_ = kVadActive;
+ } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
+ vad_activity_ = kVadUnknown;
+ }
+ if (speech_type_ != rhs.speech_type_) {
+ speech_type_ = kUndefined;
+ }
+
+ size_t offset = samples_per_channel_ * num_channels_;
+ for (size_t i = 0; i < rhs.samples_per_channel_ * rhs.num_channels_; i++) {
+ data_[offset + i] = rhs.data_[i];
+ }
+ samples_per_channel_ += rhs.samples_per_channel_;
+ return *this;
+}
+
+namespace {
+inline int16_t ClampToInt16(int32_t input) {
+ if (input < -0x00008000) {
+ return -0x8000;
+ } else if (input > 0x00007FFF) {
+ return 0x7FFF;
+ } else {
+ return static_cast<int16_t>(input);
+ }
+}
+}
+
+inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
+ // Sanity check
+ assert((num_channels_ > 0) && (num_channels_ < 3));
+ assert(interleaved_ == rhs.interleaved_);
+ if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
+ if (num_channels_ != rhs.num_channels_) return *this;
+
+ bool noPrevData = false;
+ if (samples_per_channel_ != rhs.samples_per_channel_) {
+ if (samples_per_channel_ == 0) {
+ // special case we have no data to start with
+ samples_per_channel_ = rhs.samples_per_channel_;
+ noPrevData = true;
+ } else {
+ return *this;
+ }
+ }
+
+ if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
+ vad_activity_ = kVadActive;
+ } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
+ vad_activity_ = kVadUnknown;
+ }
+
+ if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined;
+
+ if (noPrevData) {
+ memcpy(data_, rhs.data_,
+ sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
+ } else {
+ // IMPROVEMENT this can be done very fast in assembly
+ for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
+ int32_t wrap_guard =
+ static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
+ data_[i] = ClampToInt16(wrap_guard);
+ }
+ }
+ energy_ = 0xffffffff;
+ return *this;
+}
+
+inline AudioFrame& AudioFrame::operator-=(const AudioFrame& rhs) {
+ // Sanity check
+ assert((num_channels_ > 0) && (num_channels_ < 3));
+ assert(interleaved_ == rhs.interleaved_);
+ if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
+
+ if ((samples_per_channel_ != rhs.samples_per_channel_) ||
+ (num_channels_ != rhs.num_channels_)) {
+ return *this;
+ }
+ if ((vad_activity_ != kVadPassive) || rhs.vad_activity_ != kVadPassive) {
+ vad_activity_ = kVadUnknown;
+ }
+ speech_type_ = kUndefined;
+
+ for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
+ int32_t wrap_guard =
+ static_cast<int32_t>(data_[i]) - static_cast<int32_t>(rhs.data_[i]);
+ data_[i] = ClampToInt16(wrap_guard);
+ }
+ energy_ = 0xffffffff;
+ return *this;
+}
+
+inline bool IsNewerSequenceNumber(uint16_t sequence_number,
+ uint16_t prev_sequence_number) {
+ // Distinguish between elements that are exactly 0x8000 apart.
+ // If s1>s2 and |s1-s2| = 0x8000: IsNewer(s1,s2)=true, IsNewer(s2,s1)=false
+ // rather than having IsNewer(s1,s2) = IsNewer(s2,s1) = false.
+ if (static_cast<uint16_t>(sequence_number - prev_sequence_number) == 0x8000) {
+ return sequence_number > prev_sequence_number;
+ }
+ return sequence_number != prev_sequence_number &&
+ static_cast<uint16_t>(sequence_number - prev_sequence_number) < 0x8000;
+}
+
+inline bool IsNewerTimestamp(uint32_t timestamp, uint32_t prev_timestamp) {
+ // Distinguish between elements that are exactly 0x80000000 apart.
+ // If t1>t2 and |t1-t2| = 0x80000000: IsNewer(t1,t2)=true,
+ // IsNewer(t2,t1)=false
+ // rather than having IsNewer(t1,t2) = IsNewer(t2,t1) = false.
+ if (static_cast<uint32_t>(timestamp - prev_timestamp) == 0x80000000) {
+ return timestamp > prev_timestamp;
+ }
+ return timestamp != prev_timestamp &&
+ static_cast<uint32_t>(timestamp - prev_timestamp) < 0x80000000;
+}
+
+inline uint16_t LatestSequenceNumber(uint16_t sequence_number1,
+ uint16_t sequence_number2) {
+ return IsNewerSequenceNumber(sequence_number1, sequence_number2)
+ ? sequence_number1
+ : sequence_number2;
+}
+
+inline uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) {
+ return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2;
+}
+
+// Utility class to unwrap a sequence number to a larger type, for easier
+// handling large ranges. Note that sequence numbers will never be unwrapped
+// to a negative value.
+class SequenceNumberUnwrapper {
+ public:
+ SequenceNumberUnwrapper() : last_seq_(-1) {}
+
+ // Get the unwrapped sequence, but don't update the internal state.
+ int64_t UnwrapWithoutUpdate(uint16_t sequence_number) {
+ if (last_seq_ == -1)
+ return sequence_number;
+
+ uint16_t cropped_last = static_cast<uint16_t>(last_seq_);
+ int64_t delta = sequence_number - cropped_last;
+ if (IsNewerSequenceNumber(sequence_number, cropped_last)) {
+ if (delta < 0)
+ delta += (1 << 16); // Wrap forwards.
+ } else if (delta > 0 && (last_seq_ + delta - (1 << 16)) >= 0) {
+ // If sequence_number is older but delta is positive, this is a backwards
+ // wrap-around. However, don't wrap backwards past 0 (unwrapped).
+ delta -= (1 << 16);
+ }
+
+ return last_seq_ + delta;
+ }
+
+ // Only update the internal state to the specified last (unwrapped) sequence.
+ void UpdateLast(int64_t last_sequence) { last_seq_ = last_sequence; }
+
+ // Unwrap the sequence number and update the internal state.
+ int64_t Unwrap(uint16_t sequence_number) {
+ int64_t unwrapped = UnwrapWithoutUpdate(sequence_number);
+ UpdateLast(unwrapped);
+ return unwrapped;
+ }
+
+ private:
+ int64_t last_seq_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
diff --git a/webrtc/modules/interface/module.h b/webrtc/modules/interface/module.h
index ffd3065..06054c3 100644
--- a/webrtc/modules/interface/module.h
+++ b/webrtc/modules/interface/module.h
@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef MODULES_INTERFACE_MODULE_H_
-#define MODULES_INTERFACE_MODULE_H_
+#ifndef WEBRTC_MODULES_INCLUDE_MODULE_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_H_
+
+#pragma message("WARNING: webrtc/modules/include is DEPRECATED; use webrtc/modules/include")
#include "webrtc/typedefs.h"
@@ -78,4 +80,4 @@
} // namespace webrtc
-#endif // MODULES_INTERFACE_MODULE_H_
+#endif // WEBRTC_MODULES_INCLUDE_MODULE_H_
diff --git a/webrtc/modules/interface/module_common_types.h b/webrtc/modules/interface/module_common_types.h
index 45e93d8..e76a714 100644
--- a/webrtc/modules/interface/module_common_types.h
+++ b/webrtc/modules/interface/module_common_types.h
@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef MODULE_COMMON_TYPES_H
-#define MODULE_COMMON_TYPES_H
+#ifndef WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
+
+#pragma message("WARNING: webrtc/modules/include is DEPRECATED; use webrtc/modules/include")
#include <assert.h>
#include <string.h> // memcpy
@@ -807,4 +809,4 @@
} // namespace webrtc
-#endif // MODULE_COMMON_TYPES_H
+#endif // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
diff --git a/webrtc/modules/media_file/BUILD.gn b/webrtc/modules/media_file/BUILD.gn
index 05cfb4e..70abb51 100644
--- a/webrtc/modules/media_file/BUILD.gn
+++ b/webrtc/modules/media_file/BUILD.gn
@@ -14,8 +14,8 @@
source_set("media_file") {
sources = [
- "interface/media_file.h",
- "interface/media_file_defines.h",
+ "include/media_file.h",
+ "include/media_file_defines.h",
"source/media_file_impl.cc",
"source/media_file_impl.h",
"source/media_file_utility.cc",
diff --git a/webrtc/modules/media_file/include/media_file.h b/webrtc/modules/media_file/include/media_file.h
new file mode 100644
index 0000000..22c727e
--- /dev/null
+++ b/webrtc/modules/media_file/include/media_file.h
@@ -0,0 +1,180 @@
+/*
+ * 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_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
+#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+class MediaFile : public Module
+{
+public:
+ // Factory method. Constructor disabled. id is the identifier for the
+ // MediaFile instance.
+ static MediaFile* CreateMediaFile(const int32_t id);
+ static void DestroyMediaFile(MediaFile* module);
+
+ // Put 10-60ms of audio data from file into the audioBuffer depending on
+ // codec frame size. dataLengthInBytes is both an input and output
+ // parameter. As input parameter it indicates the size of audioBuffer.
+ // As output parameter it indicates the number of bytes written to
+ // audioBuffer.
+ // Note: This API only play mono audio but can be used on file containing
+ // audio with more channels (in which case the audio will be converted to
+ // mono).
+ virtual int32_t PlayoutAudioData(
+ int8_t* audioBuffer,
+ size_t& dataLengthInBytes) = 0;
+
+ // Put 10-60ms, depending on codec frame size, of audio data from file into
+ // audioBufferLeft and audioBufferRight. The buffers contain the left and
+ // right channel of played out stereo audio.
+ // dataLengthInBytes is both an input and output parameter. As input
+ // parameter it indicates the size of both audioBufferLeft and
+ // audioBufferRight. As output parameter it indicates the number of bytes
+ // written to both audio buffers.
+ // Note: This API can only be successfully called for WAV files with stereo
+ // audio.
+ virtual int32_t PlayoutStereoData(
+ int8_t* audioBufferLeft,
+ int8_t* audioBufferRight,
+ size_t& dataLengthInBytes) = 0;
+
+ // Open the file specified by fileName (relative path is allowed) for
+ // reading. FileCallback::PlayNotification(..) will be called after
+ // notificationTimeMs of the file has been played if notificationTimeMs is
+ // greater than zero. If loop is true the file will be played until
+ // StopPlaying() is called. When end of file is reached the file is read
+ // from the start. format specifies the type of file fileName refers to.
+ // codecInst specifies the encoding of the audio data. Note that
+ // file formats that contain this information (like WAV files) don't need to
+ // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
+ // specify what part of the file should be read. From startPointMs ms to
+ // stopPointMs ms.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo audio is only supported for WAV files.
+ virtual int32_t StartPlayingAudioFile(
+ const char* fileName,
+ const uint32_t notificationTimeMs = 0,
+ const bool loop = false,
+ const FileFormats format = kFileFormatPcm16kHzFile,
+ const CodecInst* codecInst = NULL,
+ const uint32_t startPointMs = 0,
+ const uint32_t stopPointMs = 0) = 0;
+
+ // Prepare for playing audio from stream.
+ // FileCallback::PlayNotification(..) will be called after
+ // notificationTimeMs of the file has been played if notificationTimeMs is
+ // greater than zero. format specifies the type of file fileName refers to.
+ // codecInst specifies the encoding of the audio data. Note that
+ // file formats that contain this information (like WAV files) don't need to
+ // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
+ // specify what part of the file should be read. From startPointMs ms to
+ // stopPointMs ms.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo audio is only supported for WAV files.
+ virtual int32_t StartPlayingAudioStream(
+ InStream& stream,
+ const uint32_t notificationTimeMs = 0,
+ const FileFormats format = kFileFormatPcm16kHzFile,
+ const CodecInst* codecInst = NULL,
+ const uint32_t startPointMs = 0,
+ const uint32_t stopPointMs = 0) = 0;
+
+ // Stop playing from file or stream.
+ virtual int32_t StopPlaying() = 0;
+
+ // Return true if playing.
+ virtual bool IsPlaying() = 0;
+
+
+ // Set durationMs to the number of ms that has been played from file.
+ virtual int32_t PlayoutPositionMs(
+ uint32_t& durationMs) const = 0;
+
+ // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
+ // to file. The audio frame size is determined by the codecInst.pacsize
+ // parameter of the last sucessfull StartRecordingAudioFile(..) call.
+ // Note: bufferLength must be exactly one frame.
+ virtual int32_t IncomingAudioData(
+ const int8_t* audioBuffer,
+ const size_t bufferLength) = 0;
+
+ // Open/creates file specified by fileName for writing (relative path is
+ // allowed). FileCallback::RecordNotification(..) will be called after
+ // notificationTimeMs of audio data has been recorded if
+ // notificationTimeMs is greater than zero.
+ // format specifies the type of file that should be created/opened.
+ // codecInst specifies the encoding of the audio data. maxSizeBytes
+ // specifies the number of bytes allowed to be written to file if it is
+ // greater than zero.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo is only supported for WAV files.
+ virtual int32_t StartRecordingAudioFile(
+ const char* fileName,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const uint32_t notificationTimeMs = 0,
+ const uint32_t maxSizeBytes = 0) = 0;
+
+ // Prepare for recording audio to stream.
+ // FileCallback::RecordNotification(..) will be called after
+ // notificationTimeMs of audio data has been recorded if
+ // notificationTimeMs is greater than zero.
+ // format specifies the type of file that stream should correspond to.
+ // codecInst specifies the encoding of the audio data.
+ // Note: codecInst.channels should be set to 2 for stereo (and 1 for
+ // mono). Stereo is only supported for WAV files.
+ virtual int32_t StartRecordingAudioStream(
+ OutStream& stream,
+ const FileFormats format,
+ const CodecInst& codecInst,
+ const uint32_t notificationTimeMs = 0) = 0;
+
+ // Stop recording to file or stream.
+ virtual int32_t StopRecording() = 0;
+
+ // Return true if recording.
+ virtual bool IsRecording() = 0;
+
+ // Set durationMs to the number of ms that has been recorded to file.
+ virtual int32_t RecordDurationMs(uint32_t& durationMs) = 0;
+
+ // Return true if recording or playing is stereo.
+ virtual bool IsStereo() = 0;
+
+ // Register callback to receive media file related notifications. Disables
+ // callbacks if callback is NULL.
+ virtual int32_t SetModuleFileCallback(FileCallback* callback) = 0;
+
+ // Set durationMs to the size of the file (in ms) specified by fileName.
+ // format specifies the type of file fileName refers to. freqInHz specifies
+ // the sampling frequency of the file.
+ virtual int32_t FileDurationMs(
+ const char* fileName,
+ uint32_t& durationMs,
+ const FileFormats format,
+ const uint32_t freqInHz = 16000) = 0;
+
+ // Update codecInst according to the current audio codec being used for
+ // reading or writing.
+ virtual int32_t codec_info(CodecInst& codecInst) const = 0;
+
+protected:
+ MediaFile() {}
+ virtual ~MediaFile() {}
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
diff --git a/webrtc/modules/media_file/include/media_file_defines.h b/webrtc/modules/media_file/include/media_file_defines.h
new file mode 100644
index 0000000..05e7e72
--- /dev/null
+++ b/webrtc/modules/media_file/include/media_file_defines.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (c) 2011 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_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
+#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
+
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+// Callback class for the MediaFile class.
+class FileCallback
+{
+public:
+ virtual ~FileCallback(){}
+
+ // This function is called by MediaFile when a file has been playing for
+ // durationMs ms. id is the identifier for the MediaFile instance calling
+ // the callback.
+ virtual void PlayNotification(const int32_t id,
+ const uint32_t durationMs) = 0;
+
+ // This function is called by MediaFile when a file has been recording for
+ // durationMs ms. id is the identifier for the MediaFile instance calling
+ // the callback.
+ virtual void RecordNotification(const int32_t id,
+ const uint32_t durationMs) = 0;
+
+ // This function is called by MediaFile when a file has been stopped
+ // playing. id is the identifier for the MediaFile instance calling the
+ // callback.
+ virtual void PlayFileEnded(const int32_t id) = 0;
+
+ // This function is called by MediaFile when a file has been stopped
+ // recording. id is the identifier for the MediaFile instance calling the
+ // callback.
+ virtual void RecordFileEnded(const int32_t id) = 0;
+
+protected:
+ FileCallback() {}
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
diff --git a/webrtc/modules/media_file/interface/media_file.h b/webrtc/modules/media_file/interface/media_file.h
index 5b09ad4..3be5f08 100644
--- a/webrtc/modules/media_file/interface/media_file.h
+++ b/webrtc/modules/media_file/interface/media_file.h
@@ -8,13 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
+#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
+
+#pragma message("WARNING: media_file/interface is DEPRECATED; use media_file/include")
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -177,4 +179,4 @@
virtual ~MediaFile() {}
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
diff --git a/webrtc/modules/media_file/interface/media_file_defines.h b/webrtc/modules/media_file/interface/media_file_defines.h
index ded71a8..345fe54 100644
--- a/webrtc/modules/media_file/interface/media_file_defines.h
+++ b/webrtc/modules/media_file/interface/media_file_defines.h
@@ -8,11 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#ifndef WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
+#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
+
+#pragma message("WARNING: media_file/interface is DEPRECATED; use media_file/include")
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -48,4 +50,4 @@
FileCallback() {}
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
+#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
diff --git a/webrtc/modules/media_file/media_file.gypi b/webrtc/modules/media_file/media_file.gypi
index 4ec80c3..b3d8968 100644
--- a/webrtc/modules/media_file/media_file.gypi
+++ b/webrtc/modules/media_file/media_file.gypi
@@ -17,8 +17,8 @@
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
],
'sources': [
- 'interface/media_file.h',
- 'interface/media_file_defines.h',
+ 'include/media_file.h',
+ 'include/media_file_defines.h',
'source/media_file_impl.cc',
'source/media_file_impl.h',
'source/media_file_utility.cc',
diff --git a/webrtc/modules/media_file/source/media_file_impl.h b/webrtc/modules/media_file/source/media_file_impl.h
index cdb54d8..6184979 100644
--- a/webrtc/modules/media_file/source/media_file_impl.h
+++ b/webrtc/modules/media_file/source/media_file_impl.h
@@ -12,9 +12,9 @@
#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
#include "webrtc/modules/media_file/source/media_file_utility.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/media_file/source/media_file_unittest.cc b/webrtc/modules/media_file/source/media_file_unittest.cc
index 370d132..4a50c2e 100644
--- a/webrtc/modules/media_file/source/media_file_unittest.cc
+++ b/webrtc/modules/media_file/source/media_file_unittest.cc
@@ -9,7 +9,7 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
+#include "webrtc/modules/media_file/include/media_file.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
diff --git a/webrtc/modules/media_file/source/media_file_utility.cc b/webrtc/modules/media_file/source/media_file_utility.cc
index 61ae442..fad7fe4 100644
--- a/webrtc/modules/media_file/source/media_file_utility.cc
+++ b/webrtc/modules/media_file/source/media_file_utility.cc
@@ -19,7 +19,7 @@
#include "webrtc/common_audio/wav_header.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/modules/media_file/source/media_file_utility.h b/webrtc/modules/media_file/source/media_file_utility.h
index 2823cec..65921bd 100644
--- a/webrtc/modules/media_file/source/media_file_utility.h
+++ b/webrtc/modules/media_file/source/media_file_utility.h
@@ -15,7 +15,7 @@
#include <stdio.h>
#include "webrtc/common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
namespace webrtc {
class InStream;
diff --git a/webrtc/modules/module_common_types_unittest.cc b/webrtc/modules/module_common_types_unittest.cc
index bc0b7a1..acd5847 100644
--- a/webrtc/modules/module_common_types_unittest.cc
+++ b/webrtc/modules/module_common_types_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "testing/gtest/include/gtest/gtest.h"
diff --git a/webrtc/modules/pacing/include/paced_sender.h b/webrtc/modules/pacing/include/paced_sender.h
index f142f55..2c8e36e 100644
--- a/webrtc/modules/pacing/include/paced_sender.h
+++ b/webrtc/modules/pacing/include/paced_sender.h
@@ -16,8 +16,8 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/pacing/include/packet_router.h b/webrtc/modules/pacing/include/packet_router.h
index 9d461d1..7a65295 100644
--- a/webrtc/modules/pacing/include/packet_router.h
+++ b/webrtc/modules/pacing/include/packet_router.h
@@ -19,7 +19,7 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc
index 5d7ae17..4088715 100644
--- a/webrtc/modules/pacing/paced_sender.cc
+++ b/webrtc/modules/pacing/paced_sender.cc
@@ -16,7 +16,7 @@
#include <queue>
#include <set>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/pacing/bitrate_prober.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/pacing/packet_router.cc b/webrtc/modules/pacing/packet_router.cc
index 563773b..7af74ec 100644
--- a/webrtc/modules/pacing/packet_router.cc
+++ b/webrtc/modules/pacing/packet_router.cc
@@ -12,8 +12,8 @@
#include "webrtc/base/atomicops.h"
#include "webrtc/base/checks.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
namespace webrtc {
diff --git a/webrtc/modules/pacing/packet_router_unittest.cc b/webrtc/modules/pacing/packet_router_unittest.cc
index eecb137..f17f797 100644
--- a/webrtc/modules/pacing/packet_router_unittest.cc
+++ b/webrtc/modules/pacing/packet_router_unittest.cc
@@ -14,7 +14,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/base/scoped_ptr.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h b/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
index 4bd9d8c..0734cbf 100644
--- a/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
+++ b/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
@@ -17,9 +17,9 @@
#include <vector>
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc b/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
index 3dee305..961d6bf 100644
--- a/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
+++ b/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
@@ -14,7 +14,7 @@
#include <cassert>
#include "webrtc/base/logging.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/overuse_detector.h b/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
index bb69a8a..56e9c14 100644
--- a/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
+++ b/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
@@ -13,7 +13,7 @@
#include <list>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
index b7f9f65..5c45370 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
@@ -15,7 +15,7 @@
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/modules/pacing/include/packet_router.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
index e867ff7..98a68b3 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
+++ b/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
@@ -15,7 +15,7 @@
#include <vector>
#include "webrtc/base/criticalsection.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
index f837638..f2ff768 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
@@ -14,7 +14,7 @@
#include "webrtc/base/common.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
#include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
#include "webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
index 64dfa85..b3df712 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.h
@@ -12,7 +12,7 @@
#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BWE_TEST_BASELINEFILE_H_
#include <string>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
index 2881eba..d470324 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.h
@@ -16,7 +16,7 @@
#include <string>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
index 6b24cf3..6b5f04b 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
@@ -24,12 +24,12 @@
#include "webrtc/base/common.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/pacing/include/paced_sender.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "webrtc/modules/remote_bitrate_estimator/test/packet.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/test/random.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
index d77447f..171d196 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
@@ -21,7 +21,7 @@
#include "webrtc/base/common.h"
#include "webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
namespace webrtc {
namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
index eee90cf..bf23d09 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
@@ -20,7 +20,7 @@
#include <list>
#include <map>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
#include "webrtc/voice_engine/channel.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
index b18b9f0..9599b01 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
@@ -17,7 +17,7 @@
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
namespace webrtc {
namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
index a02abc6..b7e4f97 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
@@ -16,7 +16,7 @@
#include "webrtc/base/common.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
namespace webrtc {
namespace testing {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet.h b/webrtc/modules/remote_bitrate_estimator/test/packet.h
index 11885a4..4a361c4 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet.h
@@ -16,7 +16,7 @@
#include <vector>
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc b/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
index f70c212..793e064 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
@@ -14,10 +14,10 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/common.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
index f1faa49..3bcbc0a 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
@@ -15,7 +15,7 @@
#include <sstream>
#include "webrtc/base/checks.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
#include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
index c42647e..0b3741d 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
@@ -17,7 +17,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
index 9493805..083d1fd 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
@@ -17,8 +17,8 @@
#include "gflags/gflags.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/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/test/rtp_file_reader.h"
namespace flags {
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
index 19e4a07..4574faf 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
@@ -14,8 +14,8 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/test/rtp_file_reader.h"
class Observer : public webrtc::RemoteBitrateObserver {
diff --git a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
index e277481..bf69872 100644
--- a/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
+++ b/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
@@ -14,8 +14,8 @@
#include "webrtc/base/format_macros.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/test/rtp_file_reader.h"
int main(int argc, char** argv) {
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
index f2e073a..332590b 100644
--- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc
@@ -15,7 +15,7 @@
#include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
namespace webrtc {
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
index 58829b0..93f30e6 100644
--- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h
@@ -15,8 +15,8 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
diff --git a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
index b2bc646..64d0e55 100644
--- a/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter_unittest.cc
@@ -18,9 +18,9 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
#include "webrtc/system_wrappers/include/clock.h"
using ::testing::_;
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index c651424..6239c04 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -10,14 +10,14 @@
source_set("rtp_rtcp") {
sources = [
- "interface/fec_receiver.h",
- "interface/receive_statistics.h",
- "interface/remote_ntp_time_estimator.h",
- "interface/rtp_header_parser.h",
- "interface/rtp_payload_registry.h",
- "interface/rtp_receiver.h",
- "interface/rtp_rtcp.h",
- "interface/rtp_rtcp_defines.h",
+ "include/fec_receiver.h",
+ "include/receive_statistics.h",
+ "include/remote_ntp_time_estimator.h",
+ "include/rtp_header_parser.h",
+ "include/rtp_payload_registry.h",
+ "include/rtp_receiver.h",
+ "include/rtp_rtcp.h",
+ "include/rtp_rtcp_defines.h",
"mocks/mock_rtp_rtcp.h",
"source/bitrate.cc",
"source/bitrate.h",
diff --git a/webrtc/modules/rtp_rtcp/include/fec_receiver.h b/webrtc/modules/rtp_rtcp/include/fec_receiver.h
new file mode 100644
index 0000000..65e85ad
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/fec_receiver.h
@@ -0,0 +1,46 @@
+/*
+ * 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_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct FecPacketCounter {
+ FecPacketCounter()
+ : num_packets(0),
+ num_fec_packets(0),
+ num_recovered_packets(0) {}
+
+ size_t num_packets; // Number of received packets.
+ size_t num_fec_packets; // Number of received FEC packets.
+ size_t num_recovered_packets; // Number of recovered media packets using FEC.
+};
+
+class FecReceiver {
+ public:
+ static FecReceiver* Create(RtpData* callback);
+
+ virtual ~FecReceiver() {}
+
+ virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
+ const uint8_t* incoming_rtp_packet,
+ size_t packet_length,
+ uint8_t ulpfec_payload_type) = 0;
+
+ virtual int32_t ProcessReceivedFec() = 0;
+
+ virtual FecPacketCounter GetPacketCounter() const = 0;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/include/receive_statistics.h b/webrtc/modules/rtp_rtcp/include/receive_statistics.h
new file mode 100644
index 0000000..b4a7cd0
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/receive_statistics.h
@@ -0,0 +1,102 @@
+/*
+ * 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_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+
+#include <map>
+
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Clock;
+
+class StreamStatistician {
+ public:
+ virtual ~StreamStatistician();
+
+ virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 0;
+ virtual void GetDataCounters(size_t* bytes_received,
+ uint32_t* packets_received) const = 0;
+
+ // Gets received stream data counters (includes reset counter values).
+ virtual void GetReceiveStreamDataCounters(
+ StreamDataCounters* data_counters) const = 0;
+
+ virtual uint32_t BitrateReceived() const = 0;
+
+ // Returns true if the packet with RTP header |header| is likely to be a
+ // retransmitted packet, false otherwise.
+ virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
+ int64_t min_rtt) const = 0;
+
+ // Returns true if |sequence_number| is received in order, false otherwise.
+ virtual bool IsPacketInOrder(uint16_t sequence_number) const = 0;
+};
+
+typedef std::map<uint32_t, StreamStatistician*> StatisticianMap;
+
+class ReceiveStatistics : public Module {
+ public:
+ virtual ~ReceiveStatistics() {}
+
+ static ReceiveStatistics* Create(Clock* clock);
+
+ // Updates the receive statistics with this packet.
+ virtual void IncomingPacket(const RTPHeader& rtp_header,
+ size_t packet_length,
+ bool retransmitted) = 0;
+
+ // Increment counter for number of FEC packets received.
+ virtual void FecPacketReceived(const RTPHeader& header,
+ size_t packet_length) = 0;
+
+ // Returns a map of all statisticians which have seen an incoming packet
+ // during the last two seconds.
+ virtual StatisticianMap GetActiveStatisticians() const = 0;
+
+ // Returns a pointer to the statistician of an ssrc.
+ virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
+
+ // Sets the max reordering threshold in number of packets.
+ virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
+
+ // Called on new RTCP stats creation.
+ virtual void RegisterRtcpStatisticsCallback(
+ RtcpStatisticsCallback* callback) = 0;
+
+ // Called on new RTP stats creation.
+ virtual void RegisterRtpStatisticsCallback(
+ StreamDataCountersCallback* callback) = 0;
+};
+
+class NullReceiveStatistics : public ReceiveStatistics {
+ public:
+ void IncomingPacket(const RTPHeader& rtp_header,
+ size_t packet_length,
+ bool retransmitted) override;
+ void FecPacketReceived(const RTPHeader& header,
+ size_t packet_length) override;
+ StatisticianMap GetActiveStatisticians() const override;
+ StreamStatistician* GetStatistician(uint32_t ssrc) const override;
+ int64_t TimeUntilNextProcess() override;
+ int32_t Process() override;
+ void SetMaxReorderingThreshold(int max_reordering_threshold) override;
+ void RegisterRtcpStatisticsCallback(
+ RtcpStatisticsCallback* callback) override;
+ void RegisterRtpStatisticsCallback(
+ StreamDataCountersCallback* callback) override;
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
diff --git a/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
new file mode 100644
index 0000000..56c6e48
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
@@ -0,0 +1,51 @@
+/*
+ * 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_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
+
+namespace webrtc {
+
+class Clock;
+class TimestampExtrapolator;
+
+// RemoteNtpTimeEstimator can be used to estimate a given RTP timestamp's NTP
+// time in local timebase.
+// Note that it needs to be trained with at least 2 RTCP SR (by calling
+// |UpdateRtcpTimestamp|) before it can be used.
+class RemoteNtpTimeEstimator {
+ public:
+ explicit RemoteNtpTimeEstimator(Clock* clock);
+
+ ~RemoteNtpTimeEstimator();
+
+ // Updates the estimator with round trip time |rtt|, NTP seconds |ntp_secs|,
+ // NTP fraction |ntp_frac| and RTP timestamp |rtcp_timestamp|.
+ bool UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac,
+ uint32_t rtp_timestamp);
+
+ // Estimates the NTP timestamp in local timebase from |rtp_timestamp|.
+ // Returns the NTP timestamp in ms when success. -1 if failed.
+ int64_t Estimate(uint32_t rtp_timestamp);
+
+ private:
+ Clock* clock_;
+ rtc::scoped_ptr<TimestampExtrapolator> ts_extrapolator_;
+ RtcpList rtcp_list_;
+ int64_t last_timing_log_ms_;
+ RTC_DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_cvo.h b/webrtc/modules/rtp_rtcp/include/rtp_cvo.h
new file mode 100644
index 0000000..2e30d89
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_cvo.h
@@ -0,0 +1,54 @@
+/*
+ * 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_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+
+#include "webrtc/common_video/rotation.h"
+
+namespace webrtc {
+
+// Please refer to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/
+// 12.07.00_60/ts_126114v120700p.pdf Section 7.4.5. The rotation of a frame is
+// the clockwise angle the frames must be rotated in order to display the frames
+// correctly if the display is rotated in its natural orientation.
+inline uint8_t ConvertVideoRotationToCVOByte(VideoRotation rotation) {
+ switch (rotation) {
+ case kVideoRotation_0:
+ return 0;
+ case kVideoRotation_90:
+ return 1;
+ case kVideoRotation_180:
+ return 2;
+ case kVideoRotation_270:
+ return 3;
+ }
+ assert(false);
+ return 0;
+}
+
+inline VideoRotation ConvertCVOByteToVideoRotation(uint8_t rotation) {
+ switch (rotation) {
+ case 0:
+ return kVideoRotation_0;
+ case 1:
+ return kVideoRotation_90;
+ case 2:
+ return kVideoRotation_180;
+ break;
+ case 3:
+ return kVideoRotation_270;
+ default:
+ assert(false);
+ return kVideoRotation_0;
+ }
+}
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h b/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
new file mode 100644
index 0000000..329de32
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
@@ -0,0 +1,44 @@
+/*
+ * 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_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+struct RTPHeader;
+
+class RtpHeaderParser {
+ public:
+ static RtpHeaderParser* Create();
+ virtual ~RtpHeaderParser() {}
+
+ // Returns true if the packet is an RTCP packet, false otherwise.
+ static bool IsRtcp(const uint8_t* packet, size_t length);
+
+ // Parses the packet and stores the parsed packet in |header|. Returns true on
+ // success, false otherwise.
+ // This method is thread-safe in the sense that it can parse multiple packets
+ // at once.
+ virtual bool Parse(const uint8_t* packet,
+ size_t length,
+ RTPHeader* header) const = 0;
+
+ // Registers an RTP header extension and binds it to |id|.
+ virtual bool RegisterRtpHeaderExtension(RTPExtensionType type,
+ uint8_t id) = 0;
+
+ // De-registers an RTP header extension.
+ virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
new file mode 100644
index 0000000..c2f467a
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
@@ -0,0 +1,193 @@
+/*
+ * 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_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+
+namespace webrtc {
+
+// This strategy deals with the audio/video-specific aspects
+// of payload handling.
+class RTPPayloadStrategy {
+ public:
+ virtual ~RTPPayloadStrategy() {}
+
+ virtual bool CodecsMustBeUnique() const = 0;
+
+ virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload,
+ const uint32_t frequency,
+ const uint8_t channels,
+ const uint32_t rate) const = 0;
+
+ virtual void UpdatePayloadRate(RtpUtility::Payload* payload,
+ const uint32_t rate) const = 0;
+
+ virtual RtpUtility::Payload* CreatePayloadType(
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const int8_t payloadType,
+ const uint32_t frequency,
+ const uint8_t channels,
+ const uint32_t rate) const = 0;
+
+ virtual int GetPayloadTypeFrequency(
+ const RtpUtility::Payload& payload) const = 0;
+
+ static RTPPayloadStrategy* CreateStrategy(const bool handling_audio);
+
+ protected:
+ RTPPayloadStrategy() {}
+};
+
+class RTPPayloadRegistry {
+ public:
+ // The registry takes ownership of the strategy.
+ RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
+ ~RTPPayloadRegistry();
+
+ int32_t RegisterReceivePayload(
+ const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+ const int8_t payload_type,
+ const uint32_t frequency,
+ const uint8_t channels,
+ const uint32_t rate,
+ bool* created_new_payload_type);
+
+ int32_t DeRegisterReceivePayload(
+ const int8_t payload_type);
+
+ int32_t ReceivePayloadType(
+ const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+ const uint32_t frequency,
+ const uint8_t channels,
+ const uint32_t rate,
+ int8_t* payload_type) const;
+
+ bool RtxEnabled() const;
+
+ void SetRtxSsrc(uint32_t ssrc);
+
+ bool GetRtxSsrc(uint32_t* ssrc) const;
+
+ void SetRtxPayloadType(int payload_type, int associated_payload_type);
+
+ bool IsRtx(const RTPHeader& header) const;
+
+ // DEPRECATED. Use RestoreOriginalPacket below that takes a uint8_t*
+ // restored_packet, instead of a uint8_t**.
+ // TODO(noahric): Remove this when all callers have been updated.
+ bool RestoreOriginalPacket(uint8_t** restored_packet,
+ const uint8_t* packet,
+ size_t* packet_length,
+ uint32_t original_ssrc,
+ const RTPHeader& header) const;
+
+ bool RestoreOriginalPacket(uint8_t* restored_packet,
+ const uint8_t* packet,
+ size_t* packet_length,
+ uint32_t original_ssrc,
+ const RTPHeader& header) const;
+
+ bool IsRed(const RTPHeader& header) const;
+
+ // Returns true if the media of this RTP packet is encapsulated within an
+ // extra header, such as RTX or RED.
+ bool IsEncapsulated(const RTPHeader& header) const;
+
+ bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
+
+ int GetPayloadTypeFrequency(uint8_t payload_type) const;
+
+ bool PayloadTypeToPayload(const uint8_t payload_type,
+ RtpUtility::Payload*& payload) const;
+
+ void ResetLastReceivedPayloadTypes() {
+ CriticalSectionScoped cs(crit_sect_.get());
+ last_received_payload_type_ = -1;
+ last_received_media_payload_type_ = -1;
+ }
+
+ // This sets the payload type of the packets being received from the network
+ // on the media SSRC. For instance if packets are encapsulated with RED, this
+ // payload type will be the RED payload type.
+ void SetIncomingPayloadType(const RTPHeader& header);
+
+ // Returns true if the new media payload type has not changed.
+ bool ReportMediaPayloadType(uint8_t media_payload_type);
+
+ int8_t red_payload_type() const {
+ CriticalSectionScoped cs(crit_sect_.get());
+ return red_payload_type_;
+ }
+ int8_t ulpfec_payload_type() const {
+ CriticalSectionScoped cs(crit_sect_.get());
+ return ulpfec_payload_type_;
+ }
+ int8_t last_received_payload_type() const {
+ CriticalSectionScoped cs(crit_sect_.get());
+ return last_received_payload_type_;
+ }
+ void set_last_received_payload_type(int8_t last_received_payload_type) {
+ CriticalSectionScoped cs(crit_sect_.get());
+ last_received_payload_type_ = last_received_payload_type;
+ }
+
+ int8_t last_received_media_payload_type() const {
+ CriticalSectionScoped cs(crit_sect_.get());
+ return last_received_media_payload_type_;
+ };
+
+ bool use_rtx_payload_mapping_on_restore() const {
+ CriticalSectionScoped cs(crit_sect_.get());
+ return use_rtx_payload_mapping_on_restore_;
+ }
+
+ void set_use_rtx_payload_mapping_on_restore(bool val) {
+ CriticalSectionScoped cs(crit_sect_.get());
+ use_rtx_payload_mapping_on_restore_ = val;
+ }
+
+ private:
+ // Prunes the payload type map of the specific payload type, if it exists.
+ void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
+ const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+ const size_t payload_name_length,
+ const uint32_t frequency,
+ const uint8_t channels,
+ const uint32_t rate);
+
+ bool IsRtxInternal(const RTPHeader& header) const;
+
+ rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ RtpUtility::PayloadTypeMap payload_type_map_;
+ rtc::scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_;
+ int8_t red_payload_type_;
+ int8_t ulpfec_payload_type_;
+ int8_t incoming_payload_type_;
+ int8_t last_received_payload_type_;
+ int8_t last_received_media_payload_type_;
+ bool rtx_;
+ // TODO(changbin): Remove rtx_payload_type_ once interop with old clients that
+ // only understand one RTX PT is no longer needed.
+ int rtx_payload_type_;
+ // Mapping rtx_payload_type_map_[rtx] = associated.
+ std::map<int, int> rtx_payload_type_map_;
+ // When true, use rtx_payload_type_map_ when restoring RTX packets to get the
+ // correct payload type.
+ bool use_rtx_payload_mapping_on_restore_;
+ uint32_t ssrc_rtx_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
new file mode 100644
index 0000000..d257a30
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
@@ -0,0 +1,103 @@
+/*
+ * 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_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class RTPPayloadRegistry;
+
+class TelephoneEventHandler {
+ public:
+ virtual ~TelephoneEventHandler() {}
+
+ // The following three methods implement the TelephoneEventHandler interface.
+ // Forward DTMFs to decoder for playout.
+ virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
+
+ // Is forwarding of outband telephone events turned on/off?
+ virtual bool TelephoneEventForwardToDecoder() const = 0;
+
+ // Is TelephoneEvent configured with payload type payload_type
+ virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
+};
+
+class RtpReceiver {
+ public:
+ // Creates a video-enabled RTP receiver.
+ static RtpReceiver* CreateVideoReceiver(
+ Clock* clock,
+ RtpData* incoming_payload_callback,
+ RtpFeedback* incoming_messages_callback,
+ RTPPayloadRegistry* rtp_payload_registry);
+
+ // Creates an audio-enabled RTP receiver.
+ static RtpReceiver* CreateAudioReceiver(
+ Clock* clock,
+ RtpAudioFeedback* incoming_audio_feedback,
+ RtpData* incoming_payload_callback,
+ RtpFeedback* incoming_messages_callback,
+ RTPPayloadRegistry* rtp_payload_registry);
+
+ virtual ~RtpReceiver() {}
+
+ // Returns a TelephoneEventHandler if available.
+ virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
+
+ // Registers a receive payload in the payload registry and notifies the media
+ // receiver strategy.
+ virtual int32_t RegisterReceivePayload(
+ const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+ const int8_t payload_type,
+ const uint32_t frequency,
+ const uint8_t channels,
+ const uint32_t rate) = 0;
+
+ // De-registers |payload_type| from the payload registry.
+ virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
+
+ // Parses the media specific parts of an RTP packet and updates the receiver
+ // state. This for instance means that any changes in SSRC and payload type is
+ // detected and acted upon.
+ virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
+ const uint8_t* payload,
+ size_t payload_length,
+ PayloadUnion payload_specific,
+ bool in_order) = 0;
+
+ // Returns the currently configured NACK method.
+ virtual NACKMethod NACK() const = 0;
+
+ // Turn negative acknowledgement (NACK) requests on/off.
+ virtual void SetNACKStatus(const NACKMethod method) = 0;
+
+ // Gets the last received timestamp. Returns true if a packet has been
+ // received, false otherwise.
+ virtual bool Timestamp(uint32_t* timestamp) const = 0;
+ // Gets the time in milliseconds when the last timestamp was received.
+ // Returns true if a packet has been received, false otherwise.
+ virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0;
+
+ // Returns the remote SSRC of the currently received RTP stream.
+ virtual uint32_t SSRC() const = 0;
+
+ // Returns the current remote CSRCs.
+ virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
+
+ // Returns the current energy of the RTP stream received.
+ virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
+};
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
new file mode 100644
index 0000000..39aeba2
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -0,0 +1,641 @@
+/*
+ * 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_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+
+#include <set>
+#include <vector>
+
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+// Forward declarations.
+class ReceiveStatistics;
+class RemoteBitrateEstimator;
+class RtpReceiver;
+class Transport;
+namespace rtcp {
+class TransportFeedback;
+}
+
+class RtpRtcp : public Module {
+ public:
+ struct Configuration {
+ Configuration();
+
+ /* id - Unique identifier of this RTP/RTCP module object
+ * audio - True for a audio version of the RTP/RTCP module
+ * object false will create a video version
+ * clock - The clock to use to read time. If NULL object
+ * will be using the system clock.
+ * incoming_data - Callback object that will receive the incoming
+ * data. May not be NULL; default callback will do
+ * nothing.
+ * incoming_messages - Callback object that will receive the incoming
+ * RTP messages. May not be NULL; default callback
+ * will do nothing.
+ * outgoing_transport - Transport object that will be called when packets
+ * are ready to be sent out on the network
+ * intra_frame_callback - Called when the receiver request a intra frame.
+ * bandwidth_callback - Called when we receive a changed estimate from
+ * the receiver of out stream.
+ * audio_messages - Telephone events. May not be NULL; default
+ * callback will do nothing.
+ * remote_bitrate_estimator - Estimates the bandwidth available for a set of
+ * streams from the same client.
+ * paced_sender - Spread any bursts of packets into smaller
+ * bursts to minimize packet loss.
+ */
+ bool audio;
+ bool receiver_only;
+ Clock* clock;
+ ReceiveStatistics* receive_statistics;
+ Transport* outgoing_transport;
+ RtcpIntraFrameObserver* intra_frame_callback;
+ RtcpBandwidthObserver* bandwidth_callback;
+ TransportFeedbackObserver* transport_feedback_callback;
+ RtcpRttStats* rtt_stats;
+ RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
+ RtpAudioFeedback* audio_messages;
+ RemoteBitrateEstimator* remote_bitrate_estimator;
+ RtpPacketSender* paced_sender;
+ TransportSequenceNumberAllocator* transport_sequence_number_allocator;
+ BitrateStatisticsObserver* send_bitrate_observer;
+ FrameCountObserver* send_frame_count_observer;
+ SendSideDelayObserver* send_side_delay_observer;
+ };
+
+ /*
+ * Create a RTP/RTCP module object using the system clock.
+ *
+ * configuration - Configuration of the RTP/RTCP module.
+ */
+ static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);
+
+ /**************************************************************************
+ *
+ * Receiver functions
+ *
+ ***************************************************************************/
+
+ virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
+ size_t incoming_packet_length) = 0;
+
+ virtual void SetRemoteSSRC(uint32_t ssrc) = 0;
+
+ /**************************************************************************
+ *
+ * Sender
+ *
+ ***************************************************************************/
+
+ /*
+ * set MTU
+ *
+ * size - Max transfer unit in bytes, default is 1500
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SetMaxTransferUnit(uint16_t size) = 0;
+
+ /*
+ * set transtport overhead
+ * default is IPv4 and UDP with no encryption
+ *
+ * TCP - true for TCP false UDP
+ * IPv6 - true for IP version 6 false for version 4
+ * authenticationOverhead - number of bytes to leave for an
+ * authentication header
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SetTransportOverhead(
+ bool TCP,
+ bool IPV6,
+ uint8_t authenticationOverhead = 0) = 0;
+
+ /*
+ * Get max payload length
+ *
+ * A combination of the configuration MaxTransferUnit and
+ * TransportOverhead.
+ * Does not account FEC/ULP/RED overhead if FEC is enabled.
+ * Does not account for RTP headers
+ */
+ virtual uint16_t MaxPayloadLength() const = 0;
+
+ /*
+ * Get max data payload length
+ *
+ * A combination of the configuration MaxTransferUnit, headers and
+ * TransportOverhead.
+ * Takes into account FEC/ULP/RED overhead if FEC is enabled.
+ * Takes into account RTP headers
+ */
+ virtual uint16_t MaxDataPayloadLength() const = 0;
+
+ /*
+ * set codec name and payload type
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t RegisterSendPayload(
+ const CodecInst& voiceCodec) = 0;
+
+ /*
+ * set codec name and payload type
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t RegisterSendPayload(
+ const VideoCodec& videoCodec) = 0;
+
+ /*
+ * Unregister a send payload
+ *
+ * payloadType - payload type of codec
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0;
+
+ /*
+ * (De)register RTP header extension type and id.
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
+ uint8_t id) = 0;
+
+ virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0;
+
+ /*
+ * get start timestamp
+ */
+ virtual uint32_t StartTimestamp() const = 0;
+
+ /*
+ * configure start timestamp, default is a random number
+ *
+ * timestamp - start timestamp
+ */
+ virtual void SetStartTimestamp(uint32_t timestamp) = 0;
+
+ /*
+ * Get SequenceNumber
+ */
+ virtual uint16_t SequenceNumber() const = 0;
+
+ /*
+ * Set SequenceNumber, default is a random number
+ */
+ virtual void SetSequenceNumber(uint16_t seq) = 0;
+
+ // Returns true if the ssrc matched this module, false otherwise.
+ virtual bool SetRtpStateForSsrc(uint32_t ssrc,
+ const RtpState& rtp_state) = 0;
+ virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
+
+ /*
+ * Get SSRC
+ */
+ virtual uint32_t SSRC() const = 0;
+
+ /*
+ * configure SSRC, default is a random number
+ */
+ virtual void SetSSRC(uint32_t ssrc) = 0;
+
+ /*
+ * Set CSRC
+ *
+ * csrcs - vector of CSRCs
+ */
+ virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0;
+
+ /*
+ * Turn on/off sending RTX (RFC 4588). The modes can be set as a combination
+ * of values of the enumerator RtxMode.
+ */
+ virtual void SetRtxSendStatus(int modes) = 0;
+
+ /*
+ * Get status of sending RTX (RFC 4588). The returned value can be
+ * a combination of values of the enumerator RtxMode.
+ */
+ virtual int RtxSendStatus() const = 0;
+
+ // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX,
+ // only the SSRC is set.
+ virtual void SetRtxSsrc(uint32_t ssrc) = 0;
+
+ // Sets the payload type to use when sending RTX packets. Note that this
+ // doesn't enable RTX, only the payload type is set.
+ virtual void SetRtxSendPayloadType(int payload_type,
+ int associated_payload_type) = 0;
+
+ // Gets the payload type pair of (RTX, associated) to use when sending RTX
+ // packets.
+ virtual std::pair<int, int> RtxSendPayloadType() const = 0;
+
+ /*
+ * sends kRtcpByeCode when going from true to false
+ *
+ * sending - on/off
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SetSendingStatus(bool sending) = 0;
+
+ /*
+ * get send status
+ */
+ virtual bool Sending() const = 0;
+
+ /*
+ * Starts/Stops media packets, on by default
+ *
+ * sending - on/off
+ */
+ virtual void SetSendingMediaStatus(bool sending) = 0;
+
+ /*
+ * get send status
+ */
+ virtual bool SendingMedia() const = 0;
+
+ /*
+ * get sent bitrate in Kbit/s
+ */
+ virtual void BitrateSent(uint32_t* totalRate,
+ uint32_t* videoRate,
+ uint32_t* fecRate,
+ uint32_t* nackRate) const = 0;
+
+ /*
+ * Used by the codec module to deliver a video or audio frame for
+ * packetization.
+ *
+ * frameType - type of frame to send
+ * payloadType - payload type of frame to send
+ * timestamp - timestamp of frame to send
+ * payloadData - payload buffer of frame to send
+ * payloadSize - size of payload buffer to send
+ * fragmentation - fragmentation offset data for fragmented frames such
+ * as layers or RED
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SendOutgoingData(
+ FrameType frameType,
+ int8_t payloadType,
+ uint32_t timeStamp,
+ int64_t capture_time_ms,
+ const uint8_t* payloadData,
+ size_t payloadSize,
+ const RTPFragmentationHeader* fragmentation = NULL,
+ const RTPVideoHeader* rtpVideoHdr = NULL) = 0;
+
+ virtual bool TimeToSendPacket(uint32_t ssrc,
+ uint16_t sequence_number,
+ int64_t capture_time_ms,
+ bool retransmission) = 0;
+
+ virtual size_t TimeToSendPadding(size_t bytes) = 0;
+
+ // Called on generation of new statistics after an RTP send.
+ virtual void RegisterSendChannelRtpStatisticsCallback(
+ StreamDataCountersCallback* callback) = 0;
+ virtual StreamDataCountersCallback*
+ GetSendChannelRtpStatisticsCallback() const = 0;
+
+ /**************************************************************************
+ *
+ * RTCP
+ *
+ ***************************************************************************/
+
+ /*
+ * Get RTCP status
+ */
+ virtual RtcpMode RTCP() const = 0;
+
+ /*
+ * configure RTCP status i.e on(compound or non- compound)/off
+ *
+ * method - RTCP method to use
+ */
+ virtual void SetRTCPStatus(RtcpMode method) = 0;
+
+ /*
+ * Set RTCP CName (i.e unique identifier)
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SetCNAME(const char* c_name) = 0;
+
+ /*
+ * Get remote CName
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t RemoteCNAME(uint32_t remoteSSRC,
+ char cName[RTCP_CNAME_SIZE]) const = 0;
+
+ /*
+ * Get remote NTP
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t RemoteNTP(
+ uint32_t *ReceivedNTPsecs,
+ uint32_t *ReceivedNTPfrac,
+ uint32_t *RTCPArrivalTimeSecs,
+ uint32_t *RTCPArrivalTimeFrac,
+ uint32_t *rtcp_timestamp) const = 0;
+
+ /*
+ * AddMixedCNAME
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0;
+
+ /*
+ * RemoveMixedCNAME
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t RemoveMixedCNAME(uint32_t SSRC) = 0;
+
+ /*
+ * Get RoundTripTime
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t RTT(uint32_t remoteSSRC,
+ int64_t* RTT,
+ int64_t* avgRTT,
+ int64_t* minRTT,
+ int64_t* maxRTT) const = 0;
+
+ /*
+ * Force a send of a RTCP packet
+ * periodic SR and RR are triggered via the process function
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0;
+
+ /*
+ * Force a send of a RTCP packet with more than one packet type.
+ * periodic SR and RR are triggered via the process function
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SendCompoundRTCP(
+ const std::set<RTCPPacketType>& rtcpPacketTypes) = 0;
+
+ /*
+ * Good state of RTP receiver inform sender
+ */
+ virtual int32_t SendRTCPReferencePictureSelection(
+ const uint64_t pictureID) = 0;
+
+ /*
+ * Send a RTCP Slice Loss Indication (SLI)
+ * 6 least significant bits of pictureID
+ */
+ virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0;
+
+ /*
+ * Statistics of the amount of data sent
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t DataCountersRTP(
+ size_t* bytesSent,
+ uint32_t* packetsSent) const = 0;
+
+ /*
+ * Get send statistics for the RTP and RTX stream.
+ */
+ virtual void GetSendStreamDataCounters(
+ StreamDataCounters* rtp_counters,
+ StreamDataCounters* rtx_counters) const = 0;
+
+ /*
+ * Get packet loss statistics for the RTP stream.
+ */
+ virtual void GetRtpPacketLossStats(
+ bool outgoing,
+ uint32_t ssrc,
+ struct RtpPacketLossStats* loss_stats) const = 0;
+
+ /*
+ * Get received RTCP sender info
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0;
+
+ /*
+ * Get received RTCP report block
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t RemoteRTCPStat(
+ std::vector<RTCPReportBlock>* receiveBlocks) const = 0;
+
+ /*
+ * (APP) Application specific data
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SetRTCPApplicationSpecificData(uint8_t subType,
+ uint32_t name,
+ const uint8_t* data,
+ uint16_t length) = 0;
+ /*
+ * (XR) VOIP metric
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SetRTCPVoIPMetrics(
+ const RTCPVoIPMetric* VoIPMetric) = 0;
+
+ /*
+ * (XR) Receiver Reference Time Report
+ */
+ virtual void SetRtcpXrRrtrStatus(bool enable) = 0;
+
+ virtual bool RtcpXrRrtrStatus() const = 0;
+
+ /*
+ * (REMB) Receiver Estimated Max Bitrate
+ */
+ virtual bool REMB() const = 0;
+
+ virtual void SetREMBStatus(bool enable) = 0;
+
+ virtual void SetREMBData(uint32_t bitrate,
+ const std::vector<uint32_t>& ssrcs) = 0;
+
+ /*
+ * (TMMBR) Temporary Max Media Bit Rate
+ */
+ virtual bool TMMBR() const = 0;
+
+ virtual void SetTMMBRStatus(bool enable) = 0;
+
+ /*
+ * (NACK)
+ */
+
+ /*
+ * TODO(holmer): Propagate this API to VideoEngine.
+ * Returns the currently configured selective retransmission settings.
+ */
+ virtual int SelectiveRetransmissions() const = 0;
+
+ /*
+ * TODO(holmer): Propagate this API to VideoEngine.
+ * Sets the selective retransmission settings, which will decide which
+ * packets will be retransmitted if NACKed. Settings are constructed by
+ * combining the constants in enum RetransmissionMode with bitwise OR.
+ * All packets are retransmitted if kRetransmitAllPackets is set, while no
+ * packets are retransmitted if kRetransmitOff is set.
+ * By default all packets except FEC packets are retransmitted. For VP8
+ * with temporal scalability only base layer packets are retransmitted.
+ *
+ * Returns -1 on failure, otherwise 0.
+ */
+ virtual int SetSelectiveRetransmissions(uint8_t settings) = 0;
+
+ /*
+ * Send a Negative acknowledgement packet
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SendNACK(const uint16_t* nackList, uint16_t size) = 0;
+
+ /*
+ * Store the sent packets, needed to answer to a Negative acknowledgement
+ * requests
+ */
+ virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
+
+ // Returns true if the module is configured to store packets.
+ virtual bool StorePackets() const = 0;
+
+ // Called on receipt of RTCP report block from remote side.
+ virtual void RegisterRtcpStatisticsCallback(
+ RtcpStatisticsCallback* callback) = 0;
+ virtual RtcpStatisticsCallback*
+ GetRtcpStatisticsCallback() = 0;
+ // BWE feedback packets.
+ virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0;
+
+ /**************************************************************************
+ *
+ * Audio
+ *
+ ***************************************************************************/
+
+ /*
+ * set audio packet size, used to determine when it's time to send a DTMF
+ * packet in silence (CNG)
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SetAudioPacketSize(uint16_t packetSizeSamples) = 0;
+
+ /*
+ * Send a TelephoneEvent tone using RFC 2833 (4733)
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SendTelephoneEventOutband(uint8_t key,
+ uint16_t time_ms,
+ uint8_t level) = 0;
+
+ /*
+ * Set payload type for Redundant Audio Data RFC 2198
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SetSendREDPayloadType(int8_t payloadType) = 0;
+
+ /*
+ * Get payload type for Redundant Audio Data RFC 2198
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SendREDPayloadType(
+ int8_t& payloadType) const = 0;
+
+ /*
+ * Store the audio level in dBov for header-extension-for-audio-level-
+ * indication.
+ * This API shall be called before transmision of an RTP packet to ensure
+ * that the |level| part of the extended RTP header is updated.
+ *
+ * return -1 on failure else 0.
+ */
+ virtual int32_t SetAudioLevel(uint8_t level_dBov) = 0;
+
+ /**************************************************************************
+ *
+ * Video
+ *
+ ***************************************************************************/
+
+ /*
+ * Set the target send bitrate
+ */
+ virtual void SetTargetSendBitrate(uint32_t bitrate_bps) = 0;
+
+ /*
+ * Turn on/off generic FEC
+ */
+ virtual void SetGenericFECStatus(bool enable,
+ uint8_t payload_type_red,
+ uint8_t payload_type_fec) = 0;
+
+ /*
+ * Get generic FEC setting
+ */
+ virtual void GenericFECStatus(bool& enable,
+ uint8_t& payloadTypeRED,
+ uint8_t& payloadTypeFEC) = 0;
+
+
+ virtual int32_t SetFecParameters(
+ const FecProtectionParams* delta_params,
+ const FecProtectionParams* key_params) = 0;
+
+ /*
+ * Set method for requestion a new key frame
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
+
+ /*
+ * send a request for a keyframe
+ *
+ * return -1 on failure else 0
+ */
+ virtual int32_t RequestKeyFrame() = 0;
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
new file mode 100644
index 0000000..6373de2
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -0,0 +1,440 @@
+/*
+ * 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_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+
+#include <stddef.h>
+#include <list>
+
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/typedefs.h"
+
+#define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination
+#define IP_PACKET_SIZE 1500 // we assume ethernet
+#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
+#define TIMEOUT_SEI_MESSAGES_MS 30000 // in milliseconds
+
+namespace webrtc {
+namespace rtcp {
+class TransportFeedback;
+}
+
+const int kVideoPayloadTypeFrequency = 90000;
+
+// Minimum RTP header size in bytes.
+const uint8_t kRtpHeaderSize = 12;
+
+struct AudioPayload
+{
+ uint32_t frequency;
+ uint8_t channels;
+ uint32_t rate;
+};
+
+struct VideoPayload
+{
+ RtpVideoCodecTypes videoCodecType;
+ uint32_t maxRate;
+};
+
+union PayloadUnion
+{
+ AudioPayload Audio;
+ VideoPayload Video;
+};
+
+enum RTPAliveType
+{
+ kRtpDead = 0,
+ kRtpNoRtp = 1,
+ kRtpAlive = 2
+};
+
+enum ProtectionType {
+ kUnprotectedPacket,
+ kProtectedPacket
+};
+
+enum StorageType {
+ kDontRetransmit,
+ kAllowRetransmission
+};
+
+enum RTPExtensionType {
+ kRtpExtensionNone,
+ kRtpExtensionTransmissionTimeOffset,
+ kRtpExtensionAudioLevel,
+ kRtpExtensionAbsoluteSendTime,
+ kRtpExtensionVideoRotation,
+ kRtpExtensionTransportSequenceNumber,
+};
+
+enum RTCPAppSubTypes
+{
+ kAppSubtypeBwe = 0x00
+};
+
+// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
+enum RTCPPacketType : uint32_t {
+ kRtcpReport = 0x0001,
+ kRtcpSr = 0x0002,
+ kRtcpRr = 0x0004,
+ kRtcpSdes = 0x0008,
+ kRtcpBye = 0x0010,
+ kRtcpPli = 0x0020,
+ kRtcpNack = 0x0040,
+ kRtcpFir = 0x0080,
+ kRtcpTmmbr = 0x0100,
+ kRtcpTmmbn = 0x0200,
+ kRtcpSrReq = 0x0400,
+ kRtcpXrVoipMetric = 0x0800,
+ kRtcpApp = 0x1000,
+ kRtcpSli = 0x4000,
+ kRtcpRpsi = 0x8000,
+ kRtcpRemb = 0x10000,
+ kRtcpTransmissionTimeOffset = 0x20000,
+ kRtcpXrReceiverReferenceTime = 0x40000,
+ kRtcpXrDlrrReportBlock = 0x80000,
+ kRtcpTransportFeedback = 0x100000,
+};
+
+enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };
+
+enum RtpRtcpPacketType
+{
+ kPacketRtp = 0,
+ kPacketKeepAlive = 1
+};
+
+enum NACKMethod
+{
+ kNackOff = 0,
+ kNackRtcp = 2
+};
+
+enum RetransmissionMode : uint8_t {
+ kRetransmitOff = 0x0,
+ kRetransmitFECPackets = 0x1,
+ kRetransmitBaseLayer = 0x2,
+ kRetransmitHigherLayers = 0x4,
+ kRetransmitAllPackets = 0xFF
+};
+
+enum RtxMode {
+ kRtxOff = 0x0,
+ kRtxRetransmitted = 0x1, // Only send retransmissions over RTX.
+ kRtxRedundantPayloads = 0x2 // Preventively send redundant payloads
+ // instead of padding.
+};
+
+const size_t kRtxHeaderSize = 2;
+
+struct RTCPSenderInfo
+{
+ uint32_t NTPseconds;
+ uint32_t NTPfraction;
+ uint32_t RTPtimeStamp;
+ uint32_t sendPacketCount;
+ uint32_t sendOctetCount;
+};
+
+struct RTCPReportBlock {
+ RTCPReportBlock()
+ : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0),
+ extendedHighSeqNum(0), jitter(0), lastSR(0),
+ delaySinceLastSR(0) {}
+
+ RTCPReportBlock(uint32_t remote_ssrc,
+ uint32_t source_ssrc,
+ uint8_t fraction_lost,
+ uint32_t cumulative_lost,
+ uint32_t extended_high_sequence_number,
+ uint32_t jitter,
+ uint32_t last_sender_report,
+ uint32_t delay_since_last_sender_report)
+ : remoteSSRC(remote_ssrc),
+ sourceSSRC(source_ssrc),
+ fractionLost(fraction_lost),
+ cumulativeLost(cumulative_lost),
+ extendedHighSeqNum(extended_high_sequence_number),
+ jitter(jitter),
+ lastSR(last_sender_report),
+ delaySinceLastSR(delay_since_last_sender_report) {}
+
+ // Fields as described by RFC 3550 6.4.2.
+ uint32_t remoteSSRC; // SSRC of sender of this report.
+ uint32_t sourceSSRC; // SSRC of the RTP packet sender.
+ uint8_t fractionLost;
+ uint32_t cumulativeLost; // 24 bits valid.
+ uint32_t extendedHighSeqNum;
+ uint32_t jitter;
+ uint32_t lastSR;
+ uint32_t delaySinceLastSR;
+};
+
+struct RtcpReceiveTimeInfo {
+ // Fields as described by RFC 3611 4.5.
+ uint32_t sourceSSRC;
+ uint32_t lastRR;
+ uint32_t delaySinceLastRR;
+};
+
+typedef std::list<RTCPReportBlock> ReportBlockList;
+
+struct RtpState {
+ RtpState()
+ : sequence_number(0),
+ start_timestamp(0),
+ timestamp(0),
+ capture_time_ms(-1),
+ last_timestamp_time_ms(-1),
+ media_has_been_sent(false) {}
+ uint16_t sequence_number;
+ uint32_t start_timestamp;
+ uint32_t timestamp;
+ int64_t capture_time_ms;
+ int64_t last_timestamp_time_ms;
+ bool media_has_been_sent;
+};
+
+class RtpData
+{
+public:
+ virtual ~RtpData() {}
+
+ virtual int32_t OnReceivedPayloadData(
+ const uint8_t* payloadData,
+ const size_t payloadSize,
+ const WebRtcRTPHeader* rtpHeader) = 0;
+
+ virtual bool OnRecoveredPacket(const uint8_t* packet,
+ size_t packet_length) = 0;
+};
+
+class RtpFeedback
+{
+public:
+ virtual ~RtpFeedback() {}
+
+ // Receiving payload change or SSRC change. (return success!)
+ /*
+ * channels - number of channels in codec (1 = mono, 2 = stereo)
+ */
+ virtual int32_t OnInitializeDecoder(
+ const int8_t payloadType,
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const int frequency,
+ const uint8_t channels,
+ const uint32_t rate) = 0;
+
+ virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
+
+ virtual void OnIncomingCSRCChanged(const uint32_t CSRC,
+ const bool added) = 0;
+};
+
+class RtpAudioFeedback {
+ public:
+ virtual void OnPlayTelephoneEvent(const uint8_t event,
+ const uint16_t lengthMs,
+ const uint8_t volume) = 0;
+
+ protected:
+ virtual ~RtpAudioFeedback() {}
+};
+
+class RtcpIntraFrameObserver {
+ public:
+ virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
+
+ virtual void OnReceivedSLI(uint32_t ssrc,
+ uint8_t picture_id) = 0;
+
+ virtual void OnReceivedRPSI(uint32_t ssrc,
+ uint64_t picture_id) = 0;
+
+ virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) = 0;
+
+ virtual ~RtcpIntraFrameObserver() {}
+};
+
+class RtcpBandwidthObserver {
+ public:
+ // REMB or TMMBR
+ virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;
+
+ virtual void OnReceivedRtcpReceiverReport(
+ const ReportBlockList& report_blocks,
+ int64_t rtt,
+ int64_t now_ms) = 0;
+
+ virtual ~RtcpBandwidthObserver() {}
+};
+
+struct PacketInfo {
+ PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number)
+ : PacketInfo(-1, arrival_time_ms, -1, sequence_number, 0, false) {}
+
+ PacketInfo(int64_t arrival_time_ms,
+ int64_t send_time_ms,
+ uint16_t sequence_number,
+ size_t payload_size,
+ bool was_paced)
+ : PacketInfo(-1,
+ arrival_time_ms,
+ send_time_ms,
+ sequence_number,
+ payload_size,
+ was_paced) {}
+
+ PacketInfo(int64_t creation_time_ms,
+ int64_t arrival_time_ms,
+ int64_t send_time_ms,
+ uint16_t sequence_number,
+ size_t payload_size,
+ bool was_paced)
+ : creation_time_ms(creation_time_ms),
+ arrival_time_ms(arrival_time_ms),
+ send_time_ms(send_time_ms),
+ sequence_number(sequence_number),
+ payload_size(payload_size),
+ was_paced(was_paced) {}
+
+ // Time corresponding to when this object was created.
+ int64_t creation_time_ms;
+ // Time corresponding to when the packet was received. Timestamped with the
+ // receiver's clock.
+ int64_t arrival_time_ms;
+ // Time corresponding to when the packet was sent, timestamped with the
+ // sender's clock.
+ int64_t send_time_ms;
+ // Packet identifier, incremented with 1 for every packet generated by the
+ // sender.
+ uint16_t sequence_number;
+ // Size of the packet excluding RTP headers.
+ size_t payload_size;
+ // True if the packet was paced out by the pacer.
+ bool was_paced;
+};
+
+class TransportFeedbackObserver {
+ public:
+ TransportFeedbackObserver() {}
+ virtual ~TransportFeedbackObserver() {}
+
+ // Note: Transport-wide sequence number as sequence number. Arrival time
+ // must be set to 0.
+ virtual void AddPacket(uint16_t sequence_number,
+ size_t length,
+ bool was_paced) = 0;
+
+ virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
+};
+
+class RtcpRttStats {
+ public:
+ virtual void OnRttUpdate(int64_t rtt) = 0;
+
+ virtual int64_t LastProcessedRtt() const = 0;
+
+ virtual ~RtcpRttStats() {};
+};
+
+// Null object version of RtpFeedback.
+class NullRtpFeedback : public RtpFeedback {
+ public:
+ virtual ~NullRtpFeedback() {}
+
+ int32_t OnInitializeDecoder(const int8_t payloadType,
+ const char payloadName[RTP_PAYLOAD_NAME_SIZE],
+ const int frequency,
+ const uint8_t channels,
+ const uint32_t rate) override {
+ return 0;
+ }
+
+ void OnIncomingSSRCChanged(const uint32_t ssrc) override {}
+ void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
+};
+
+// Null object version of RtpData.
+class NullRtpData : public RtpData {
+ public:
+ virtual ~NullRtpData() {}
+
+ int32_t OnReceivedPayloadData(const uint8_t* payloadData,
+ const size_t payloadSize,
+ const WebRtcRTPHeader* rtpHeader) override {
+ return 0;
+ }
+
+ bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
+ return true;
+ }
+};
+
+// Null object version of RtpAudioFeedback.
+class NullRtpAudioFeedback : public RtpAudioFeedback {
+ public:
+ virtual ~NullRtpAudioFeedback() {}
+
+ void OnPlayTelephoneEvent(const uint8_t event,
+ const uint16_t lengthMs,
+ const uint8_t volume) override {}
+};
+
+// Statistics about packet loss for a single directional connection. All values
+// are totals since the connection initiated.
+struct RtpPacketLossStats {
+ // The number of packets lost in events where no adjacent packets were also
+ // lost.
+ uint64_t single_packet_loss_count;
+ // The number of events in which more than one adjacent packet was lost.
+ uint64_t multiple_packet_loss_event_count;
+ // The number of packets lost in events where more than one adjacent packet
+ // was lost.
+ uint64_t multiple_packet_loss_packet_count;
+};
+
+class RtpPacketSender {
+ public:
+ RtpPacketSender() {}
+ virtual ~RtpPacketSender() {}
+
+ enum Priority {
+ kHighPriority = 0, // Pass through; will be sent immediately.
+ kNormalPriority = 2, // Put in back of the line.
+ kLowPriority = 3, // Put in back of the low priority line.
+ };
+ // Low priority packets are mixed with the normal priority packets
+ // while we are paused.
+
+ // Returns true if we send the packet now, else it will add the packet
+ // information to the queue and call TimeToSendPacket when it's time to send.
+ virtual void InsertPacket(Priority priority,
+ uint32_t ssrc,
+ uint16_t sequence_number,
+ int64_t capture_time_ms,
+ size_t bytes,
+ bool retransmission) = 0;
+};
+
+class TransportSequenceNumberAllocator {
+ public:
+ TransportSequenceNumberAllocator() {}
+ virtual ~TransportSequenceNumberAllocator() {}
+
+ virtual uint16_t AllocateSequenceNumber() = 0;
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/fec_receiver.h b/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
index 3608165..ec7ec39 100644
--- a/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
+++ b/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include")
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
@@ -43,4 +45,4 @@
virtual FecPacketCounter GetPacketCounter() const = 0;
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_FEC_RECEIVER_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
index 6bd5cd8..183eb2e 100644
--- a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
+++ b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
@@ -8,13 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
#include <map>
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -99,4 +101,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RECEIVE_STATISTICS_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
index 0ffba21..d271c9b 100644
--- a/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
+++ b/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
@@ -48,4 +50,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_REMOTE_NTP_TIME_ESTIMATOR_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h b/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
index c7a0268..6d942cb 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
@@ -7,8 +7,10 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
#include "webrtc/common_video/rotation.h"
@@ -51,4 +53,4 @@
}
} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_CVO__H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h b/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
index 2809996..9b2cabc 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
@@ -7,10 +7,12 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -41,4 +43,4 @@
virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0;
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_HEADER_PARSER_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
index 313bef1..4994eb5 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
@@ -190,4 +192,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_PAYLOAD_REGISTRY_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h b/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
index 2fb8ac5..db5f46e 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
@@ -8,10 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -100,4 +102,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RECEIVER_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
index f907408..d004f67 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
@@ -8,14 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
#include <set>
#include <vector>
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
// Forward declarations.
@@ -638,4 +640,4 @@
virtual int32_t RequestKeyFrame() = 0;
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
index 6936352..ac3954b 100644
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
@@ -8,13 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
+
+#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
#include <stddef.h>
#include <list>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/typedefs.h"
@@ -437,4 +439,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
+#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
index bc4aec8..ca5434e 100644
--- a/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
@@ -13,9 +13,9 @@
#include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
index e35a75c..57bea0b 100644
--- a/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
+++ b/webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
@@ -17,14 +17,14 @@
],
'sources': [
# Common
- 'interface/fec_receiver.h',
- 'interface/receive_statistics.h',
- 'interface/remote_ntp_time_estimator.h',
- 'interface/rtp_header_parser.h',
- 'interface/rtp_payload_registry.h',
- 'interface/rtp_receiver.h',
- 'interface/rtp_rtcp.h',
- 'interface/rtp_rtcp_defines.h',
+ 'include/fec_receiver.h',
+ 'include/receive_statistics.h',
+ 'include/remote_ntp_time_estimator.h',
+ 'include/rtp_header_parser.h',
+ 'include/rtp_payload_registry.h',
+ 'include/rtp_receiver.h',
+ 'include/rtp_rtcp.h',
+ 'include/rtp_rtcp_defines.h',
'source/bitrate.cc',
'source/bitrate.h',
'source/byte_io.h',
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
index 24db39b..b79f6ba 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
@@ -14,8 +14,8 @@
// This header is included to get the nested declaration of Packet structure.
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
index f64b537..bb22e1d 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/fec_receiver_unittest.cc
@@ -15,8 +15,8 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/fec_test_helper.h"
diff --git a/webrtc/modules/rtp_rtcp/source/fec_test_helper.h b/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
index e1791ad..a5de924 100644
--- a/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
+++ b/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_TEST_HELPER_H_
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
index aad418f..7a9a8be 100644
--- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
+++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
@@ -18,7 +18,7 @@
#include <iterator>
#include "webrtc/base/logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.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/forward_error_correction_internal.h"
diff --git a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
index f92f014..9ba6ce0 100644
--- a/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
+++ b/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
@@ -15,7 +15,7 @@
#include <vector>
#include "webrtc/base/scoped_ref_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/ref_count.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
index f577cba..549821f 100644
--- a/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h
@@ -12,7 +12,7 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_MOCK_MOCK_RTP_PAYLOAD_REGISTRY_H_
#include "testing/gmock/include/gmock/gmock.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 07a3693..d95c963 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -16,12 +16,12 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/transport.h"
using namespace webrtc;
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
index fe42990..0cd8012 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVE_STATISTICS_IMPL_H_
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include <algorithm>
diff --git a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index 5b52228..c265c17 100644
--- a/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -11,7 +11,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
index 74fc9cd..ccc15ec 100644
--- a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
+++ b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "webrtc/base/logging.h"
#include "webrtc/system_wrappers/include/clock.h"
diff --git a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
index bc9cf2e..797c788 100644
--- a/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
@@ -11,7 +11,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "webrtc/system_wrappers/include/clock.h"
using ::testing::_;
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
index 3c34957..2956bc7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
@@ -18,7 +18,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
index 4cc1f38..ad6fd16 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
@@ -15,7 +15,7 @@
#include <vector>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
index 2723976..fc43de5 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -16,7 +16,7 @@
#include <set>
#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
index 37b7b88..a697fd2 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h
@@ -14,7 +14,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" // RTCPReportBlock
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPReportBlock
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
index 9ec9283..1bdd6d1 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
@@ -20,8 +20,8 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
index f05d512..e1f8737 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_utility.h
@@ -14,7 +14,7 @@
#include <stddef.h> // size_t, ptrdiff_t
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.h b/webrtc/modules/rtp_rtcp/source/rtp_format.h
index 18225f9..3519499 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.h
@@ -14,8 +14,8 @@
#include <string>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index aeef443..c422577 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -11,7 +11,7 @@
#include <string.h>
#include "webrtc/base/logging.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/h264_sps_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index 1a14b55..d29e3d4 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -13,7 +13,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
index 39b64c6..b47e9b9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
@@ -11,7 +11,7 @@
#include <string>
#include "webrtc/base/logging.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
index 63db349..d62ecba 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
@@ -30,7 +30,7 @@
#include <vector>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
index 2fe9632..6684768 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
@@ -19,7 +19,7 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
index abce7e7..3feca43 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
@@ -25,7 +25,7 @@
#include <string>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
index 7be3c2e..263b2d7 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -13,7 +13,7 @@
#include <map>
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
index 520cf7a..ca37750 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
@@ -15,7 +15,7 @@
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
index 266bad8..82c813f 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
@@ -7,7 +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/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
index e97d11e..cece640 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
@@ -16,8 +16,8 @@
#include <vector>
#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
index 00a6ac7..d39991e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
@@ -12,7 +12,7 @@
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/video_engine/vie_defines.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
index 38d1450..a5bfd71 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
index 0b9bf27..e20c94e 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/mock/mock_rtp_payload_strategy.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
index 176852e..b2413f3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
@@ -14,8 +14,8 @@
#include <set>
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index e1ebd0c..bd7bd1b 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -16,8 +16,8 @@
#include <string.h>
#include "webrtc/base/logging.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
index 46741d5..4ec3531 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
@@ -12,8 +12,8 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
index 37c3e6e..0e0a34a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
@@ -12,8 +12,8 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_STRATEGY_H_
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
index 1af2d48..53051dd 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
@@ -16,8 +16,8 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
index 23128df..56f761a 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
@@ -12,7 +12,7 @@
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index c9b6686..ada6667 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -15,7 +15,7 @@
#include <vector>
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
index 4c94764..a5e2076 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
@@ -12,8 +12,8 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "webrtc/system_wrappers/include/scoped_vector.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 50f4768..dc544fb 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -16,7 +16,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index a134370..8fc0696 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -17,7 +17,7 @@
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 1fc9a89..f5df5b3 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -14,7 +14,7 @@
#include <string.h> //memcpy
#include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index fde6d47..332a0f8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -17,9 +17,9 @@
#include "webrtc/base/buffer.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
index 6606277..0209510 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -18,7 +18,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.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/producer_fec.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index f412542..03ed600 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -16,7 +16,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
#include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
index af20f97..1c615e6 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_utility.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_utility.h
@@ -13,8 +13,8 @@
#include <stddef.h> // size_t, ptrdiff_t
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
index 53b678f..4716dfb 100644
--- a/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
+++ b/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
@@ -14,7 +14,7 @@
#include <vector>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
index 711be4a..2d70289 100644
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
+++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
@@ -14,7 +14,7 @@
#include <stdio.h>
#include <string>
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/test/channel_transport/udp_transport.h"
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
index bd83962..59742b2 100644
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
+++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
@@ -13,7 +13,7 @@
#include <stdlib.h>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/thread_wrapper.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
index 47f2880..c1d91cd 100644
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
+++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
@@ -13,7 +13,7 @@
#include <stdio.h>
#include <stdlib.h>
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
diff --git a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
index f394412..02fbe0a 100644
--- a/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
+++ b/webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
@@ -11,8 +11,8 @@
#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/thread_wrapper.h"
#include "webrtc/test/channel_transport/udp_transport.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
index 73334a8..f55b4b5 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
@@ -11,12 +11,12 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/transport.h"
namespace webrtc {
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 745386d..36456f1 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -15,8 +15,8 @@
#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
using namespace webrtc;
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
index e9d8112..8faa2dd 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
@@ -14,9 +14,9 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
index 30a6a1c..16ea540 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
@@ -15,9 +15,9 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.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/rtp_receiver_video.h"
#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn
index 163515c..6704cd6 100644
--- a/webrtc/modules/utility/BUILD.gn
+++ b/webrtc/modules/utility/BUILD.gn
@@ -10,12 +10,12 @@
source_set("utility") {
sources = [
- "interface/audio_frame_operations.h",
- "interface/file_player.h",
- "interface/file_recorder.h",
- "interface/helpers_android.h",
- "interface/jvm_android.h",
- "interface/process_thread.h",
+ "include/audio_frame_operations.h",
+ "include/file_player.h",
+ "include/file_recorder.h",
+ "include/helpers_android.h",
+ "include/jvm_android.h",
+ "include/process_thread.h",
"source/audio_frame_operations.cc",
"source/coder.cc",
"source/coder.h",
diff --git a/webrtc/modules/utility/include/audio_frame_operations.h b/webrtc/modules/utility/include/audio_frame_operations.h
new file mode 100644
index 0000000..1551d86
--- /dev/null
+++ b/webrtc/modules/utility/include/audio_frame_operations.h
@@ -0,0 +1,58 @@
+/*
+ * 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_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class AudioFrame;
+
+// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
+// Change reference parameters to pointers. Consider using a namespace rather
+// than a class.
+class AudioFrameOperations {
+ public:
+ // Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
+ // operation, meaning src_audio and dst_audio must point to different
+ // buffers. It is the caller's responsibility to ensure that |dst_audio| is
+ // sufficiently large.
+ static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel,
+ int16_t* dst_audio);
+ // |frame.num_channels_| will be updated. This version checks for sufficient
+ // buffer size and that |num_channels_| is mono.
+ static int MonoToStereo(AudioFrame* frame);
+
+ // Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
+ // operation, meaning |src_audio| and |dst_audio| may point to the same
+ // buffer.
+ static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel,
+ int16_t* dst_audio);
+ // |frame.num_channels_| will be updated. This version checks that
+ // |num_channels_| is stereo.
+ static int StereoToMono(AudioFrame* frame);
+
+ // Swap the left and right channels of |frame|. Fails silently if |frame| is
+ // not stereo.
+ static void SwapStereoChannels(AudioFrame* frame);
+
+ // Zeros out the audio and sets |frame.energy| to zero.
+ static void Mute(AudioFrame& frame);
+
+ static int Scale(float left, float right, AudioFrame& frame);
+
+ static int ScaleWithSat(float scale, AudioFrame& frame);
+};
+
+} // namespace webrtc
+
+#endif // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
diff --git a/webrtc/modules/utility/include/file_player.h b/webrtc/modules/utility/include/file_player.h
new file mode 100644
index 0000000..4ca134a
--- /dev/null
+++ b/webrtc/modules/utility/include/file_player.h
@@ -0,0 +1,111 @@
+/*
+ * 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_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+class FileCallback;
+
+class FilePlayer
+{
+public:
+ // The largest decoded frame size in samples (60ms with 32kHz sample rate).
+ enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
+ enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
+
+ // Note: will return NULL for unsupported formats.
+ static FilePlayer* CreateFilePlayer(const uint32_t instanceID,
+ const FileFormats fileFormat);
+
+ static void DestroyFilePlayer(FilePlayer* player);
+
+ // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
+ // will be set to the number of samples read (not the number of samples per
+ // channel).
+ virtual int Get10msAudioFromFile(
+ int16_t* outBuffer,
+ size_t& lengthInSamples,
+ int frequencyInHz) = 0;
+
+ // Register callback for receiving file playing notifications.
+ virtual int32_t RegisterModuleFileCallback(
+ FileCallback* callback) = 0;
+
+ // API for playing audio from fileName to channel.
+ // Note: codecInst is used for pre-encoded files.
+ virtual int32_t StartPlayingFile(
+ const char* fileName,
+ bool loop,
+ uint32_t startPosition,
+ float volumeScaling,
+ uint32_t notification,
+ uint32_t stopPosition = 0,
+ const CodecInst* codecInst = NULL) = 0;
+
+ // Note: codecInst is used for pre-encoded files.
+ virtual int32_t StartPlayingFile(
+ InStream& sourceStream,
+ uint32_t startPosition,
+ float volumeScaling,
+ uint32_t notification,
+ uint32_t stopPosition = 0,
+ const CodecInst* codecInst = NULL) = 0;
+
+ virtual int32_t StopPlayingFile() = 0;
+
+ virtual bool IsPlayingFile() const = 0;
+
+ virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0;
+
+ // Set audioCodec to the currently used audio codec.
+ virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0;
+
+ virtual int32_t Frequency() const = 0;
+
+ // Note: scaleFactor is in the range [0.0 - 2.0]
+ virtual int32_t SetAudioScaling(float scaleFactor) = 0;
+
+ // Return the time in ms until next video frame should be pulled (by
+ // calling GetVideoFromFile(..)).
+ // Note: this API reads one video frame from file. This means that it should
+ // be called exactly once per GetVideoFromFile(..) API call.
+ virtual int32_t TimeUntilNextVideoFrame() { return -1;}
+
+ virtual int32_t StartPlayingVideoFile(
+ const char* /*fileName*/,
+ bool /*loop*/,
+ bool /*videoOnly*/) { return -1;}
+
+ virtual int32_t video_codec_info(VideoCodec& /*videoCodec*/) const
+ {return -1;}
+
+ virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/) { return -1; }
+
+ // Same as GetVideoFromFile(). videoFrame will have the resolution specified
+ // by the width outWidth and height outHeight in pixels.
+ virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/,
+ const uint32_t /*outWidth*/,
+ const uint32_t /*outHeight*/) {
+ return -1;
+ }
+
+protected:
+ virtual ~FilePlayer() {}
+
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
diff --git a/webrtc/modules/utility/include/file_recorder.h b/webrtc/modules/utility/include/file_recorder.h
new file mode 100644
index 0000000..0c285d4
--- /dev/null
+++ b/webrtc/modules/utility/include/file_recorder.h
@@ -0,0 +1,84 @@
+/*
+ * 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_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+
+#include "webrtc/common_types.h"
+#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
+#include "webrtc/system_wrappers/include/tick_util.h"
+#include "webrtc/typedefs.h"
+#include "webrtc/video_frame.h"
+
+namespace webrtc {
+
+class FileRecorder
+{
+public:
+
+ // Note: will return NULL for unsupported formats.
+ static FileRecorder* CreateFileRecorder(const uint32_t instanceID,
+ const FileFormats fileFormat);
+
+ static void DestroyFileRecorder(FileRecorder* recorder);
+
+ virtual int32_t RegisterModuleFileCallback(
+ FileCallback* callback) = 0;
+
+ virtual FileFormats RecordingFileFormat() const = 0;
+
+ virtual int32_t StartRecordingAudioFile(
+ const char* fileName,
+ const CodecInst& codecInst,
+ uint32_t notification) = 0;
+
+ virtual int32_t StartRecordingAudioFile(
+ OutStream& destStream,
+ const CodecInst& codecInst,
+ uint32_t notification) = 0;
+
+ // Stop recording.
+ // Note: this API is for both audio and video.
+ virtual int32_t StopRecording() = 0;
+
+ // Return true if recording.
+ // Note: this API is for both audio and video.
+ virtual bool IsRecording() const = 0;
+
+ virtual int32_t codec_info(CodecInst& codecInst) const = 0;
+
+ // Write frame to file. Frame should contain 10ms of un-ecoded audio data.
+ virtual int32_t RecordAudioToFile(
+ const AudioFrame& frame,
+ const TickTime* playoutTS = NULL) = 0;
+
+ // Open/create the file specified by fileName for writing audio/video data
+ // (relative path is allowed). audioCodecInst specifies the encoding of the
+ // audio data. videoCodecInst specifies the encoding of the video data.
+ // Only video data will be recorded if videoOnly is true. amrFormat
+ // specifies the amr/amrwb storage format.
+ // Note: the file format is AVI.
+ virtual int32_t StartRecordingVideoFile(
+ const char* fileName,
+ const CodecInst& audioCodecInst,
+ const VideoCodec& videoCodecInst,
+ bool videoOnly = false) = 0;
+
+ // Record the video frame in videoFrame to AVI file.
+ virtual int32_t RecordVideoToFile(const VideoFrame& videoFrame) = 0;
+
+protected:
+ virtual ~FileRecorder() {}
+
+};
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
diff --git a/webrtc/modules/utility/include/helpers_android.h b/webrtc/modules/utility/include/helpers_android.h
new file mode 100644
index 0000000..2840ca9
--- /dev/null
+++ b/webrtc/modules/utility/include/helpers_android.h
@@ -0,0 +1,87 @@
+/*
+ * 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_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+
+#include <jni.h>
+#include <string>
+
+// Abort the process if |jni| has a Java exception pending.
+// TODO(henrika): merge with CHECK_JNI_EXCEPTION() in jni_helpers.h.
+#define CHECK_EXCEPTION(jni) \
+ RTC_CHECK(!jni->ExceptionCheck()) \
+ << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
+
+namespace webrtc {
+
+// Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
+JNIEnv* GetEnv(JavaVM* jvm);
+
+// Return a |jlong| that will correctly convert back to |ptr|. This is needed
+// because the alternative (of silently passing a 32-bit pointer to a vararg
+// function expecting a 64-bit param) picks up garbage in the high 32 bits.
+jlong PointerTojlong(void* ptr);
+
+// JNIEnv-helper methods that wraps the API which uses the JNI interface
+// pointer (JNIEnv*). It allows us to RTC_CHECK success and that no Java
+// exception is thrown while calling the method.
+jmethodID GetMethodID(
+ JNIEnv* jni, jclass c, const char* name, const char* signature);
+
+jmethodID GetStaticMethodID(
+ JNIEnv* jni, jclass c, const char* name, const char* signature);
+
+jclass FindClass(JNIEnv* jni, const char* name);
+
+jobject NewGlobalRef(JNIEnv* jni, jobject o);
+
+void DeleteGlobalRef(JNIEnv* jni, jobject o);
+
+// Return thread ID as a string.
+std::string GetThreadId();
+
+// Return thread ID as string suitable for debug logging.
+std::string GetThreadInfo();
+
+// Attach thread to JVM if necessary and detach at scope end if originally
+// attached.
+class AttachThreadScoped {
+ public:
+ explicit AttachThreadScoped(JavaVM* jvm);
+ ~AttachThreadScoped();
+ JNIEnv* env();
+
+ private:
+ bool attached_;
+ JavaVM* jvm_;
+ JNIEnv* env_;
+};
+
+// Scoped holder for global Java refs.
+template<class T> // T is jclass, jobject, jintArray, etc.
+class ScopedGlobalRef {
+ public:
+ ScopedGlobalRef(JNIEnv* jni, T obj)
+ : jni_(jni), obj_(static_cast<T>(NewGlobalRef(jni, obj))) {}
+ ~ScopedGlobalRef() {
+ DeleteGlobalRef(jni_, obj_);
+ }
+ T operator*() const {
+ return obj_;
+ }
+ private:
+ JNIEnv* jni_;
+ T obj_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
diff --git a/webrtc/modules/utility/include/helpers_ios.h b/webrtc/modules/utility/include/helpers_ios.h
new file mode 100644
index 0000000..a5a07ac
--- /dev/null
+++ b/webrtc/modules/utility/include/helpers_ios.h
@@ -0,0 +1,59 @@
+/*
+ * 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_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+
+#if defined(WEBRTC_IOS)
+
+#include <string>
+
+namespace webrtc {
+namespace ios {
+
+bool CheckAndLogError(BOOL success, NSError* error);
+
+std::string StdStringFromNSString(NSString* nsString);
+
+// Return thread ID as a string.
+std::string GetThreadId();
+
+// Return thread ID as string suitable for debug logging.
+std::string GetThreadInfo();
+
+// Returns [NSThread currentThread] description as string.
+// Example: <NSThread: 0x170066d80>{number = 1, name = main}
+std::string GetCurrentThreadDescription();
+
+std::string GetAudioSessionCategory();
+
+// Returns the current name of the operating system.
+std::string GetSystemName();
+
+// Returns the current version of the operating system.
+std::string GetSystemVersion();
+
+// Returns the version of the operating system as a floating point value.
+float GetSystemVersionAsFloat();
+
+// Returns the device type.
+// Examples: ”iPhone” and ”iPod touch”.
+std::string GetDeviceType();
+
+// Returns a more detailed device name.
+// Examples: "iPhone 5s (GSM)" and "iPhone 6 Plus".
+std::string GetDeviceName();
+
+} // namespace ios
+} // namespace webrtc
+
+#endif // defined(WEBRTC_IOS)
+
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
diff --git a/webrtc/modules/utility/include/jvm_android.h b/webrtc/modules/utility/include/jvm_android.h
new file mode 100644
index 0000000..f527dff
--- /dev/null
+++ b/webrtc/modules/utility/include/jvm_android.h
@@ -0,0 +1,185 @@
+/*
+ * 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_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+
+#include <jni.h>
+#include <string>
+
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
+
+namespace webrtc {
+
+// The JNI interface pointer (JNIEnv) is valid only in the current thread.
+// Should another thread need to access the Java VM, it must first call
+// AttachCurrentThread() to attach itself to the VM and obtain a JNI interface
+// pointer. The native thread remains attached to the VM until it calls
+// DetachCurrentThread() to detach.
+class AttachCurrentThreadIfNeeded {
+ public:
+ AttachCurrentThreadIfNeeded();
+ ~AttachCurrentThreadIfNeeded();
+
+ private:
+ rtc::ThreadChecker thread_checker_;
+ bool attached_;
+};
+
+// This class is created by the NativeRegistration class and is used to wrap
+// the actual Java object handle (jobject) on which we can call methods from
+// C++ in to Java. See example in JVM for more details.
+// TODO(henrika): extend support for type of function calls.
+class GlobalRef {
+ public:
+ GlobalRef(JNIEnv* jni, jobject object);
+ ~GlobalRef();
+
+ jboolean CallBooleanMethod(jmethodID methodID, ...);
+ jint CallIntMethod(jmethodID methodID, ...);
+ void CallVoidMethod(jmethodID methodID, ...);
+
+ private:
+ JNIEnv* const jni_;
+ const jobject j_object_;
+};
+
+// Wraps the jclass object on which we can call GetMethodId() functions to
+// query method IDs.
+class JavaClass {
+ public:
+ JavaClass(JNIEnv* jni, jclass clazz) : jni_(jni), j_class_(clazz) {}
+ ~JavaClass() {}
+
+ jmethodID GetMethodId(const char* name, const char* signature);
+ jmethodID GetStaticMethodId(const char* name, const char* signature);
+ jobject CallStaticObjectMethod(jmethodID methodID, ...);
+
+ protected:
+ JNIEnv* const jni_;
+ jclass const j_class_;
+};
+
+// Adds support of the NewObject factory method to the JavaClass class.
+// See example in JVM for more details on how to use it.
+class NativeRegistration : public JavaClass {
+ public:
+ NativeRegistration(JNIEnv* jni, jclass clazz);
+ ~NativeRegistration();
+
+ rtc::scoped_ptr<GlobalRef> NewObject(
+ const char* name, const char* signature, ...);
+
+ private:
+ JNIEnv* const jni_;
+};
+
+// This class is created by the JVM class and is used to expose methods that
+// needs the JNI interface pointer but its main purpose is to create a
+// NativeRegistration object given name of a Java class and a list of native
+// methods. See example in JVM for more details.
+class JNIEnvironment {
+ public:
+ explicit JNIEnvironment(JNIEnv* jni);
+ ~JNIEnvironment();
+
+ // Registers native methods with the Java class specified by |name|.
+ // Note that the class name must be one of the names in the static
+ // |loaded_classes| array defined in jvm_android.cc.
+ // This method must be called on the construction thread.
+ rtc::scoped_ptr<NativeRegistration> RegisterNatives(
+ const char* name, const JNINativeMethod *methods, int num_methods);
+
+ // Converts from Java string to std::string.
+ // This method must be called on the construction thread.
+ std::string JavaToStdString(const jstring& j_string);
+
+ private:
+ rtc::ThreadChecker thread_checker_;
+ JNIEnv* const jni_;
+};
+
+// Main class for working with Java from C++ using JNI in WebRTC.
+//
+// Example usage:
+//
+// // At initialization (e.g. in JNI_OnLoad), call JVM::Initialize.
+// JNIEnv* jni = ::base::android::AttachCurrentThread();
+// JavaVM* jvm = NULL;
+// jni->GetJavaVM(&jvm);
+// jobject context = ::base::android::GetApplicationContext();
+// webrtc::JVM::Initialize(jvm, context);
+//
+// // Header (.h) file of example class called User.
+// rtc::scoped_ptr<JNIEnvironment> env;
+// rtc::scoped_ptr<NativeRegistration> reg;
+// rtc::scoped_ptr<GlobalRef> obj;
+//
+// // Construction (in .cc file) of User class.
+// User::User() {
+// // Calling thread must be attached to the JVM.
+// env = JVM::GetInstance()->environment();
+// reg = env->RegisterNatives("org/webrtc/WebRtcTest", ,);
+// obj = reg->NewObject("<init>", ,);
+// }
+//
+// // Each User method can now use |reg| and |obj| and call Java functions
+// // in WebRtcTest.java, e.g. boolean init() {}.
+// bool User::Foo() {
+// jmethodID id = reg->GetMethodId("init", "()Z");
+// return obj->CallBooleanMethod(id);
+// }
+//
+// // And finally, e.g. in JNI_OnUnLoad, call JVM::Uninitialize.
+// JVM::Uninitialize();
+class JVM {
+ public:
+ // Stores global handles to the Java VM interface and the application context.
+ // Should be called once on a thread that is attached to the JVM.
+ static void Initialize(JavaVM* jvm, jobject context);
+ // Clears handles stored in Initialize(). Must be called on same thread as
+ // Initialize().
+ static void Uninitialize();
+ // Gives access to the global Java VM interface pointer, which then can be
+ // used to create a valid JNIEnvironment object or to get a JavaClass object.
+ static JVM* GetInstance();
+
+ // Creates a JNIEnvironment object.
+ // This method returns a NULL pointer if AttachCurrentThread() has not been
+ // called successfully. Use the AttachCurrentThreadIfNeeded class if needed.
+ rtc::scoped_ptr<JNIEnvironment> environment();
+
+ // Returns a JavaClass object given class |name|.
+ // Note that the class name must be one of the names in the static
+ // |loaded_classes| array defined in jvm_android.cc.
+ // This method must be called on the construction thread.
+ JavaClass GetClass(const char* name);
+
+ // TODO(henrika): can we make these private?
+ JavaVM* jvm() const { return jvm_; }
+ jobject context() const { return context_; }
+
+ protected:
+ JVM(JavaVM* jvm, jobject context);
+ ~JVM();
+
+ private:
+ JNIEnv* jni() const { return GetEnv(jvm_); }
+
+ rtc::ThreadChecker thread_checker_;
+ JavaVM* const jvm_;
+ jobject context_;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
diff --git a/webrtc/modules/utility/include/mock/mock_process_thread.h b/webrtc/modules/utility/include/mock/mock_process_thread.h
new file mode 100644
index 0000000..56d92f4
--- /dev/null
+++ b/webrtc/modules/utility/include/mock/mock_process_thread.h
@@ -0,0 +1,38 @@
+/*
+ * 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_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
+
+#include "webrtc/modules/utility/include/process_thread.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+
+namespace webrtc {
+
+class MockProcessThread : public ProcessThread {
+ public:
+ MOCK_METHOD0(Start, void());
+ MOCK_METHOD0(Stop, void());
+ MOCK_METHOD1(WakeUp, void(Module* module));
+ MOCK_METHOD1(PostTask, void(ProcessTask* task));
+ MOCK_METHOD1(RegisterModule, void(Module* module));
+ MOCK_METHOD1(DeRegisterModule, void(Module* module));
+
+ // MOCK_METHOD1 gets confused with mocking this method, so we work around it
+ // by overriding the method from the interface and forwarding the call to a
+ // mocked, simpler method.
+ void PostTask(rtc::scoped_ptr<ProcessTask> task) override {
+ PostTask(task.get());
+ }
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/include/process_thread.h b/webrtc/modules/utility/include/process_thread.h
new file mode 100644
index 0000000..285a5ea
--- /dev/null
+++ b/webrtc/modules/utility/include/process_thread.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (c) 2011 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_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
+
+#include "webrtc/typedefs.h"
+#include "webrtc/base/scoped_ptr.h"
+
+namespace webrtc {
+class Module;
+
+class ProcessTask {
+ public:
+ ProcessTask() {}
+ virtual ~ProcessTask() {}
+
+ virtual void Run() = 0;
+};
+
+class ProcessThread {
+ public:
+ virtual ~ProcessThread();
+
+ static rtc::scoped_ptr<ProcessThread> Create(const char* thread_name);
+
+ // Starts the worker thread. Must be called from the construction thread.
+ virtual void Start() = 0;
+
+ // Stops the worker thread. Must be called from the construction thread.
+ virtual void Stop() = 0;
+
+ // Wakes the thread up to give a module a chance to do processing right
+ // away. This causes the worker thread to wake up and requery the specified
+ // module for when it should be called back. (Typically the module should
+ // return 0 from TimeUntilNextProcess on the worker thread at that point).
+ // Can be called on any thread.
+ virtual void WakeUp(Module* module) = 0;
+
+ // Queues a task object to run on the worker thread. Ownership of the
+ // task object is transferred to the ProcessThread and the object will
+ // either be deleted after running on the worker thread, or on the
+ // construction thread of the ProcessThread instance, if the task did not
+ // get a chance to run (e.g. posting the task while shutting down or when
+ // the thread never runs).
+ virtual void PostTask(rtc::scoped_ptr<ProcessTask> task) = 0;
+
+ // Adds a module that will start to receive callbacks on the worker thread.
+ // Can be called from any thread.
+ virtual void RegisterModule(Module* module) = 0;
+
+ // Removes a previously registered module.
+ // Can be called from any thread.
+ virtual void DeRegisterModule(Module* module) = 0;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/interface/audio_frame_operations.h b/webrtc/modules/utility/interface/audio_frame_operations.h
index c2af68a..017352a 100644
--- a/webrtc/modules/utility/interface/audio_frame_operations.h
+++ b/webrtc/modules/utility/interface/audio_frame_operations.h
@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
-#define WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
#include "webrtc/typedefs.h"
@@ -55,4 +57,4 @@
} // namespace webrtc
-#endif // #ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H_
+#endif // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
diff --git a/webrtc/modules/utility/interface/file_player.h b/webrtc/modules/utility/interface/file_player.h
index 44f03e4..d6737e1 100644
--- a/webrtc/modules/utility/interface/file_player.h
+++ b/webrtc/modules/utility/interface/file_player.h
@@ -8,12 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_frame.h"
@@ -108,4 +110,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
diff --git a/webrtc/modules/utility/interface/file_recorder.h b/webrtc/modules/utility/interface/file_recorder.h
index f2ce785..c7e26f6 100644
--- a/webrtc/modules/utility/interface/file_recorder.h
+++ b/webrtc/modules/utility/interface/file_recorder.h
@@ -8,13 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_frame.h"
@@ -81,4 +83,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
diff --git a/webrtc/modules/utility/interface/helpers_android.h b/webrtc/modules/utility/interface/helpers_android.h
index 5c73fe4..80dd676 100644
--- a/webrtc/modules/utility/interface/helpers_android.h
+++ b/webrtc/modules/utility/interface/helpers_android.h
@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
#include <jni.h>
#include <string>
@@ -84,4 +86,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_ANDROID_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
diff --git a/webrtc/modules/utility/interface/helpers_ios.h b/webrtc/modules/utility/interface/helpers_ios.h
index a5edee0..6013ee9 100644
--- a/webrtc/modules/utility/interface/helpers_ios.h
+++ b/webrtc/modules/utility/interface/helpers_ios.h
@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
#if defined(WEBRTC_IOS)
@@ -56,4 +58,4 @@
#endif // defined(WEBRTC_IOS)
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_HELPERS_IOS_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
diff --git a/webrtc/modules/utility/interface/jvm_android.h b/webrtc/modules/utility/interface/jvm_android.h
index 0744fdb..a417c1b 100644
--- a/webrtc/modules/utility/interface/jvm_android.h
+++ b/webrtc/modules/utility/interface/jvm_android.h
@@ -8,15 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
#include <jni.h>
#include <string>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
namespace webrtc {
@@ -182,4 +184,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_JVM_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
diff --git a/webrtc/modules/utility/interface/mock/mock_process_thread.h b/webrtc/modules/utility/interface/mock/mock_process_thread.h
index fd108a8..c494d4c 100644
--- a/webrtc/modules/utility/interface/mock/mock_process_thread.h
+++ b/webrtc/modules/utility/interface/mock/mock_process_thread.h
@@ -8,10 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
-#include "webrtc/modules/utility/interface/process_thread.h"
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
+
+#include "webrtc/modules/utility/include/process_thread.h"
#include "testing/gmock/include/gmock/gmock.h"
@@ -35,4 +37,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_MOCK_PROCESS_THREAD_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/interface/process_thread.h b/webrtc/modules/utility/interface/process_thread.h
index 451a5a3..1a7a875 100644
--- a/webrtc/modules/utility/interface/process_thread.h
+++ b/webrtc/modules/utility/interface/process_thread.h
@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
+#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
+#define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
+
+#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
#include "webrtc/typedefs.h"
#include "webrtc/base/scoped_ptr.h"
@@ -63,4 +65,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
+#endif // WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/source/audio_frame_operations.cc b/webrtc/modules/utility/source/audio_frame_operations.cc
index c07ca1f..fe09d79 100644
--- a/webrtc/modules/utility/source/audio_frame_operations.cc
+++ b/webrtc/modules/utility/source/audio_frame_operations.cc
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
namespace webrtc {
diff --git a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
index c278cdd..fff8f44 100644
--- a/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
+++ b/webrtc/modules/utility/source/audio_frame_operations_unittest.cc
@@ -10,8 +10,8 @@
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
namespace webrtc {
namespace {
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index 4ec5f9b..18b690d 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -9,7 +9,7 @@
*/
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/utility/source/coder.h"
namespace webrtc {
diff --git a/webrtc/modules/utility/source/file_player_impl.h b/webrtc/modules/utility/source/file_player_impl.h
index f411db9..2dfe682 100644
--- a/webrtc/modules/utility/source/file_player_impl.h
+++ b/webrtc/modules/utility/source/file_player_impl.h
@@ -14,9 +14,9 @@
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/utility/interface/file_player.h"
+#include "webrtc/modules/media_file/include/media_file.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_player.h"
#include "webrtc/modules/utility/source/coder.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/modules/utility/source/file_player_unittests.cc b/webrtc/modules/utility/source/file_player_unittests.cc
index 4b65acd..754e124 100644
--- a/webrtc/modules/utility/source/file_player_unittests.cc
+++ b/webrtc/modules/utility/source/file_player_unittests.cc
@@ -10,7 +10,7 @@
// Unit tests for FilePlayer.
-#include "webrtc/modules/utility/interface/file_player.h"
+#include "webrtc/modules/utility/include/file_player.h"
#include <stdio.h>
#include <string>
diff --git a/webrtc/modules/utility/source/file_recorder_impl.cc b/webrtc/modules/utility/source/file_recorder_impl.cc
index 13926de..cbd7c08 100644
--- a/webrtc/modules/utility/source/file_recorder_impl.cc
+++ b/webrtc/modules/utility/source/file_recorder_impl.cc
@@ -10,7 +10,7 @@
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
+#include "webrtc/modules/media_file/include/media_file.h"
#include "webrtc/modules/utility/source/file_recorder_impl.h"
#include "webrtc/system_wrappers/include/logging.h"
diff --git a/webrtc/modules/utility/source/file_recorder_impl.h b/webrtc/modules/utility/source/file_recorder_impl.h
index 8ea96bd..01072cd 100644
--- a/webrtc/modules/utility/source/file_recorder_impl.h
+++ b/webrtc/modules/utility/source/file_recorder_impl.h
@@ -20,10 +20,10 @@
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
-#include "webrtc/modules/media_file/interface/media_file_defines.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/media_file/include/media_file.h"
+#include "webrtc/modules/media_file/include/media_file_defines.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/modules/utility/source/coder.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/thread_wrapper.h"
diff --git a/webrtc/modules/utility/source/helpers_android.cc b/webrtc/modules/utility/source/helpers_android.cc
index 25652f2..aea35f8 100644
--- a/webrtc/modules/utility/source/helpers_android.cc
+++ b/webrtc/modules/utility/source/helpers_android.cc
@@ -9,7 +9,7 @@
*/
#include "webrtc/base/checks.h"
-#include "webrtc/modules/utility/interface/helpers_android.h"
+#include "webrtc/modules/utility/include/helpers_android.h"
#include <android/log.h>
#include <assert.h>
diff --git a/webrtc/modules/utility/source/helpers_ios.mm b/webrtc/modules/utility/source/helpers_ios.mm
index 90b7c8f..2d0ac09 100644
--- a/webrtc/modules/utility/source/helpers_ios.mm
+++ b/webrtc/modules/utility/source/helpers_ios.mm
@@ -18,7 +18,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/utility/interface/helpers_ios.h"
+#include "webrtc/modules/utility/include/helpers_ios.h"
namespace webrtc {
namespace ios {
diff --git a/webrtc/modules/utility/source/jvm_android.cc b/webrtc/modules/utility/source/jvm_android.cc
index 648c168..eb37fda 100644
--- a/webrtc/modules/utility/source/jvm_android.cc
+++ b/webrtc/modules/utility/source/jvm_android.cc
@@ -10,7 +10,7 @@
#include <android/log.h>
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
#include "webrtc/base/checks.h"
diff --git a/webrtc/modules/utility/source/process_thread_impl.cc b/webrtc/modules/utility/source/process_thread_impl.cc
index 04fa887..9c739de 100644
--- a/webrtc/modules/utility/source/process_thread_impl.cc
+++ b/webrtc/modules/utility/source/process_thread_impl.cc
@@ -11,7 +11,7 @@
#include "webrtc/modules/utility/source/process_thread_impl.h"
#include "webrtc/base/checks.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/modules/utility/source/process_thread_impl.h b/webrtc/modules/utility/source/process_thread_impl.h
index 4e5861b..0a95665 100644
--- a/webrtc/modules/utility/source/process_thread_impl.h
+++ b/webrtc/modules/utility/source/process_thread_impl.h
@@ -16,7 +16,7 @@
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/thread_wrapper.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/utility/source/process_thread_impl_unittest.cc b/webrtc/modules/utility/source/process_thread_impl_unittest.cc
index e080545..34ed5c5 100644
--- a/webrtc/modules/utility/source/process_thread_impl_unittest.cc
+++ b/webrtc/modules/utility/source/process_thread_impl_unittest.cc
@@ -10,7 +10,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
#include "webrtc/modules/utility/source/process_thread_impl.h"
#include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/modules/utility/utility.gypi b/webrtc/modules/utility/utility.gypi
index 38c9e3e..e5b0a4d 100644
--- a/webrtc/modules/utility/utility.gypi
+++ b/webrtc/modules/utility/utility.gypi
@@ -18,13 +18,13 @@
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
- 'interface/audio_frame_operations.h',
- 'interface/file_player.h',
- 'interface/file_recorder.h',
- 'interface/helpers_android.h',
- 'interface/helpers_ios.h',
- 'interface/jvm_android.h',
- 'interface/process_thread.h',
+ 'include/audio_frame_operations.h',
+ 'include/file_player.h',
+ 'include/file_recorder.h',
+ 'include/helpers_android.h',
+ 'include/helpers_ios.h',
+ 'include/jvm_android.h',
+ 'include/process_thread.h',
'source/audio_frame_operations.cc',
'source/coder.cc',
'source/coder.h',
diff --git a/webrtc/modules/video_capture/include/video_capture.h b/webrtc/modules/video_capture/include/video_capture.h
index 09b4502..a032d3c 100644
--- a/webrtc/modules/video_capture/include/video_capture.h
+++ b/webrtc/modules/video_capture/include/video_capture.h
@@ -12,7 +12,7 @@
#define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_H_
#include "webrtc/common_video/rotation.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
#include "webrtc/modules/video_capture/include/video_capture_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/video_capture/include/video_capture_defines.h b/webrtc/modules/video_capture/include/video_capture_defines.h
index 1dee4fa..f62ddad 100644
--- a/webrtc/modules/video_capture/include/video_capture_defines.h
+++ b/webrtc/modules/video_capture/include/video_capture_defines.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_
#define WEBRTC_MODULES_VIDEO_CAPTURE_INCLUDE_VIDEO_CAPTURE_DEFINES_H_
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_frame.h"
diff --git a/webrtc/modules/video_capture/test/video_capture_unittest.cc b/webrtc/modules/video_capture/test/video_capture_unittest.cc
index 2b8786b..3ec164b 100644
--- a/webrtc/modules/video_capture/test/video_capture_unittest.cc
+++ b/webrtc/modules/video_capture/test/video_capture_unittest.cc
@@ -17,7 +17,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/modules/video_capture/include/video_capture.h"
#include "webrtc/modules/video_capture/include/video_capture_factory.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/video_capture/video_capture_impl.cc b/webrtc/modules/video_capture/video_capture_impl.cc
index 4046181..90730cd 100644
--- a/webrtc/modules/video_capture/video_capture_impl.cc
+++ b/webrtc/modules/video_capture/video_capture_impl.cc
@@ -14,7 +14,7 @@
#include "webrtc/base/trace_event.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_capture/video_capture_config.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
index 230dea9..bdb079b 100644
--- a/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
+++ b/webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h
@@ -19,7 +19,7 @@
#include <CoreMedia/CoreMedia.h>
#include "webrtc/base/buffer.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h b/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
index 6363ab7..5115c4b 100644
--- a/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
+++ b/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
@@ -14,7 +14,7 @@
#include <vector>
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/codecs/interface/video_error_codes.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_decoder.h"
diff --git a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
index da6008b..47f0cf3 100644
--- a/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
@@ -13,7 +13,7 @@
#include <stdlib.h>
#include <string.h>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
diff --git a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
index 029ccd1..4f4e435 100644
--- a/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
+++ b/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
@@ -24,7 +24,7 @@
#include "webrtc/common.h"
#include "webrtc/common_types.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
#include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
diff --git a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
index 97f0846..7ebe1a2 100644
--- a/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
+++ b/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
@@ -26,7 +26,7 @@
#include "webrtc/base/trace_event.h"
#include "webrtc/common.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/modules/video_coding/main/interface/video_coding.h b/webrtc/modules/video_coding/main/interface/video_coding.h
index 67f7b63..94a7b19 100644
--- a/webrtc/modules/video_coding/main/interface/video_coding.h
+++ b/webrtc/modules/video_coding/main/interface/video_coding.h
@@ -21,8 +21,8 @@
#include <windows.h>
#endif
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/video_frame.h"
diff --git a/webrtc/modules/video_coding/main/interface/video_coding_defines.h b/webrtc/modules/video_coding/main/interface/video_coding_defines.h
index fd38d64..c35bbc1 100644
--- a/webrtc/modules/video_coding/main/interface/video_coding_defines.h
+++ b/webrtc/modules/video_coding/main/interface/video_coding_defines.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_frame.h"
diff --git a/webrtc/modules/video_coding/main/source/codec_timer.h b/webrtc/modules/video_coding/main/source/codec_timer.h
index 9268e8d..cb7e813 100644
--- a/webrtc/modules/video_coding/main/source/codec_timer.h
+++ b/webrtc/modules/video_coding/main/source/codec_timer.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc
diff --git a/webrtc/modules/video_coding/main/source/content_metrics_processing.cc b/webrtc/modules/video_coding/main/source/content_metrics_processing.cc
index 757ffb0..ae5e633 100644
--- a/webrtc/modules/video_coding/main/source/content_metrics_processing.cc
+++ b/webrtc/modules/video_coding/main/source/content_metrics_processing.cc
@@ -12,7 +12,7 @@
#include <math.h>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/video_coding/main/source/decoding_state.cc b/webrtc/modules/video_coding/main/source/decoding_state.cc
index cc92f1c..a3da7c6 100644
--- a/webrtc/modules/video_coding/main/source/decoding_state.cc
+++ b/webrtc/modules/video_coding/main/source/decoding_state.cc
@@ -10,7 +10,7 @@
#include "webrtc/modules/video_coding/main/source/decoding_state.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
#include "webrtc/modules/video_coding/main/source/packet.h"
diff --git a/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc b/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
index feae701..30b5786 100644
--- a/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/decoding_state_unittest.cc
@@ -11,7 +11,7 @@
#include <string.h>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/source/decoding_state.h"
#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
diff --git a/webrtc/modules/video_coding/main/source/encoded_frame.h b/webrtc/modules/video_coding/main/source/encoded_frame.h
index 608578c..dc6bbb8 100644
--- a/webrtc/modules/video_coding/main/source/encoded_frame.h
+++ b/webrtc/modules/video_coding/main/source/encoded_frame.h
@@ -15,7 +15,7 @@
#include "webrtc/common_types.h"
#include "webrtc/common_video/interface/video_image.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
diff --git a/webrtc/modules/video_coding/main/source/frame_buffer.h b/webrtc/modules/video_coding/main/source/frame_buffer.h
index ab4ff65..ee38a2e 100644
--- a/webrtc/modules/video_coding/main/source/frame_buffer.h
+++ b/webrtc/modules/video_coding/main/source/frame_buffer.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_FRAME_BUFFER_H_
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
diff --git a/webrtc/modules/video_coding/main/source/generic_decoder.h b/webrtc/modules/video_coding/main/source/generic_decoder.h
index 09929e6..c1298bb 100644
--- a/webrtc/modules/video_coding/main/source/generic_decoder.h
+++ b/webrtc/modules/video_coding/main/source/generic_decoder.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
#include "webrtc/modules/video_coding/main/source/encoded_frame.h"
#include "webrtc/modules/video_coding/main/source/timestamp_map.h"
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
index bfdd786..bc63411 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
@@ -16,7 +16,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/source/frame_buffer.h"
#include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.h b/webrtc/modules/video_coding/main/source/jitter_buffer.h
index f4a3638..9bde97c 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.h
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.h
@@ -18,7 +18,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/thread_annotations.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
#include "webrtc/modules/video_coding/main/source/decoding_state.h"
diff --git a/webrtc/modules/video_coding/main/source/media_opt_util.cc b/webrtc/modules/video_coding/main/source/media_opt_util.cc
index 51decbe..002958e 100644
--- a/webrtc/modules/video_coding/main/source/media_opt_util.cc
+++ b/webrtc/modules/video_coding/main/source/media_opt_util.cc
@@ -15,7 +15,7 @@
#include <limits.h>
#include <math.h>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
#include "webrtc/modules/video_coding/main/source/fec_tables_xor.h"
diff --git a/webrtc/modules/video_coding/main/source/media_optimization.h b/webrtc/modules/video_coding/main/source/media_optimization.h
index c4feeff..6ea5c01 100644
--- a/webrtc/modules/video_coding/main/source/media_optimization.h
+++ b/webrtc/modules/video_coding/main/source/media_optimization.h
@@ -14,7 +14,7 @@
#include <list>
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/source/media_opt_util.h"
#include "webrtc/modules/video_coding/main/source/qm_select.h"
diff --git a/webrtc/modules/video_coding/main/source/packet.cc b/webrtc/modules/video_coding/main/source/packet.cc
index fd5a6ab..3438448 100644
--- a/webrtc/modules/video_coding/main/source/packet.cc
+++ b/webrtc/modules/video_coding/main/source/packet.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/source/packet.h"
#include <assert.h>
diff --git a/webrtc/modules/video_coding/main/source/packet.h b/webrtc/modules/video_coding/main/source/packet.h
index 80bf532..040cc88 100644
--- a/webrtc/modules/video_coding/main/source/packet.h
+++ b/webrtc/modules/video_coding/main/source/packet.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
#define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/video_coding/main/source/qm_select.cc b/webrtc/modules/video_coding/main/source/qm_select.cc
index e86d075..be8fcfc 100644
--- a/webrtc/modules/video_coding/main/source/qm_select.cc
+++ b/webrtc/modules/video_coding/main/source/qm_select.cc
@@ -12,7 +12,7 @@
#include <math.h>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
#include "webrtc/modules/video_coding/main/source/internal_defines.h"
#include "webrtc/modules/video_coding/main/source/qm_select_data.h"
diff --git a/webrtc/modules/video_coding/main/source/qm_select_unittest.cc b/webrtc/modules/video_coding/main/source/qm_select_unittest.cc
index 6abc0d3..518df34 100644
--- a/webrtc/modules/video_coding/main/source/qm_select_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/qm_select_unittest.cc
@@ -15,7 +15,7 @@
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/source/qm_select.h"
namespace webrtc {
diff --git a/webrtc/modules/video_coding/main/source/session_info.h b/webrtc/modules/video_coding/main/source/session_info.h
index 88071e1..37e9768 100644
--- a/webrtc/modules/video_coding/main/source/session_info.h
+++ b/webrtc/modules/video_coding/main/source/session_info.h
@@ -13,7 +13,7 @@
#include <list>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/source/packet.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/modules/video_coding/main/source/session_info_unittest.cc b/webrtc/modules/video_coding/main/source/session_info_unittest.cc
index 58c352d..8d57287 100644
--- a/webrtc/modules/video_coding/main/source/session_info_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/session_info_unittest.cc
@@ -11,7 +11,7 @@
#include <string.h>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/source/packet.h"
#include "webrtc/modules/video_coding/main/source/session_info.h"
diff --git a/webrtc/modules/video_coding/main/source/timestamp_map.cc b/webrtc/modules/video_coding/main/source/timestamp_map.cc
index c68a5af..d11f949 100644
--- a/webrtc/modules/video_coding/main/source/timestamp_map.cc
+++ b/webrtc/modules/video_coding/main/source/timestamp_map.cc
@@ -11,7 +11,7 @@
#include <assert.h>
#include <stdlib.h>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/source/timestamp_map.h"
namespace webrtc {
diff --git a/webrtc/modules/video_coding/main/test/receiver_tests.h b/webrtc/modules/video_coding/main/test/receiver_tests.h
index 6d7b7be..5335c99 100644
--- a/webrtc/modules/video_coding/main/test/receiver_tests.h
+++ b/webrtc/modules/video_coding/main/test/receiver_tests.h
@@ -12,8 +12,8 @@
#define WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/test/test_util.h"
#include "webrtc/modules/video_coding/main/test/video_source.h"
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.cc b/webrtc/modules/video_coding/main/test/rtp_player.cc
index 6717cf2..5fed3b1 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.cc
+++ b/webrtc/modules/video_coding/main/test/rtp_player.cc
@@ -15,10 +15,10 @@
#include <map>
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/video_coding/main/source/internal_defines.h"
#include "webrtc/modules/video_coding/main/test/test_util.h"
#include "webrtc/system_wrappers/include/clock.h"
diff --git a/webrtc/modules/video_coding/main/test/rtp_player.h b/webrtc/modules/video_coding/main/test/rtp_player.h
index 7459231..a2ecadd 100644
--- a/webrtc/modules/video_coding/main/test/rtp_player.h
+++ b/webrtc/modules/video_coding/main/test/rtp_player.h
@@ -14,7 +14,7 @@
#include <string>
#include <vector>
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/video_coding/main/test/test_util.h b/webrtc/modules/video_coding/main/test/test_util.h
index 27f66fe..68b2c23 100644
--- a/webrtc/modules/video_coding/main/test/test_util.h
+++ b/webrtc/modules/video_coding/main/test/test_util.h
@@ -18,7 +18,7 @@
#include <string>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
diff --git a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc b/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
index 2d874cd..d930805 100644
--- a/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
+++ b/webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
@@ -14,7 +14,7 @@
#include <algorithm>
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/video_coding/main/test/test_util.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/video_processing/main/interface/video_processing.h b/webrtc/modules/video_processing/main/interface/video_processing.h
index 30af99f..377a098 100644
--- a/webrtc/modules/video_processing/main/interface/video_processing.h
+++ b/webrtc/modules/video_processing/main/interface/video_processing.h
@@ -18,8 +18,8 @@
#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
#define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
-#include "webrtc/modules/interface/module.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
#include "webrtc/video_frame.h"
diff --git a/webrtc/modules/video_processing/main/source/content_analysis.h b/webrtc/modules/video_processing/main/source/content_analysis.h
index 510c1b4..5b0767a 100644
--- a/webrtc/modules/video_processing/main/source/content_analysis.h
+++ b/webrtc/modules/video_processing/main/source/content_analysis.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_CONTENT_ANALYSIS_H
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_frame.h"
diff --git a/webrtc/modules/video_processing/main/source/spatial_resampler.h b/webrtc/modules/video_processing/main/source/spatial_resampler.h
index f965a40..9a2d8f5 100644
--- a/webrtc/modules/video_processing/main/source/spatial_resampler.h
+++ b/webrtc/modules/video_processing/main/source/spatial_resampler.h
@@ -13,7 +13,7 @@
#include "webrtc/typedefs.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_processing/main/interface/video_processing_defines.h"
#include "webrtc/common_video/libyuv/include/scaler.h"
diff --git a/webrtc/modules/video_processing/main/source/video_decimator.h b/webrtc/modules/video_processing/main/source/video_decimator.h
index 3d4573c..c052c30 100644
--- a/webrtc/modules/video_processing/main/source/video_decimator.h
+++ b/webrtc/modules/video_processing/main/source/video_decimator.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H
#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_SOURCE_VIDEO_DECIMATOR_H
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
diff --git a/webrtc/modules/video_render/external/video_render_external_impl.h b/webrtc/modules/video_render/external/video_render_external_impl.h
index 9230e60..a8b663f 100644
--- a/webrtc/modules/video_render/external/video_render_external_impl.h
+++ b/webrtc/modules/video_render/external/video_render_external_impl.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/video_render/i_video_render.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
diff --git a/webrtc/modules/video_render/include/video_render.h b/webrtc/modules/video_render/include/video_render.h
index 51fcce1..edd2302 100644
--- a/webrtc/modules/video_render/include/video_render.h
+++ b/webrtc/modules/video_render/include/video_render.h
@@ -20,7 +20,7 @@
*
*/
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
#include "webrtc/modules/video_render/include/video_render_defines.h"
namespace webrtc {
diff --git a/webrtc/modules/video_render/include/video_render_defines.h b/webrtc/modules/video_render/include/video_render_defines.h
index f8f4803..1602d66 100644
--- a/webrtc/modules/video_render/include/video_render_defines.h
+++ b/webrtc/modules/video_render/include/video_render_defines.h
@@ -13,7 +13,7 @@
#include "webrtc/common_types.h"
#include "webrtc/common_video/interface/incoming_video_stream.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc
{
diff --git a/webrtc/modules/video_render/test/testAPI/testAPI.cc b/webrtc/modules/video_render/test/testAPI/testAPI.cc
index 256d031..4de554c 100644
--- a/webrtc/modules/video_render/test/testAPI/testAPI.cc
+++ b/webrtc/modules/video_render/test/testAPI/testAPI.cc
@@ -32,8 +32,8 @@
#endif
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/modules/video_render/include/video_render.h"
#include "webrtc/modules/video_render/include/video_render_defines.h"
#include "webrtc/system_wrappers/include/sleep.h"
diff --git a/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm b/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
index dd57397..a8808dd 100644
--- a/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
+++ b/webrtc/modules/video_render/test/testAPI/testAPI_mac.mm
@@ -20,8 +20,8 @@
#import "webrtc/modules/video_render/mac/cocoa_render_view.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/modules/video_render/include/video_render.h"
#include "webrtc/modules/video_render/include/video_render_defines.h"
#include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/test/layer_filtering_transport.cc b/webrtc/test/layer_filtering_transport.cc
index 5533a4c..9cf02ed 100644
--- a/webrtc/test/layer_filtering_transport.cc
+++ b/webrtc/test/layer_filtering_transport.cc
@@ -9,7 +9,8 @@
*/
#include "webrtc/base/checks.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.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/rtp_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/test/layer_filtering_transport.h"
diff --git a/webrtc/test/rtp_rtcp_observer.h b/webrtc/test/rtp_rtcp_observer.h
index 89b6dd0..c7e463e 100644
--- a/webrtc/test/rtp_rtcp_observer.h
+++ b/webrtc/test/rtp_rtcp_observer.h
@@ -16,7 +16,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/criticalsection.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/test/constants.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/tools/agc/activity_metric.cc b/webrtc/tools/agc/activity_metric.cc
index 18e7c6d..3f8235e 100644
--- a/webrtc/tools/agc/activity_metric.cc
+++ b/webrtc/tools/agc/activity_metric.cc
@@ -24,7 +24,7 @@
#include "webrtc/modules/audio_processing/vad/common.h"
#include "webrtc/modules/audio_processing/vad/pitch_based_vad.h"
#include "webrtc/modules/audio_processing/vad/standalone_vad.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
static const int kAgcAnalWindowSamples = 100;
static const double kDefaultActivityThreshold = 0.3;
diff --git a/webrtc/tools/agc/test_utils.cc b/webrtc/tools/agc/test_utils.cc
index 81819c5..a0ed747 100644
--- a/webrtc/tools/agc/test_utils.cc
+++ b/webrtc/tools/agc/test_utils.cc
@@ -14,7 +14,7 @@
#include <algorithm>
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc
index 70efe3b..e2dce3f 100644
--- a/webrtc/video/rampup_tests.cc
+++ b/webrtc/video/rampup_tests.cc
@@ -16,10 +16,10 @@
#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/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/thread_wrapper.h"
diff --git a/webrtc/video/replay.cc b/webrtc/video/replay.cc
index f54909e..4849248 100644
--- a/webrtc/video/replay.cc
+++ b/webrtc/video/replay.cc
@@ -20,7 +20,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/call.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/encoder_settings.h"
diff --git a/webrtc/video/video_capture_input.cc b/webrtc/video/video_capture_input.cc
index 42bc65f..834849c 100644
--- a/webrtc/video/video_capture_input.cc
+++ b/webrtc/video/video_capture_input.cc
@@ -13,8 +13,8 @@
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/modules/video_capture/include/video_capture_factory.h"
#include "webrtc/modules/video_processing/main/interface/video_processing.h"
#include "webrtc/modules/video_render/include/video_render_defines.h"
diff --git a/webrtc/video/video_capture_input_unittest.cc b/webrtc/video/video_capture_input_unittest.cc
index e8bc2ad..12ef076 100644
--- a/webrtc/video/video_capture_input_unittest.cc
+++ b/webrtc/video/video_capture_input_unittest.cc
@@ -15,7 +15,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/ref_count.h"
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index 333f00d..ebceb56 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -22,6 +22,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/call.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/test/layer_filtering_transport.h"
diff --git a/webrtc/video/video_send_stream.h b/webrtc/video/video_send_stream.h
index 36a87e3..6b0d302 100644
--- a/webrtc/video/video_send_stream.h
+++ b/webrtc/video/video_send_stream.h
@@ -17,7 +17,7 @@
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/video/encoded_frame_callback_adapter.h"
#include "webrtc/video/send_statistics_proxy.h"
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 59011a6..e19dc48 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -20,8 +20,8 @@
#include "webrtc/call.h"
#include "webrtc/call/transport_adapter.h"
#include "webrtc/frame_callback.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
diff --git a/webrtc/video_engine/call_stats.cc b/webrtc/video_engine/call_stats.cc
index 0b71cc3..4d5338c 100644
--- a/webrtc/video_engine/call_stats.cc
+++ b/webrtc/video_engine/call_stats.cc
@@ -12,7 +12,7 @@
#include <assert.h>
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/video_engine/call_stats.h b/webrtc/video_engine/call_stats.h
index a17330a..d0a0b53 100644
--- a/webrtc/video_engine/call_stats.h
+++ b/webrtc/video_engine/call_stats.h
@@ -15,7 +15,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
namespace webrtc {
diff --git a/webrtc/video_engine/call_stats_unittest.cc b/webrtc/video_engine/call_stats_unittest.cc
index 4fb88df..bfba5cb 100644
--- a/webrtc/video_engine/call_stats_unittest.cc
+++ b/webrtc/video_engine/call_stats_unittest.cc
@@ -12,7 +12,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/video_engine/call_stats.h"
diff --git a/webrtc/video_engine/encoder_state_feedback.cc b/webrtc/video_engine/encoder_state_feedback.cc
index 1c376b2..16ee8b9 100644
--- a/webrtc/video_engine/encoder_state_feedback.cc
+++ b/webrtc/video_engine/encoder_state_feedback.cc
@@ -13,7 +13,7 @@
#include <assert.h>
#include "webrtc/base/checks.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/video_engine/vie_encoder.h"
diff --git a/webrtc/video_engine/encoder_state_feedback_unittest.cc b/webrtc/video_engine/encoder_state_feedback_unittest.cc
index 9787acc..ea252dc 100644
--- a/webrtc/video_engine/encoder_state_feedback_unittest.cc
+++ b/webrtc/video_engine/encoder_state_feedback_unittest.cc
@@ -20,8 +20,8 @@
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/pacing/include/paced_sender.h"
#include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
#include "webrtc/video_engine/payload_router.h"
#include "webrtc/video_engine/vie_encoder.h"
diff --git a/webrtc/video_engine/overuse_frame_detector.h b/webrtc/video_engine/overuse_frame_detector.h
index aff4b43..0deb5ba 100644
--- a/webrtc/video_engine/overuse_frame_detector.h
+++ b/webrtc/video_engine/overuse_frame_detector.h
@@ -17,7 +17,7 @@
#include "webrtc/base/exp_filter.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
namespace webrtc {
diff --git a/webrtc/video_engine/payload_router.cc b/webrtc/video_engine/payload_router.cc
index 3af3d48..85b294b 100644
--- a/webrtc/video_engine/payload_router.cc
+++ b/webrtc/video_engine/payload_router.cc
@@ -11,8 +11,8 @@
#include "webrtc/video_engine/payload_router.h"
#include "webrtc/base/checks.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
diff --git a/webrtc/video_engine/payload_router_unittest.cc b/webrtc/video_engine/payload_router_unittest.cc
index de39157..acaa400 100644
--- a/webrtc/video_engine/payload_router_unittest.cc
+++ b/webrtc/video_engine/payload_router_unittest.cc
@@ -14,7 +14,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/video_engine/payload_router.h"
diff --git a/webrtc/video_engine/report_block_stats.h b/webrtc/video_engine/report_block_stats.h
index dadcc9d..dd43072 100644
--- a/webrtc/video_engine/report_block_stats.h
+++ b/webrtc/video_engine/report_block_stats.h
@@ -15,7 +15,7 @@
#include <vector>
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index 9140cf1..b4e8332 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -21,9 +21,9 @@
#include "webrtc/frame_callback.h"
#include "webrtc/modules/pacing/include/paced_sender.h"
#include "webrtc/modules/pacing/include/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.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_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_processing/main/interface/video_processing.h"
#include "webrtc/modules/video_render/include/video_render_defines.h"
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index b1c1bba..20eae7d 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -16,8 +16,8 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/tick_util.h"
diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc
index 0f4a5a1..96ed2f9 100644
--- a/webrtc/video_engine/vie_encoder.cc
+++ b/webrtc/video_engine/vie_encoder.cc
@@ -22,7 +22,7 @@
#include "webrtc/frame_callback.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/pacing/include/paced_sender.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
diff --git a/webrtc/video_engine/vie_encoder.h b/webrtc/video_engine/vie_encoder.h
index 54aacdb..872fbb6 100644
--- a/webrtc/video_engine/vie_encoder.h
+++ b/webrtc/video_engine/vie_encoder.h
@@ -20,7 +20,7 @@
#include "webrtc/common_types.h"
#include "webrtc/frame_callback.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_allocator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
#include "webrtc/modules/video_processing/main/interface/video_processing.h"
#include "webrtc/typedefs.h"
diff --git a/webrtc/video_engine/vie_receiver.cc b/webrtc/video_engine/vie_receiver.cc
index 2e3b588..c99fa76 100644
--- a/webrtc/video_engine/vie_receiver.cc
+++ b/webrtc/video_engine/vie_receiver.cc
@@ -14,14 +14,14 @@
#include "webrtc/base/logging.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/fec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/metrics.h"
diff --git a/webrtc/video_engine/vie_receiver.h b/webrtc/video_engine/vie_receiver.h
index cd069ea..20a9627 100644
--- a/webrtc/video_engine/vie_receiver.h
+++ b/webrtc/video_engine/vie_receiver.h
@@ -15,8 +15,8 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
#include "webrtc/video_engine/vie_defines.h"
diff --git a/webrtc/video_engine/vie_remb.cc b/webrtc/video_engine/vie_remb.cc
index b347f2e..3901d6d 100644
--- a/webrtc/video_engine/vie_remb.cc
+++ b/webrtc/video_engine/vie_remb.cc
@@ -14,8 +14,8 @@
#include <algorithm>
-#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/tick_util.h"
#include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/video_engine/vie_remb.h b/webrtc/video_engine/vie_remb.h
index 9f38259..6a79ffe 100644
--- a/webrtc/video_engine/vie_remb.h
+++ b/webrtc/video_engine/vie_remb.h
@@ -16,9 +16,9 @@
#include <vector>
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
diff --git a/webrtc/video_engine/vie_remb_unittest.cc b/webrtc/video_engine/vie_remb_unittest.cc
index 3289c4b..b6fbf29 100644
--- a/webrtc/video_engine/vie_remb_unittest.cc
+++ b/webrtc/video_engine/vie_remb_unittest.cc
@@ -17,9 +17,9 @@
#include <vector>
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/mock/mock_process_thread.h"
+#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/video_engine/vie_remb.h"
diff --git a/webrtc/video_engine/vie_sync_module.cc b/webrtc/video_engine/vie_sync_module.cc
index e7327eb..b2e2713 100644
--- a/webrtc/video_engine/vie_sync_module.cc
+++ b/webrtc/video_engine/vie_sync_module.cc
@@ -12,8 +12,8 @@
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/video_engine/stream_synchronization.h"
diff --git a/webrtc/video_engine/vie_sync_module.h b/webrtc/video_engine/vie_sync_module.h
index 2a343b8..dcd8072 100644
--- a/webrtc/video_engine/vie_sync_module.h
+++ b/webrtc/video_engine/vie_sync_module.h
@@ -15,7 +15,7 @@
#define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/video_engine/stream_synchronization.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index ed47b57..0a3e696 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -19,13 +19,13 @@
#include "webrtc/config.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.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/system_wrappers/include/trace.h"
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 0f7b543..ba18aaa 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -15,14 +15,14 @@
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
#include "webrtc/modules/audio_processing/rms_level.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "webrtc/modules/utility/interface/file_player.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/dtmf_inband.h"
#include "webrtc/voice_engine/dtmf_inband_queue.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
diff --git a/webrtc/voice_engine/level_indicator.cc b/webrtc/voice_engine/level_indicator.cc
index 059b15f..68a837e 100644
--- a/webrtc/voice_engine/level_indicator.cc
+++ b/webrtc/voice_engine/level_indicator.cc
@@ -9,7 +9,7 @@
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/voice_engine/level_indicator.h"
diff --git a/webrtc/voice_engine/monitor_module.h b/webrtc/voice_engine/monitor_module.h
index 42ea74d..fe915b3 100644
--- a/webrtc/voice_engine/monitor_module.h
+++ b/webrtc/voice_engine/monitor_module.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
#define WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
-#include "webrtc/modules/interface/module.h"
+#include "webrtc/modules/include/module.h"
#include "webrtc/typedefs.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
diff --git a/webrtc/voice_engine/output_mixer.cc b/webrtc/voice_engine/output_mixer.cc
index 31b429c..1b4d2e2 100644
--- a/webrtc/voice_engine/output_mixer.cc
+++ b/webrtc/voice_engine/output_mixer.cc
@@ -11,7 +11,7 @@
#include "webrtc/voice_engine/output_mixer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/voice_engine/output_mixer.h b/webrtc/voice_engine/output_mixer.h
index 71e55e4..8c0a2de 100644
--- a/webrtc/voice_engine/output_mixer.h
+++ b/webrtc/voice_engine/output_mixer.h
@@ -13,9 +13,9 @@
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h"
-#include "webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer.h"
+#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/dtmf_inband.h"
#include "webrtc/voice_engine/level_indicator.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
diff --git a/webrtc/voice_engine/shared_data.h b/webrtc/voice_engine/shared_data.h
index 311bfa0..9c3d4b1 100644
--- a/webrtc/voice_engine/shared_data.h
+++ b/webrtc/voice_engine/shared_data.h
@@ -14,7 +14,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/voice_engine/channel_manager.h"
#include "webrtc/voice_engine/statistics.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
diff --git a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
index 7b45e6d..fb430eb 100644
--- a/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/webrtc/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -19,7 +19,7 @@
#include "webrtc/base/basictypes.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.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"
diff --git a/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc b/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
index b4daba5..4f86010 100644
--- a/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/external_media_test.cc
@@ -9,7 +9,7 @@
*/
#include "webrtc/base/arraysize.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/voice_engine/include/voe_external_media.h"
#include "webrtc/voice_engine/test/auto_test/fakes/fake_media_process.h"
#include "webrtc/voice_engine/test/auto_test/fixtures/after_streaming_fixture.h"
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
index 780a7f7..1dc15df 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/system_wrappers/include/atomic32.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h"
diff --git a/webrtc/voice_engine/transmit_mixer.cc b/webrtc/voice_engine/transmit_mixer.cc
index 94592cf..b237414 100644
--- a/webrtc/voice_engine/transmit_mixer.cc
+++ b/webrtc/voice_engine/transmit_mixer.cc
@@ -11,7 +11,7 @@
#include "webrtc/voice_engine/transmit_mixer.h"
#include "webrtc/base/format_macros.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
diff --git a/webrtc/voice_engine/transmit_mixer.h b/webrtc/voice_engine/transmit_mixer.h
index 714efb4..071d91d 100644
--- a/webrtc/voice_engine/transmit_mixer.h
+++ b/webrtc/voice_engine/transmit_mixer.h
@@ -15,9 +15,9 @@
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_processing/typing_detection.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/file_player.h"
-#include "webrtc/modules/utility/interface/file_recorder.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/file_player.h"
+#include "webrtc/modules/utility/include/file_recorder.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/level_indicator.h"
#include "webrtc/voice_engine/monitor_module.h"
diff --git a/webrtc/voice_engine/utility.cc b/webrtc/voice_engine/utility.cc
index 7bc7e0e..b041638 100644
--- a/webrtc/voice_engine/utility.cc
+++ b/webrtc/voice_engine/utility.cc
@@ -13,8 +13,8 @@
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/common_types.h"
-#include "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/utility/interface/audio_frame_operations.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/utility/include/audio_frame_operations.h"
#include "webrtc/system_wrappers/include/logging.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
diff --git a/webrtc/voice_engine/utility_unittest.cc b/webrtc/voice_engine/utility_unittest.cc
index 226e383..921c3e5 100644
--- a/webrtc/voice_engine/utility_unittest.cc
+++ b/webrtc/voice_engine/utility_unittest.cc
@@ -13,7 +13,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/voice_engine/utility.h"
#include "webrtc/voice_engine/voice_engine_defines.h"
diff --git a/webrtc/voice_engine/voe_base_impl.h b/webrtc/voice_engine/voe_base_impl.h
index f0ac959..5cd0acd 100644
--- a/webrtc/voice_engine/voe_base_impl.h
+++ b/webrtc/voice_engine/voe_base_impl.h
@@ -13,7 +13,7 @@
#include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/voice_engine/shared_data.h"
namespace webrtc {
diff --git a/webrtc/voice_engine/voe_file_impl.cc b/webrtc/voice_engine/voe_file_impl.cc
index 7927f9f..c2d5887 100644
--- a/webrtc/voice_engine/voe_file_impl.cc
+++ b/webrtc/voice_engine/voe_file_impl.cc
@@ -10,7 +10,7 @@
#include "webrtc/voice_engine/voe_file_impl.h"
-#include "webrtc/modules/media_file/interface/media_file.h"
+#include "webrtc/modules/media_file/include/media_file.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
diff --git a/webrtc/voice_engine/voice_engine_impl.cc b/webrtc/voice_engine/voice_engine_impl.cc
index c8761bc..814d619 100644
--- a/webrtc/voice_engine/voice_engine_impl.cc
+++ b/webrtc/voice_engine/voice_engine_impl.cc
@@ -12,7 +12,7 @@
#include "webrtc/modules/audio_device/android/audio_device_template.h"
#include "webrtc/modules/audio_device/android/audio_record_jni.h"
#include "webrtc/modules/audio_device/android/audio_track_jni.h"
-#include "webrtc/modules/utility/interface/jvm_android.h"
+#include "webrtc/modules/utility/include/jvm_android.h"
#endif
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
