Merge from Chromium at DEPS revision 267519
This commit was generated by merge_to_master.py.
Change-Id: I181351d1162410185322e6d49c5d11412bea2e28
diff --git a/build/apk_tests.gyp b/build/apk_tests.gyp
index f916cdf..0185975 100644
--- a/build/apk_tests.gyp
+++ b/build/apk_tests.gyp
@@ -89,20 +89,6 @@
],
},
{
- 'target_name': 'neteq_unittests_apk',
- 'type': 'none',
- 'variables': {
- 'test_suite_name': 'neteq_unittests',
- 'input_shlib_path': '<(SHARED_LIB_DIR)/<(SHARED_LIB_PREFIX)neteq_unittests<(SHARED_LIB_SUFFIX)',
- },
- 'dependencies': [
- '<(webrtc_root)/modules/modules.gyp:neteq_unittests',
- ],
- 'includes': [
- '../../../build/apk_test.gypi',
- ],
- },
- {
'target_name': 'system_wrappers_unittests_apk',
'type': 'none',
'variables': {
diff --git a/build/apk_tests_noop.gyp b/build/apk_tests_noop.gyp
index e8d39d7..3523e79 100644
--- a/build/apk_tests_noop.gyp
+++ b/build/apk_tests_noop.gyp
@@ -30,10 +30,6 @@
'type': 'none',
},
{
- 'target_name': 'neteq_unittests_apk',
- 'type': 'none',
- },
- {
'target_name': 'system_wrappers_unittests_apk',
'type': 'none',
},
diff --git a/build/download_vs_toolchain.py b/build/download_vs_toolchain.py
new file mode 100644
index 0000000..2462bdc
--- /dev/null
+++ b/build/download_vs_toolchain.py
@@ -0,0 +1,30 @@
+#!/usr/bin/env python
+#
+# 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.
+
+# This script is used to run the vs_toolchain.py script to download the
+# Visual Studio toolchain. It's just a temporary measure while waiting for the
+# Chrome team to move find_depot_tools into src/build to get rid of these
+# workarounds (similar one in gyp_webrtc).
+
+import os
+import sys
+
+
+script_dir = os.path.dirname(os.path.realpath(__file__))
+checkout_root = os.path.abspath(os.path.join(script_dir, os.pardir, os.pardir))
+sys.path.insert(0, os.path.join(checkout_root, 'build'))
+sys.path.insert(0, os.path.join(checkout_root, 'tools', 'find_depot_tools'))
+
+
+import vs_toolchain
+
+
+if __name__ == '__main__':
+ sys.exit(vs_toolchain.main())
diff --git a/build/gyp_webrtc b/build/gyp_webrtc
index 9ac4c14..4d5ae79 100755
--- a/build/gyp_webrtc
+++ b/build/gyp_webrtc
@@ -61,7 +61,9 @@
if not os.environ.get('GYP_GENERATORS'):
os.environ['GYP_GENERATORS'] = 'ninja'
- vs2013_runtime_dll_dirs = vs_toolchain.DownloadVsToolchain()
+ vs2013_runtime_dll_dirs = None
+ if int(os.environ.get('DEPOT_TOOLS_WIN_TOOLCHAIN', '1')):
+ vs2013_runtime_dll_dirs = vs_toolchain.SetEnvironmentAndGetRuntimeDllDirs()
# Enforce gyp syntax checking. This adds about 20% execution time.
args.append('--check')
diff --git a/call.h b/call.h
index 5b55682..932770d 100644
--- a/call.h
+++ b/call.h
@@ -54,8 +54,6 @@
: webrtc_config(NULL),
send_transport(send_transport),
voice_engine(NULL),
- trace_callback(NULL),
- trace_filter(kTraceDefault),
overuse_callback(NULL) {}
webrtc::Config* webrtc_config;
@@ -65,9 +63,6 @@
// VoiceEngine used for audio/video synchronization for this Call.
VoiceEngine* voice_engine;
- TraceCallback* trace_callback;
- uint32_t trace_filter;
-
// Callback for overuse and normal usage based on the jitter of incoming
// captured frames. 'NULL' disables the callback.
OveruseCallback* overuse_callback;
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/common_audio/common_audio.target.darwin-arm64.mk
similarity index 64%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to common_audio/common_audio.target.darwin-arm64.mk
index c84869c..8d0253d 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/common_audio/common_audio.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_common_audio_common_audio_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,61 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/common_audio/audio_util.cc \
+ third_party/webrtc/common_audio/fir_filter.cc \
+ third_party/webrtc/common_audio/resampler/push_resampler.cc \
+ third_party/webrtc/common_audio/resampler/push_sinc_resampler.cc \
+ third_party/webrtc/common_audio/resampler/resampler.cc \
+ third_party/webrtc/common_audio/resampler/sinc_resampler.cc \
+ third_party/webrtc/common_audio/signal_processing/auto_corr_to_refl_coef.c \
+ third_party/webrtc/common_audio/signal_processing/auto_correlation.c \
+ third_party/webrtc/common_audio/signal_processing/complex_fft.c \
+ third_party/webrtc/common_audio/signal_processing/complex_bit_reverse.c \
+ third_party/webrtc/common_audio/signal_processing/copy_set_operations.c \
+ third_party/webrtc/common_audio/signal_processing/cross_correlation.c \
+ third_party/webrtc/common_audio/signal_processing/division_operations.c \
+ third_party/webrtc/common_audio/signal_processing/dot_product_with_scale.c \
+ third_party/webrtc/common_audio/signal_processing/downsample_fast.c \
+ third_party/webrtc/common_audio/signal_processing/energy.c \
+ third_party/webrtc/common_audio/signal_processing/filter_ar.c \
+ third_party/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c \
+ third_party/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c \
+ third_party/webrtc/common_audio/signal_processing/get_hanning_window.c \
+ third_party/webrtc/common_audio/signal_processing/get_scaling_square.c \
+ third_party/webrtc/common_audio/signal_processing/ilbc_specific_functions.c \
+ third_party/webrtc/common_audio/signal_processing/levinson_durbin.c \
+ third_party/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c \
+ third_party/webrtc/common_audio/signal_processing/min_max_operations.c \
+ third_party/webrtc/common_audio/signal_processing/randomization_functions.c \
+ third_party/webrtc/common_audio/signal_processing/refl_coef_to_lpc.c \
+ third_party/webrtc/common_audio/signal_processing/real_fft.c \
+ third_party/webrtc/common_audio/signal_processing/resample.c \
+ third_party/webrtc/common_audio/signal_processing/resample_48khz.c \
+ third_party/webrtc/common_audio/signal_processing/resample_by_2.c \
+ third_party/webrtc/common_audio/signal_processing/resample_by_2_internal.c \
+ third_party/webrtc/common_audio/signal_processing/resample_fractional.c \
+ third_party/webrtc/common_audio/signal_processing/spl_init.c \
+ third_party/webrtc/common_audio/signal_processing/spl_sqrt.c \
+ third_party/webrtc/common_audio/signal_processing/spl_sqrt_floor.c \
+ third_party/webrtc/common_audio/signal_processing/spl_version.c \
+ third_party/webrtc/common_audio/signal_processing/splitting_filter.c \
+ third_party/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c \
+ third_party/webrtc/common_audio/signal_processing/vector_scaling_operations.c \
+ third_party/webrtc/common_audio/vad/webrtc_vad.c \
+ third_party/webrtc/common_audio/vad/vad_core.c \
+ third_party/webrtc/common_audio/vad/vad_filterbank.c \
+ third_party/webrtc/common_audio/vad/vad_gmm.c \
+ third_party/webrtc/common_audio/vad/vad_sp.c
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,17 +88,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +130,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +148,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +168,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -185,17 +183,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +225,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +244,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +274,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +288,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +310,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_common_audio_common_audio_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: common_audio
+common_audio: third_party_webrtc_common_audio_common_audio_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/common_audio/common_audio.target.linux-arm64.mk
similarity index 64%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to common_audio/common_audio.target.linux-arm64.mk
index c84869c..8d0253d 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/common_audio/common_audio.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_common_audio_common_audio_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,61 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/common_audio/audio_util.cc \
+ third_party/webrtc/common_audio/fir_filter.cc \
+ third_party/webrtc/common_audio/resampler/push_resampler.cc \
+ third_party/webrtc/common_audio/resampler/push_sinc_resampler.cc \
+ third_party/webrtc/common_audio/resampler/resampler.cc \
+ third_party/webrtc/common_audio/resampler/sinc_resampler.cc \
+ third_party/webrtc/common_audio/signal_processing/auto_corr_to_refl_coef.c \
+ third_party/webrtc/common_audio/signal_processing/auto_correlation.c \
+ third_party/webrtc/common_audio/signal_processing/complex_fft.c \
+ third_party/webrtc/common_audio/signal_processing/complex_bit_reverse.c \
+ third_party/webrtc/common_audio/signal_processing/copy_set_operations.c \
+ third_party/webrtc/common_audio/signal_processing/cross_correlation.c \
+ third_party/webrtc/common_audio/signal_processing/division_operations.c \
+ third_party/webrtc/common_audio/signal_processing/dot_product_with_scale.c \
+ third_party/webrtc/common_audio/signal_processing/downsample_fast.c \
+ third_party/webrtc/common_audio/signal_processing/energy.c \
+ third_party/webrtc/common_audio/signal_processing/filter_ar.c \
+ third_party/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c \
+ third_party/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c \
+ third_party/webrtc/common_audio/signal_processing/get_hanning_window.c \
+ third_party/webrtc/common_audio/signal_processing/get_scaling_square.c \
+ third_party/webrtc/common_audio/signal_processing/ilbc_specific_functions.c \
+ third_party/webrtc/common_audio/signal_processing/levinson_durbin.c \
+ third_party/webrtc/common_audio/signal_processing/lpc_to_refl_coef.c \
+ third_party/webrtc/common_audio/signal_processing/min_max_operations.c \
+ third_party/webrtc/common_audio/signal_processing/randomization_functions.c \
+ third_party/webrtc/common_audio/signal_processing/refl_coef_to_lpc.c \
+ third_party/webrtc/common_audio/signal_processing/real_fft.c \
+ third_party/webrtc/common_audio/signal_processing/resample.c \
+ third_party/webrtc/common_audio/signal_processing/resample_48khz.c \
+ third_party/webrtc/common_audio/signal_processing/resample_by_2.c \
+ third_party/webrtc/common_audio/signal_processing/resample_by_2_internal.c \
+ third_party/webrtc/common_audio/signal_processing/resample_fractional.c \
+ third_party/webrtc/common_audio/signal_processing/spl_init.c \
+ third_party/webrtc/common_audio/signal_processing/spl_sqrt.c \
+ third_party/webrtc/common_audio/signal_processing/spl_sqrt_floor.c \
+ third_party/webrtc/common_audio/signal_processing/spl_version.c \
+ third_party/webrtc/common_audio/signal_processing/splitting_filter.c \
+ third_party/webrtc/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c \
+ third_party/webrtc/common_audio/signal_processing/vector_scaling_operations.c \
+ third_party/webrtc/common_audio/vad/webrtc_vad.c \
+ third_party/webrtc/common_audio/vad/vad_core.c \
+ third_party/webrtc/common_audio/vad/vad_filterbank.c \
+ third_party/webrtc/common_audio/vad/vad_gmm.c \
+ third_party/webrtc/common_audio/vad/vad_sp.c
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,17 +88,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +130,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +148,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +168,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -185,17 +183,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +225,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +244,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +274,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +288,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +310,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_common_audio_common_audio_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: common_audio
+common_audio: third_party_webrtc_common_audio_common_audio_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/common_audio/resampler/push_sinc_resampler.cc b/common_audio/resampler/push_sinc_resampler.cc
index 3469ff3..0275559 100644
--- a/common_audio/resampler/push_sinc_resampler.cc
+++ b/common_audio/resampler/push_sinc_resampler.cc
@@ -10,6 +10,7 @@
#include "webrtc/common_audio/include/audio_util.h"
+#include <assert.h>
#include <string.h>
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
diff --git a/common_audio/resampler/sinc_resampler.cc b/common_audio/resampler/sinc_resampler.cc
index 5029934..84f8125 100644
--- a/common_audio/resampler/sinc_resampler.cc
+++ b/common_audio/resampler/sinc_resampler.cc
@@ -90,6 +90,7 @@
#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
#include "webrtc/typedefs.h"
+#include <assert.h>
#include <math.h>
#include <string.h>
diff --git a/common_audio/resampler/sinc_resampler_unittest.cc b/common_audio/resampler/sinc_resampler_unittest.cc
index c085cfc..9790862 100644
--- a/common_audio/resampler/sinc_resampler_unittest.cc
+++ b/common_audio/resampler/sinc_resampler_unittest.cc
@@ -62,7 +62,7 @@
static const int kChunks = 2;
int max_chunk_size = resampler.ChunkSize() * kChunks;
- scoped_array<float> resampled_destination(new float[max_chunk_size]);
+ scoped_ptr<float[]> resampled_destination(new float[max_chunk_size]);
// Verify requesting ChunkSize() frames causes a single callback.
EXPECT_CALL(mock_source, Run(_, _))
@@ -81,7 +81,7 @@
MockSource mock_source;
SincResampler resampler(kSampleRateRatio, SincResampler::kDefaultRequestSize,
&mock_source);
- scoped_array<float> resampled_destination(new float[resampler.ChunkSize()]);
+ scoped_ptr<float[]> resampled_destination(new float[resampler.ChunkSize()]);
// Fill the resampler with junk data.
EXPECT_CALL(mock_source, Run(_, _))
@@ -266,7 +266,7 @@
// Force an update to the sample rate ratio to ensure dyanmic sample rate
// changes are working correctly.
- scoped_array<float> kernel(new float[SincResampler::kKernelStorageSize]);
+ scoped_ptr<float[]> kernel(new float[SincResampler::kKernelStorageSize]);
memcpy(kernel.get(), resampler.get_kernel_for_testing(),
SincResampler::kKernelStorageSize);
resampler.SetRatio(M_PI);
@@ -278,8 +278,8 @@
// TODO(dalecurtis): If we switch to AVX/SSE optimization, we'll need to
// allocate these on 32-byte boundaries and ensure they're sized % 32 bytes.
- scoped_array<float> resampled_destination(new float[output_samples]);
- scoped_array<float> pure_destination(new float[output_samples]);
+ scoped_ptr<float[]> resampled_destination(new float[output_samples]);
+ scoped_ptr<float[]> pure_destination(new float[output_samples]);
// Generate resampled signal.
resampler.Resample(output_samples, resampled_destination.get());
diff --git a/common_audio/signal_processing/spl_init.c b/common_audio/signal_processing/spl_init.c
index 454e13b..762f9e4 100644
--- a/common_audio/signal_processing/spl_init.c
+++ b/common_audio/signal_processing/spl_init.c
@@ -65,8 +65,10 @@
WebRtcSpl_MinValueW32 = WebRtcSpl_MinValueW32Neon;
WebRtcSpl_CrossCorrelation = WebRtcSpl_CrossCorrelationNeon;
WebRtcSpl_DownsampleFast = WebRtcSpl_DownsampleFastNeon;
+ /* TODO(henrik.lundin): re-enable NEON when the crash from bug 3243 is
+ understood. */
WebRtcSpl_ScaleAndAddVectorsWithRound =
- WebRtcSpl_ScaleAndAddVectorsWithRoundNeon;
+ WebRtcSpl_ScaleAndAddVectorsWithRoundC;
WebRtcSpl_CreateRealFFT = WebRtcSpl_CreateRealFFTNeon;
WebRtcSpl_FreeRealFFT = WebRtcSpl_FreeRealFFTNeon;
WebRtcSpl_RealForwardFFT = WebRtcSpl_RealForwardFFTNeon;
diff --git a/common_types.h b/common_types.h
index 68f2baa..2d93102 100644
--- a/common_types.h
+++ b/common_types.h
@@ -720,17 +720,17 @@
// This structure will have the information about when packet is actually
// received by socket.
struct PacketTime {
- PacketTime() : timestamp(-1), max_error_us(-1) {}
- PacketTime(int64_t timestamp, int64_t max_error_us)
- : timestamp(timestamp), max_error_us(max_error_us) {
+ PacketTime() : timestamp(-1), not_before(-1) {}
+ PacketTime(int64_t timestamp, int64_t not_before)
+ : timestamp(timestamp), not_before(not_before) {
}
- int64_t timestamp; // Receive time after socket delivers the data.
- int64_t max_error_us; // Earliest possible time the data could have arrived,
- // indicating the potential error in the |timestamp|
- // value,in case the system is busy.
- // For example, the time of the last select() call.
- // If unknown, this value will be set to zero.
+ int64_t timestamp; // Receive time after socket delivers the data.
+ int64_t not_before; // Earliest possible time the data could have arrived,
+ // indicating the potential error in the |timestamp|
+ // value,in case the system is busy.
+ // For example, the time of the last select() call.
+ // If unknown, this value will be set to zero.
};
struct RTPHeaderExtension {
diff --git a/modules/NetEq.target.darwin-x86.mk b/common_video/common_video.target.darwin-arm64.mk
similarity index 62%
copy from modules/NetEq.target.darwin-x86.mk
copy to common_video/common_video.target.darwin-arm64.mk
index 3b05cd9..21e28a5 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/common_video/common_video.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_common_video_common_video_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,17 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/common_video/i420_video_frame.cc \
+ third_party/webrtc/common_video/libyuv/webrtc_libyuv.cc \
+ third_party/webrtc/common_video/libyuv/scaler.cc \
+ third_party/webrtc/common_video/plane.cc \
+ third_party/webrtc/common_video/texture_video_frame.cc
# Flags passed to both C and C++ files.
@@ -70,19 +43,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +59,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +108,12 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/libyuv/include \
+ $(LOCAL_PATH)/third_party/libyuv \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,19 +141,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +157,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +188,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +207,12 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/libyuv/include \
+ $(LOCAL_PATH)/third_party/libyuv \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +240,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +254,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +276,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_common_video_common_video_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: common_video
+common_video: third_party_webrtc_common_video_common_video_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/common_video/common_video.target.linux-arm64.mk
similarity index 62%
copy from modules/NetEq.target.darwin-x86.mk
copy to common_video/common_video.target.linux-arm64.mk
index 3b05cd9..21e28a5 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/common_video/common_video.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_common_video_common_video_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,17 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/common_video/i420_video_frame.cc \
+ third_party/webrtc/common_video/libyuv/webrtc_libyuv.cc \
+ third_party/webrtc/common_video/libyuv/scaler.cc \
+ third_party/webrtc/common_video/plane.cc \
+ third_party/webrtc/common_video/texture_video_frame.cc
# Flags passed to both C and C++ files.
@@ -70,19 +43,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +59,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +108,12 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/libyuv/include \
+ $(LOCAL_PATH)/third_party/libyuv \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,19 +141,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +157,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +188,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +207,12 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/libyuv/include \
+ $(LOCAL_PATH)/third_party/libyuv \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +240,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +254,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +276,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_common_video_common_video_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: common_video
+common_video: third_party_webrtc_common_video_common_video_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/common_video/interface/i420_video_frame.h b/common_video/interface/i420_video_frame.h
index 4579484..3f90a8e 100644
--- a/common_video/interface/i420_video_frame.h
+++ b/common_video/interface/i420_video_frame.h
@@ -15,6 +15,8 @@
//
// Storing and handling of YUV (I420) video frames.
+#include <assert.h>
+
#include "webrtc/common_video/plane.h"
#include "webrtc/system_wrappers/interface/scoped_refptr.h"
#include "webrtc/typedefs.h"
diff --git a/common_video/libyuv/libyuv_unittest.cc b/common_video/libyuv/libyuv_unittest.cc
index 3df520e..0abe7f3 100644
--- a/common_video/libyuv/libyuv_unittest.cc
+++ b/common_video/libyuv/libyuv_unittest.cc
@@ -84,7 +84,7 @@
FILE* source_file_;
I420VideoFrame orig_frame_;
- scoped_array<uint8_t> orig_buffer_;
+ scoped_ptr<uint8_t[]> orig_buffer_;
const int width_;
const int height_;
const int size_y_;
@@ -147,7 +147,7 @@
(width_ + 1) / 2,
(width_ + 1) / 2));
printf("\nConvert #%d I420 <-> I420 \n", j);
- scoped_array<uint8_t> out_i420_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> out_i420_buffer(new uint8_t[frame_length_]);
EXPECT_EQ(0, ConvertFromI420(orig_frame_, kI420, 0,
out_i420_buffer.get()));
EXPECT_EQ(0, ConvertToI420(kI420, out_i420_buffer.get(), 0, 0,
@@ -162,7 +162,7 @@
j++;
printf("\nConvert #%d I420 <-> RGB24\n", j);
- scoped_array<uint8_t> res_rgb_buffer2(new uint8_t[width_ * height_ * 3]);
+ scoped_ptr<uint8_t[]> res_rgb_buffer2(new uint8_t[width_ * height_ * 3]);
// Align the stride values for the output frame.
int stride_y = 0;
int stride_uv = 0;
@@ -184,7 +184,7 @@
j++;
printf("\nConvert #%d I420 <-> UYVY\n", j);
- scoped_array<uint8_t> out_uyvy_buffer(new uint8_t[width_ * height_ * 2]);
+ scoped_ptr<uint8_t[]> out_uyvy_buffer(new uint8_t[width_ * height_ * 2]);
EXPECT_EQ(0, ConvertFromI420(orig_frame_, kUYVY, 0, out_uyvy_buffer.get()));
EXPECT_EQ(0, ConvertToI420(kUYVY, out_uyvy_buffer.get(), 0, 0, width_,
height_, 0, kRotateNone, &res_i420_frame));
@@ -196,8 +196,8 @@
j++;
printf("\nConvert #%d I420 <-> YV12\n", j);
- scoped_array<uint8_t> outYV120Buffer(new uint8_t[frame_length_]);
- scoped_array<uint8_t> res_i420_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> outYV120Buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> res_i420_buffer(new uint8_t[frame_length_]);
I420VideoFrame yv12_frame;
EXPECT_EQ(0, ConvertFromI420(orig_frame_, kYV12, 0, outYV120Buffer.get()));
yv12_frame.CreateFrame(size_y_, outYV120Buffer.get(),
@@ -218,7 +218,7 @@
j++;
printf("\nConvert #%d I420 <-> YUY2\n", j);
- scoped_array<uint8_t> out_yuy2_buffer(new uint8_t[width_ * height_ * 2]);
+ scoped_ptr<uint8_t[]> out_yuy2_buffer(new uint8_t[width_ * height_ * 2]);
EXPECT_EQ(0, ConvertFromI420(orig_frame_, kYUY2, 0, out_yuy2_buffer.get()));
EXPECT_EQ(0, ConvertToI420(kYUY2, out_yuy2_buffer.get(), 0, 0, width_,
@@ -231,7 +231,7 @@
psnr = I420PSNR(&orig_frame_, &res_i420_frame);
EXPECT_EQ(48.0, psnr);
printf("\nConvert #%d I420 <-> RGB565\n", j);
- scoped_array<uint8_t> out_rgb565_buffer(new uint8_t[width_ * height_ * 2]);
+ scoped_ptr<uint8_t[]> out_rgb565_buffer(new uint8_t[width_ * height_ * 2]);
EXPECT_EQ(0, ConvertFromI420(orig_frame_, kRGB565, 0,
out_rgb565_buffer.get()));
@@ -250,7 +250,7 @@
EXPECT_GT(ceil(psnr), 40);
printf("\nConvert #%d I420 <-> ARGB8888\n", j);
- scoped_array<uint8_t> out_argb8888_buffer(new uint8_t[width_ * height_ * 4]);
+ scoped_ptr<uint8_t[]> out_argb8888_buffer(new uint8_t[width_ * height_ * 4]);
EXPECT_EQ(0, ConvertFromI420(orig_frame_, kARGB, 0,
out_argb8888_buffer.get()));
@@ -283,7 +283,7 @@
Calc16ByteAlignedStride(width_, &stride_y, &stride_uv);
EXPECT_EQ(0,res_i420_frame.CreateEmptyFrame(width_, height_,
stride_y, stride_uv, stride_uv));
- scoped_array<uint8_t> out_i420_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> out_i420_buffer(new uint8_t[frame_length_]);
EXPECT_EQ(0, ConvertFromI420(orig_frame_, kI420, 0,
out_i420_buffer.get()));
EXPECT_EQ(0, ConvertToI420(kI420, out_i420_buffer.get(), 0, 0,
diff --git a/common_video/libyuv/scaler_unittest.cc b/common_video/libyuv/scaler_unittest.cc
index fee10df..f186d82 100644
--- a/common_video/libyuv/scaler_unittest.cc
+++ b/common_video/libyuv/scaler_unittest.cc
@@ -99,7 +99,7 @@
kI420, kI420,
kScalePoint));
I420VideoFrame test_frame2;
- scoped_array<uint8_t> orig_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> orig_buffer(new uint8_t[frame_length_]);
EXPECT_GT(fread(orig_buffer.get(), 1, frame_length_, source_file_), 0U);
test_frame_.CreateFrame(size_y_, orig_buffer.get(),
size_uv_, orig_buffer.get() + size_y_,
@@ -442,7 +442,7 @@
total_clock = 0;
int frame_count = 0;
int src_required_size = CalcBufferSize(kI420, src_width, src_height);
- scoped_array<uint8_t> frame_buffer(new uint8_t[src_required_size]);
+ scoped_ptr<uint8_t[]> frame_buffer(new uint8_t[src_required_size]);
int size_y = src_width * src_height;
int size_uv = ((src_width + 1) / 2) * ((src_height + 1) / 2);
diff --git a/examples/android/opensl_loopback/fake_audio_device_buffer.cc b/examples/android/opensl_loopback/fake_audio_device_buffer.cc
index 81adc8f..23b60ee 100644
--- a/examples/android/opensl_loopback/fake_audio_device_buffer.cc
+++ b/examples/android/opensl_loopback/fake_audio_device_buffer.cc
@@ -22,7 +22,7 @@
next_available_buffer_(0),
record_channels_(0),
play_channels_(0) {
- buf_.reset(new scoped_array<int8_t>[kNumBuffers]);
+ buf_.reset(new scoped_ptr<int8_t[]>[kNumBuffers]);
for (int i = 0; i < kNumBuffers; ++i) {
buf_[i].reset(new int8_t[buffer_size_bytes()]);
}
diff --git a/examples/android/opensl_loopback/fake_audio_device_buffer.h b/examples/android/opensl_loopback/fake_audio_device_buffer.h
index b98ee1e..1ef866c 100644
--- a/examples/android/opensl_loopback/fake_audio_device_buffer.h
+++ b/examples/android/opensl_loopback/fake_audio_device_buffer.h
@@ -55,7 +55,7 @@
AudioManagerJni audio_manager_;
SingleRwFifo fifo_;
- scoped_array<scoped_array<int8_t> > buf_;
+ scoped_ptr<scoped_ptr<int8_t[]>[]> buf_;
int next_available_buffer_;
uint8_t record_channels_;
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/CNG.target.darwin-arm64.mk
similarity index 71%
rename from modules/NetEq.target.darwin-x86.mk
rename to modules/CNG.target.darwin-arm64.mk
index 3b05cd9..dde23ab 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/CNG.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_CNG_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,8 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c \
+ third_party/webrtc/modules/audio_coding/codecs/cng/cng_helpfuns.c
# Flags passed to both C and C++ files.
@@ -70,6 +39,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +48,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +59,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,9 +108,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -179,6 +140,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +149,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +160,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +191,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,9 +210,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -289,8 +242,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +256,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +278,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_CNG_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: CNG
+CNG: third_party_webrtc_modules_CNG_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/CNG.target.linux-arm64.mk
similarity index 71%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/CNG.target.linux-arm64.mk
index 3b05cd9..dde23ab 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/CNG.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_CNG_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,8 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c \
+ third_party/webrtc/modules/audio_coding/codecs/cng/cng_helpfuns.c
# Flags passed to both C and C++ files.
@@ -70,6 +39,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +48,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +59,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,9 +108,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -179,6 +140,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +149,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +160,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +191,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,9 +210,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -289,8 +242,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +256,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +278,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_CNG_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: CNG
+CNG: third_party_webrtc_modules_CNG_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/G711.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/G711.target.darwin-arm64.mk
index 3b05cd9..d27a628 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/G711.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_G711_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,8 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/g711/g711_interface.c \
+ third_party/webrtc/modules/audio_coding/codecs/g711/g711.c
# Flags passed to both C and C++ files.
@@ -70,6 +39,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +48,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +59,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +108,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +137,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +146,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +157,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +188,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +207,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +236,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +250,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +272,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_G711_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: G711
+G711: third_party_webrtc_modules_G711_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/G711.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/G711.target.linux-arm64.mk
index 3b05cd9..d27a628 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/G711.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_G711_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,8 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/g711/g711_interface.c \
+ third_party/webrtc/modules/audio_coding/codecs/g711/g711.c
# Flags passed to both C and C++ files.
@@ -70,6 +39,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +48,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +59,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +108,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +137,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +146,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +157,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +188,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +207,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +236,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +250,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +272,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_G711_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: G711
+G711: third_party_webrtc_modules_G711_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/G722.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/G722.target.darwin-arm64.mk
index 3b05cd9..11a4919 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/G722.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_G722_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,9 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/g722/g722_interface.c \
+ third_party/webrtc/modules/audio_coding/codecs/g722/g722_encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
# Flags passed to both C and C++ files.
@@ -70,6 +40,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +49,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +60,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +91,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +109,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +138,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +147,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +158,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +189,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +208,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +237,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +251,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +273,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_G722_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: G722
+G722: third_party_webrtc_modules_G722_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/G722.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/G722.target.linux-arm64.mk
index 3b05cd9..11a4919 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/G722.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_G722_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,9 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/g722/g722_interface.c \
+ third_party/webrtc/modules/audio_coding/codecs/g722/g722_encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
# Flags passed to both C and C++ files.
@@ -70,6 +40,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +49,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +60,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +91,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +109,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +138,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +147,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +158,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +189,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +208,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +237,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +251,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +273,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_G722_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: G722
+G722: third_party_webrtc_modules_G722_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-arm.mk b/modules/NetEq.target.darwin-arm.mk
deleted file mode 100644
index f32b25e..0000000
--- a/modules/NetEq.target.darwin-arm.mk
+++ /dev/null
@@ -1,348 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-tree-sra \
- -fuse-ld=gold \
- -Wno-psabi \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_ARCH_ARM' \
- '-DWEBRTC_ARCH_ARM_V7' \
- '-DWEBRTC_DETECT_ARM_NEON' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-abi \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-tree-sra \
- -fuse-ld=gold \
- -Wno-psabi \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_ARCH_ARM' \
- '-DWEBRTC_ARCH_ARM_V7' \
- '-DWEBRTC_DETECT_ARM_NEON' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-abi \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -Wl,-z,relro \
- -Wl,-z,now \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--icf=safe \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -Wl,-z,relro \
- -Wl,-z,now \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--icf=safe \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
-
-# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-mips.mk b/modules/NetEq.target.darwin-mips.mk
deleted file mode 100644
index 240cc6c..0000000
--- a/modules/NetEq.target.darwin-mips.mk
+++ /dev/null
@@ -1,344 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -EL \
- -mhard-float \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DMIPS32_LE' \
- '-DMIPS_FPU_LE' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-uninitialized \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -EL \
- -mhard-float \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DMIPS32_LE' \
- '-DMIPS_FPU_LE' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-uninitialized \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -EL \
- -Wl,--no-keep-memory \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -EL \
- -Wl,--no-keep-memory \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
-
-# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.linux-arm.mk b/modules/NetEq.target.linux-arm.mk
deleted file mode 100644
index f32b25e..0000000
--- a/modules/NetEq.target.linux-arm.mk
+++ /dev/null
@@ -1,348 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-tree-sra \
- -fuse-ld=gold \
- -Wno-psabi \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_ARCH_ARM' \
- '-DWEBRTC_ARCH_ARM_V7' \
- '-DWEBRTC_DETECT_ARM_NEON' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-abi \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-tree-sra \
- -fuse-ld=gold \
- -Wno-psabi \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_ARCH_ARM' \
- '-DWEBRTC_ARCH_ARM_V7' \
- '-DWEBRTC_DETECT_ARM_NEON' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-abi \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -Wl,-z,relro \
- -Wl,-z,now \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--icf=safe \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -Wl,-z,relro \
- -Wl,-z,now \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--icf=safe \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
-
-# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.linux-mips.mk b/modules/NetEq.target.linux-mips.mk
deleted file mode 100644
index 240cc6c..0000000
--- a/modules/NetEq.target.linux-mips.mk
+++ /dev/null
@@ -1,344 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -EL \
- -mhard-float \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DMIPS32_LE' \
- '-DMIPS_FPU_LE' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-uninitialized \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -EL \
- -mhard-float \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DMIPS32_LE' \
- '-DMIPS_FPU_LE' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-uninitialized \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -EL \
- -Wl,--no-keep-memory \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -EL \
- -Wl,--no-keep-memory \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
-
-# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.linux-x86.mk b/modules/NetEq.target.linux-x86.mk
deleted file mode 100644
index 3b05cd9..0000000
--- a/modules/NetEq.target.linux-x86.mk
+++ /dev/null
@@ -1,338 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -fno-stack-protector \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -fno-stack-protector \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -m32 \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -m32 \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
-
-# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.linux-x86_64.mk b/modules/NetEq.target.linux-x86_64.mk
deleted file mode 100644
index c84869c..0000000
--- a/modules/NetEq.target.linux-x86_64.mk
+++ /dev/null
@@ -1,338 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -Wno-unused-local-typedefs \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -Wno-unused-local-typedefs \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -m64 \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -m64 \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
-
-# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/acm2.target.darwin-x86_64.mk b/modules/NetEq4.target.darwin-arm64.mk
similarity index 75%
copy from modules/acm2.target.darwin-x86_64.mk
copy to modules/NetEq4.target.darwin-arm64.mk
index e2c8787..b7f5bd4 100644
--- a/modules/acm2.target.darwin-x86_64.mk
+++ b/modules/NetEq4.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
+LOCAL_MODULE := third_party_webrtc_modules_NetEq4_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -24,39 +24,43 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
+ third_party/webrtc/modules/audio_coding/neteq4/accelerate.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/audio_classifier.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/audio_decoder_impl.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/audio_decoder.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/audio_vector.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/background_noise.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/buffer_level_filter.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/comfort_noise.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/decision_logic.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/decision_logic_fax.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/decision_logic_normal.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/decoder_database.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/delay_manager.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/delay_peak_detector.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/dtmf_buffer.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/expand.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/merge.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/neteq.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/statistics_calculator.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/normal.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/packet_buffer.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/payload_splitter.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/post_decode_vad.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/random_vector.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/rtcp.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/sync_buffer.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/time_stretch.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,13 +80,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -141,24 +141,21 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
+ $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -176,7 +173,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -196,13 +192,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -262,24 +254,21 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
+ $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -307,8 +296,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -323,8 +310,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -347,10 +332,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
+gyp_all_modules: third_party_webrtc_modules_NetEq4_gyp
# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
+.PHONY: NetEq4
+NetEq4: third_party_webrtc_modules_NetEq4_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/acm2.target.darwin-x86_64.mk b/modules/NetEq4.target.linux-arm64.mk
similarity index 75%
copy from modules/acm2.target.darwin-x86_64.mk
copy to modules/NetEq4.target.linux-arm64.mk
index e2c8787..b7f5bd4 100644
--- a/modules/acm2.target.darwin-x86_64.mk
+++ b/modules/NetEq4.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
+LOCAL_MODULE := third_party_webrtc_modules_NetEq4_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -24,39 +24,43 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
+ third_party/webrtc/modules/audio_coding/neteq4/accelerate.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/audio_classifier.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/audio_decoder_impl.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/audio_decoder.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/audio_multi_vector.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/audio_vector.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/background_noise.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/buffer_level_filter.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/comfort_noise.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/decision_logic.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/decision_logic_fax.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/decision_logic_normal.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/decoder_database.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/delay_manager.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/delay_peak_detector.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/dsp_helper.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/dtmf_buffer.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/dtmf_tone_generator.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/expand.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/merge.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/neteq_impl.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/neteq.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/statistics_calculator.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/normal.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/packet_buffer.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/payload_splitter.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/post_decode_vad.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/random_vector.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/rtcp.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/sync_buffer.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc \
+ third_party/webrtc/modules/audio_coding/neteq4/time_stretch.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,13 +80,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -141,24 +141,21 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
+ $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -176,7 +173,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -196,13 +192,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -262,24 +254,21 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
+ $(LOCAL_PATH)/third_party/webrtc \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -307,8 +296,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -323,8 +310,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -347,10 +332,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
+gyp_all_modules: third_party_webrtc_modules_NetEq4_gyp
# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
+.PHONY: NetEq4
+NetEq4: third_party_webrtc_modules_NetEq4_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/PCM16B.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/PCM16B.target.darwin-arm64.mk
index 3b05cd9..c27577f 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/PCM16B.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_PCM16B_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,7 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
# Flags passed to both C and C++ files.
@@ -70,6 +38,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +47,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +58,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +89,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +107,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +136,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +145,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +156,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +187,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +206,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +235,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +249,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +271,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_PCM16B_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: PCM16B
+PCM16B: third_party_webrtc_modules_PCM16B_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/PCM16B.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/PCM16B.target.linux-arm64.mk
index 3b05cd9..c27577f 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/PCM16B.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_PCM16B_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,7 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
# Flags passed to both C and C++ files.
@@ -70,6 +38,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +47,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +58,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +89,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +107,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +136,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +145,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +156,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +187,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +206,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +235,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +249,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +271,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_PCM16B_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: PCM16B
+PCM16B: third_party_webrtc_modules_PCM16B_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/acm2.target.darwin-arm.mk b/modules/acm2.target.darwin-arm.mk
deleted file mode 100644
index 222798a..0000000
--- a/modules/acm2.target.darwin-arm.mk
+++ /dev/null
@@ -1,366 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-tree-sra \
- -fuse-ld=gold \
- -Wno-psabi \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_ARCH_ARM' \
- '-DWEBRTC_ARCH_ARM_V7' \
- '-DWEBRTC_DETECT_ARM_NEON' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-abi \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-tree-sra \
- -fuse-ld=gold \
- -Wno-psabi \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_ARCH_ARM' \
- '-DWEBRTC_ARCH_ARM_V7' \
- '-DWEBRTC_DETECT_ARM_NEON' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-abi \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -Wl,-z,relro \
- -Wl,-z,now \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--icf=safe \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -Wl,-z,relro \
- -Wl,-z,now \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--icf=safe \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
-
-# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/acm2.target.darwin-mips.mk b/modules/acm2.target.darwin-mips.mk
deleted file mode 100644
index 3c220bb..0000000
--- a/modules/acm2.target.darwin-mips.mk
+++ /dev/null
@@ -1,362 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -EL \
- -mhard-float \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DMIPS32_LE' \
- '-DMIPS_FPU_LE' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-uninitialized \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -EL \
- -mhard-float \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DMIPS32_LE' \
- '-DMIPS_FPU_LE' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-uninitialized \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -EL \
- -Wl,--no-keep-memory \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -EL \
- -Wl,--no-keep-memory \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
-
-# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/acm2.target.darwin-x86.mk b/modules/acm2.target.darwin-x86.mk
deleted file mode 100644
index 94be33a..0000000
--- a/modules/acm2.target.darwin-x86.mk
+++ /dev/null
@@ -1,356 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -fno-stack-protector \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -fno-stack-protector \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -m32 \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -m32 \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
-
-# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/acm2.target.linux-arm.mk b/modules/acm2.target.linux-arm.mk
deleted file mode 100644
index 222798a..0000000
--- a/modules/acm2.target.linux-arm.mk
+++ /dev/null
@@ -1,366 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-tree-sra \
- -fuse-ld=gold \
- -Wno-psabi \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_ARCH_ARM' \
- '-DWEBRTC_ARCH_ARM_V7' \
- '-DWEBRTC_DETECT_ARM_NEON' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-abi \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-tree-sra \
- -fuse-ld=gold \
- -Wno-psabi \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_ARCH_ARM' \
- '-DWEBRTC_ARCH_ARM_V7' \
- '-DWEBRTC_DETECT_ARM_NEON' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-abi \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -Wl,-z,relro \
- -Wl,-z,now \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--icf=safe \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -Wl,-z,relro \
- -Wl,-z,now \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--icf=safe \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
-
-# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/acm2.target.linux-mips.mk b/modules/acm2.target.linux-mips.mk
deleted file mode 100644
index 3c220bb..0000000
--- a/modules/acm2.target.linux-mips.mk
+++ /dev/null
@@ -1,362 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -EL \
- -mhard-float \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DMIPS32_LE' \
- '-DMIPS_FPU_LE' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-uninitialized \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -mhard-float \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -EL \
- -mhard-float \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DMIPS32_LE' \
- '-DMIPS_FPU_LE' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-uninitialized \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -EL \
- -Wl,--no-keep-memory \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -EL \
- -Wl,--no-keep-memory \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
-
-# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/acm2.target.linux-x86.mk b/modules/acm2.target.linux-x86.mk
deleted file mode 100644
index 94be33a..0000000
--- a/modules/acm2.target.linux-x86.mk
+++ /dev/null
@@ -1,356 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -fno-stack-protector \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -fno-stack-protector \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -m32 \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -m32 \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
-
-# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/acm2.target.linux-x86_64.mk b/modules/acm2.target.linux-x86_64.mk
deleted file mode 100644
index e2c8787..0000000
--- a/modules/acm2.target.linux-x86_64.mk
+++ /dev/null
@@ -1,356 +0,0 @@
-# This file is generated by gyp; do not edit.
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
-LOCAL_MODULE_SUFFIX := .a
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
-gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
-gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
-
-# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
-
-GYP_GENERATED_OUTPUTS :=
-
-# Make sure our deps and generated files are built first.
-LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
-
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-
-GYP_COPIED_SOURCE_ORIGIN_DIRS :=
-
-LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
- third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Debug := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -Wno-unused-local-typedefs \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -g \
- -fomit-frame-pointer \
- -fdata-sections \
- -ffunction-sections \
- -funwind-tables
-
-MY_DEFS_Debug := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
- '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
- '-D_DEBUG'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Debug := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Debug := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS_Release := \
- -fstack-protector \
- --param=ssp-buffer-size=4 \
- -Werror \
- -fno-exceptions \
- -fno-strict-aliasing \
- -Wall \
- -Wno-unused-parameter \
- -Wno-missing-field-initializers \
- -fvisibility=hidden \
- -pipe \
- -fPIC \
- -Wno-unused-local-typedefs \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
- -ffunction-sections \
- -funwind-tables \
- -g \
- -fstack-protector \
- -fno-short-enums \
- -finline-limit=64 \
- -Wa,--noexecstack \
- -U_FORTIFY_SOURCE \
- -Wno-extra \
- -Wno-ignored-qualifiers \
- -Wno-type-limits \
- -Wno-unused-but-set-variable \
- -Os \
- -fno-ident \
- -fdata-sections \
- -ffunction-sections \
- -fomit-frame-pointer \
- -funwind-tables
-
-MY_DEFS_Release := \
- '-DV8_DEPRECATION_WARNINGS' \
- '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
- '-D_FILE_OFFSET_BITS=64' \
- '-DNO_TCMALLOC' \
- '-DDISABLE_NACL' \
- '-DCHROMIUM_BUILD' \
- '-DUSE_LIBJPEG_TURBO=1' \
- '-DENABLE_WEBRTC=1' \
- '-DUSE_PROPRIETARY_CODECS' \
- '-DENABLE_CONFIGURATION_POLICY' \
- '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
- '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
- '-DENABLE_EGLIMAGE=1' \
- '-DCLD_VERSION=1' \
- '-DENABLE_PRINTING=1' \
- '-DENABLE_MANAGED_USERS=1' \
- '-DWEBRTC_RESTRICT_LOGGING' \
- '-DWEBRTC_MODULE_UTILITY_VIDEO' \
- '-DWEBRTC_CHROMIUM_BUILD' \
- '-DWEBRTC_POSIX' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES' \
- '-DWEBRTC_CODEC_OPUS' \
- '-DUSE_OPENSSL=1' \
- '-DUSE_OPENSSL_CERTS=1' \
- '-D__STDC_CONSTANT_MACROS' \
- '-D__STDC_FORMAT_MACROS' \
- '-DANDROID' \
- '-D__GNU_SOURCE=1' \
- '-DUSE_STLPORT=1' \
- '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
- '-DCHROME_BUILD_ID=""' \
- '-DNDEBUG' \
- '-DNVALGRIND' \
- '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
- '-D_FORTIFY_SOURCE=2'
-
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES_Release := \
- $(LOCAL_PATH)/third_party \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g711/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/g722/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
- $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
- $(LOCAL_PATH)/third_party/opus/src/celt \
- $(LOCAL_PATH)/third_party/opus/src/src \
- $(LOCAL_PATH)/third_party/opus/src/include \
- $(PWD)/frameworks/wilhelm/include \
- $(PWD)/bionic \
- $(PWD)/external/stlport/stlport
-
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS_Release := \
- -fno-rtti \
- -fno-threadsafe-statics \
- -fvisibility-inlines-hidden \
- -Wsign-compare \
- -Wno-non-virtual-dtor \
- -Wno-sign-promo
-
-
-LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
-LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
-LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
-LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
-### Rules for final target.
-
-LOCAL_LDFLAGS_Debug := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -m64 \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,--warn-shared-textrel \
- -Wl,-O1 \
- -Wl,--as-needed
-
-
-LOCAL_LDFLAGS_Release := \
- -Wl,-z,now \
- -Wl,-z,relro \
- -Wl,--fatal-warnings \
- -Wl,-z,noexecstack \
- -fPIC \
- -m64 \
- -fuse-ld=gold \
- -nostdlib \
- -Wl,--no-undefined \
- -Wl,--exclude-libs=ALL \
- -Wl,-O1 \
- -Wl,--as-needed \
- -Wl,--gc-sections \
- -Wl,--warn-shared-textrel
-
-
-LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-
-LOCAL_STATIC_LIBRARIES :=
-
-# Enable grouping to fix circular references
-LOCAL_GROUP_STATIC_LIBRARIES := true
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libdl
-
-# Add target alias to "gyp_all_modules" target.
-.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
-
-# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_neon.S b/modules/audio_coding/codecs/isac/fix/source/transform_neon.S
index 46682ac..6713b28 100644
--- a/modules/audio_coding/codecs/isac/fix/source/transform_neon.S
+++ b/modules/audio_coding/codecs/isac/fix/source/transform_neon.S
@@ -42,7 +42,11 @@
add r5, sp, #(16 + FRAMESAMPLES * 2) @ tmpimQ16;
adr r9, WebRtcIsacfix_kCosTab1
+#if defined(__APPLE__)
+ mov r6, #:lower16:(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
+#else
mov r6, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
+#endif
add r10, r9, r6 @ WebRtcIsacfix_kSinTab1
vmov.u32 q14, #0 @ Initialize the maximum values for tmpInIm.
@@ -455,7 +459,12 @@
bgt TransformAndFindMax
adr r10, WebRtcIsacfix_kSinTab1
+#if defined(__APPLE__)
+ mov r2, #:lower16:(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
+#else
mov r2, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
+#endif
+
sub r11, r10, r2 @ WebRtcIsacfix_kCosTab1
@ Find the maximum value in the Neon registers
diff --git a/modules/audio_coding/main/OWNERS b/modules/audio_coding/main/OWNERS
index e1e6256..83880d2 100644
--- a/modules/audio_coding/main/OWNERS
+++ b/modules/audio_coding/main/OWNERS
@@ -1,3 +1,4 @@
tina.legrand@webrtc.org
turaj@webrtc.org
jan.skoglund@webrtc.org
+henrik.lundin@webrtc.org
diff --git a/modules/audio_coding/main/acm2/acm_receiver.cc b/modules/audio_coding/main/acm2/acm_receiver.cc
index 65d95b1..5e51ba8 100644
--- a/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -117,23 +117,23 @@
} // namespace
-AcmReceiver::AcmReceiver(Clock* clock)
- : id_(0),
- neteq_config_(),
- neteq_(NetEq::Create(neteq_config_)),
+AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
+ : id_(config.id),
+ neteq_(NetEq::Create(config.neteq_config)),
last_audio_decoder_(-1), // Invalid value.
decode_lock_(RWLockWrapper::CreateRWLock()),
neteq_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
vad_enabled_(true),
previous_audio_activity_(AudioFrame::kVadPassive),
- current_sample_rate_hz_(neteq_config_.sample_rate_hz),
+ current_sample_rate_hz_(config.neteq_config.sample_rate_hz),
nack_(),
nack_enabled_(false),
- clock_(clock),
+ clock_(config.clock),
av_sync_(false),
initial_delay_manager_(),
missing_packets_sync_stream_(),
late_packets_sync_stream_() {
+ assert(clock_);
for (int n = 0; n < ACMCodecDB::kMaxNumCodecs; ++n) {
decoders_[n].registered = false;
}
@@ -428,9 +428,13 @@
if (ptr_audio_buffer == audio_buffer_) {
// Data is written to local buffer.
if (need_resampling) {
- samples_per_channel = resampler_.Resample10Msec(
- audio_buffer_, current_sample_rate_hz_, desired_freq_hz,
- num_channels, audio_frame->data_);
+ samples_per_channel =
+ resampler_.Resample10Msec(audio_buffer_,
+ current_sample_rate_hz_,
+ desired_freq_hz,
+ num_channels,
+ AudioFrame::kMaxDataSizeSamples,
+ audio_frame->data_);
if (samples_per_channel < 0) {
LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed.";
return -1;
@@ -444,9 +448,13 @@
// Data is written into |audio_frame|.
if (need_resampling) {
// We might end up here ONLY if codec is changed.
- samples_per_channel = resampler_.Resample10Msec(
- audio_frame->data_, current_sample_rate_hz_, desired_freq_hz,
- num_channels, audio_buffer_);
+ samples_per_channel =
+ resampler_.Resample10Msec(audio_frame->data_,
+ current_sample_rate_hz_,
+ desired_freq_hz,
+ num_channels,
+ AudioFrame::kMaxDataSizeSamples,
+ audio_buffer_);
if (samples_per_channel < 0) {
LOG_FERR0(LS_ERROR, "AcmReceiver::GetAudio") << "Resampler Failed.";
return -1;
@@ -757,13 +765,9 @@
// exceeds a threshold.
int num_packets;
int max_num_packets;
- int buffer_size_byte;
- int max_buffer_size_byte;
const float kBufferingThresholdScale = 0.9f;
- neteq_->PacketBufferStatistics(&num_packets, &max_num_packets,
- &buffer_size_byte, &max_buffer_size_byte);
- if (num_packets > max_num_packets * kBufferingThresholdScale ||
- buffer_size_byte > max_buffer_size_byte * kBufferingThresholdScale) {
+ neteq_->PacketBufferStatistics(&num_packets, &max_num_packets);
+ if (num_packets > max_num_packets * kBufferingThresholdScale) {
initial_delay_manager_->DisableBuffering();
return false;
}
@@ -776,7 +780,6 @@
current_sample_rate_hz_ = ACMCodecDB::database_[last_audio_decoder_].plfreq;
frame->num_channels_ = decoders_[last_audio_decoder_].channels;
} else {
- current_sample_rate_hz_ = neteq_config_.sample_rate_hz;
frame->num_channels_ = 1;
}
diff --git a/modules/audio_coding/main/acm2/acm_receiver.h b/modules/audio_coding/main/acm2/acm_receiver.h
index fe44de2..7a238ae 100644
--- a/modules/audio_coding/main/acm2/acm_receiver.h
+++ b/modules/audio_coding/main/acm2/acm_receiver.h
@@ -27,7 +27,6 @@
namespace webrtc {
-class Clock;
struct CodecInst;
class CriticalSectionWrapper;
class RWLockWrapper;
@@ -48,7 +47,7 @@
};
// Constructor of the class
- explicit AcmReceiver(Clock* clock);
+ explicit AcmReceiver(const AudioCodingModule::Config& config);
// Destructor of the class.
~AcmReceiver();
@@ -339,7 +338,6 @@
void InsertStreamOfSyncPackets(InitialDelayManager::SyncStream* sync_stream);
int id_;
- NetEq::Config neteq_config_;
NetEq* neteq_;
Decoder decoders_[ACMCodecDB::kMaxNumCodecs];
int last_audio_decoder_;
diff --git a/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
index 8158860..c55eef9 100644
--- a/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ b/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
@@ -43,12 +43,14 @@
public ::testing::Test {
protected:
AcmReceiverTest()
- : receiver_(new AcmReceiver(Clock::GetRealTimeClock())),
- acm_(new AudioCodingModuleImpl(0, Clock::GetRealTimeClock())),
- timestamp_(0),
+ : timestamp_(0),
packet_sent_(false),
last_packet_send_timestamp_(timestamp_),
- last_frame_type_(kFrameEmpty) {}
+ last_frame_type_(kFrameEmpty) {
+ AudioCodingModule::Config config;
+ acm_.reset(new AudioCodingModuleImpl(config));
+ receiver_.reset(new AcmReceiver(config));
+ }
~AcmReceiverTest() {}
diff --git a/modules/audio_coding/main/acm2/acm_resampler.cc b/modules/audio_coding/main/acm2/acm_resampler.cc
index 2940922..97d87b1 100644
--- a/modules/audio_coding/main/acm2/acm_resampler.cc
+++ b/modules/audio_coding/main/acm2/acm_resampler.cc
@@ -10,6 +10,7 @@
#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
+#include <assert.h>
#include <string.h>
#include "webrtc/common_audio/resampler/include/resampler.h"
@@ -28,10 +29,15 @@
int in_freq_hz,
int out_freq_hz,
int num_audio_channels,
+ int out_capacity_samples,
int16_t* out_audio) {
int in_length = in_freq_hz * num_audio_channels / 100;
int out_length = out_freq_hz * num_audio_channels / 100;
if (in_freq_hz == out_freq_hz) {
+ if (out_capacity_samples < in_length) {
+ assert(false);
+ return -1;
+ }
memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
return in_length / num_audio_channels;
}
@@ -43,9 +49,15 @@
return -1;
}
- out_length = resampler_.Resample(in_audio, in_length, out_audio, out_length);
+ out_length =
+ resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
if (out_length == -1) {
- LOG_FERR4(LS_ERROR, Resample, in_audio, in_length, out_audio, out_length);
+ LOG_FERR4(LS_ERROR,
+ Resample,
+ in_audio,
+ in_length,
+ out_audio,
+ out_capacity_samples);
return -1;
}
diff --git a/modules/audio_coding/main/acm2/acm_resampler.h b/modules/audio_coding/main/acm2/acm_resampler.h
index 644a32f..a8fc6b6 100644
--- a/modules/audio_coding/main/acm2/acm_resampler.h
+++ b/modules/audio_coding/main/acm2/acm_resampler.h
@@ -26,6 +26,7 @@
int in_freq_hz,
int out_freq_hz,
int num_audio_channels,
+ int out_capacity_samples,
int16_t* out_audio);
private:
diff --git a/modules/audio_coding/main/acm2/audio_coding_module.cc b/modules/audio_coding/main/acm2/audio_coding_module.cc
index 1fbeae2..eca909c 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/main/acm2/audio_coding_module.cc
@@ -13,22 +13,21 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
#include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
-#include "webrtc/modules/audio_coding/main/source/audio_coding_module_impl.h"
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/trace.h"
namespace webrtc {
-const char kLegacyAcmVersion[] = "acm1";
-const char kExperimentalAcmVersion[] = "acm2";
-
// Create module
AudioCodingModule* AudioCodingModule::Create(int id) {
- return new acm2::AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
+ return Create(id, Clock::GetRealTimeClock());
}
AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
- return new acm2::AudioCodingModuleImpl(id, clock);
+ AudioCodingModule::Config config;
+ config.id = id;
+ config.clock = clock;
+ return new acm2::AudioCodingModuleImpl(config);
}
// Get number of supported codecs
@@ -95,13 +94,4 @@
}
}
-AudioCodingModule* AudioCodingModuleFactory::Create(int id) const {
- return new acm1::AudioCodingModuleImpl(static_cast<int32_t>(id),
- Clock::GetRealTimeClock());
-}
-
-AudioCodingModule* NewAudioCodingModuleFactory::Create(int id) const {
- return new acm2::AudioCodingModuleImpl(id, Clock::GetRealTimeClock());
-}
-
} // namespace webrtc
diff --git a/modules/audio_coding/main/acm2/audio_coding_module.gypi b/modules/audio_coding/main/acm2/audio_coding_module.gypi
index f51c3bf..07fe727 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module.gypi
+++ b/modules/audio_coding/main/acm2/audio_coding_module.gypi
@@ -7,9 +7,29 @@
# be found in the AUTHORS file in the root of the source tree.
{
+ 'variables': {
+ 'audio_coding_dependencies': [
+ 'CNG',
+ 'G711',
+ 'G722',
+ 'iLBC',
+ 'iSAC',
+ 'iSACFix',
+ 'PCM16B',
+ '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ 'audio_coding_defines': [],
+ 'conditions': [
+ ['include_opus==1', {
+ 'audio_coding_dependencies': ['webrtc_opus',],
+ 'audio_coding_defines': ['WEBRTC_CODEC_OPUS',],
+ }],
+ ],
+ },
'targets': [
{
- 'target_name': 'acm2',
+ 'target_name': 'audio_coding_module',
'type': 'static_library',
'defines': [
'<@(audio_coding_defines)',
@@ -93,4 +113,43 @@
],
},
],
+ 'conditions': [
+ ['include_tests==1', {
+ 'targets': [
+ {
+ 'target_name': 'delay_test',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_coding_module',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(webrtc_root)/test/test.gyp:test_support',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ ],
+ 'sources': [
+ '../test/delay_test.cc',
+ '../test/Channel.cc',
+ '../test/PCMFile.cc',
+ '../test/utility.cc',
+ ],
+ }, # delay_test
+ {
+ 'target_name': 'insert_packet_with_timing',
+ 'type': 'executable',
+ 'dependencies': [
+ 'audio_coding_module',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(webrtc_root)/test/test.gyp:test_support',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ ],
+ 'sources': [
+ '../test/insert_packet_with_timing.cc',
+ '../test/Channel.cc',
+ '../test/PCMFile.cc',
+ ],
+ }, # delay_test
+ ],
+ }],
+ ],
}
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index 26a4d30..6133d23 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -114,9 +114,10 @@
} // namespace
-AudioCodingModuleImpl::AudioCodingModuleImpl(int id, Clock* clock)
+AudioCodingModuleImpl::AudioCodingModuleImpl(
+ const AudioCodingModule::Config& config)
: packetization_callback_(NULL),
- id_(id),
+ id_(config.id),
expected_codec_ts_(0xD87F3F9F),
expected_in_ts_(0xD87F3F9F),
send_codec_inst_(),
@@ -131,7 +132,7 @@
stereo_send_(false),
current_send_codec_idx_(-1),
send_codec_registered_(false),
- receiver_(clock),
+ receiver_(config),
acm_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
vad_callback_(NULL),
is_first_red_(true),
@@ -160,8 +161,6 @@
mirror_codec_idx_[i] = -1;
}
- receiver_.set_id(id_);
-
// Allocate memory for RED.
red_buffer_ = new uint8_t[MAX_PAYLOAD_SIZE_BYTE];
@@ -202,7 +201,7 @@
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
"Cannot initialize receiver");
}
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id, "Created");
+ WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id_, "Created");
}
AudioCodingModuleImpl::~AudioCodingModuleImpl() {
@@ -1362,9 +1361,13 @@
// The result of the resampler is written to output frame.
dest_ptr_audio = preprocess_frame_.data_;
- preprocess_frame_.samples_per_channel_ = resampler_.Resample10Msec(
- src_ptr_audio, in_frame.sample_rate_hz_, send_codec_inst_.plfreq,
- preprocess_frame_.num_channels_, dest_ptr_audio);
+ preprocess_frame_.samples_per_channel_ =
+ resampler_.Resample10Msec(src_ptr_audio,
+ in_frame.sample_rate_hz_,
+ send_codec_inst_.plfreq,
+ preprocess_frame_.num_channels_,
+ AudioFrame::kMaxDataSizeSamples,
+ dest_ptr_audio);
if (preprocess_frame_.samples_per_channel_ < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, id_,
@@ -1707,7 +1710,9 @@
}
audio_frame->id_ = id_;
- audio_frame->energy_ = 0;
+ // The energy must be -1 in order to have the energy calculated later on in
+ // the AudioConferenceMixer module.
+ audio_frame->energy_ = static_cast<uint32_t>(-1);
audio_frame->timestamp_ = 0;
return 0;
}
@@ -1973,10 +1978,6 @@
return receiver_.LeastRequiredDelayMs();
}
-const char* AudioCodingModuleImpl::Version() const {
- return kExperimentalAcmVersion;
-}
-
void AudioCodingModuleImpl::GetDecodingCallStatistics(
AudioDecodingCallStats* call_stats) const {
receiver_.GetDecodingCallStatistics(call_stats);
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_impl.h b/modules/audio_coding/main/acm2/audio_coding_module_impl.h
index cdee3f9..157fc01 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_impl.h
+++ b/modules/audio_coding/main/acm2/audio_coding_module_impl.h
@@ -22,7 +22,6 @@
namespace webrtc {
-class Clock;
class CriticalSectionWrapper;
class RWLockWrapper;
@@ -33,11 +32,9 @@
class AudioCodingModuleImpl : public AudioCodingModule {
public:
- AudioCodingModuleImpl(int id, Clock* clock);
+ explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
~AudioCodingModuleImpl();
- virtual const char* Version() const;
-
// Change the unique identifier of this object.
virtual int32_t ChangeUniqueId(const int32_t id);
diff --git a/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
index 0f3e73e..c745048 100644
--- a/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
+++ b/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
@@ -58,8 +58,9 @@
class AudioCodingModuleTest : public ::testing::Test {
protected:
AudioCodingModuleTest()
- : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
- acm2_(AudioCodingModule::Create(1 /*id*/)) {}
+ : id_(1),
+ rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
+ acm_(AudioCodingModule::Create(id_)) {}
~AudioCodingModuleTest() {}
@@ -71,7 +72,7 @@
codec.pltype = kPayloadType;
// Register L16 codec in ACMs.
- ASSERT_EQ(0, acm2_->RegisterReceiveCodec(codec));
+ ASSERT_EQ(0, acm_->RegisterReceiveCodec(codec));
rtp_utility_->Populate(&rtp_header_);
}
@@ -81,19 +82,20 @@
const uint8_t kPayload[kPayloadSizeBytes] = {0};
ASSERT_EQ(0,
- acm2_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_));
+ acm_->IncomingPacket(kPayload, kPayloadSizeBytes, rtp_header_));
- ASSERT_EQ(0, acm2_->PlayoutData10Ms(-1, &audio_frame));
+ ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame));
rtp_utility_->Forward(&rtp_header_);
}
void JustPullAudio() {
AudioFrame audio_frame;
- ASSERT_EQ(0, acm2_->PlayoutData10Ms(-1, &audio_frame));
+ ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &audio_frame));
}
+ const int id_;
scoped_ptr<RtpUtility> rtp_utility_;
- scoped_ptr<AudioCodingModule> acm2_;
+ scoped_ptr<AudioCodingModule> acm_;
WebRtcRTPHeader rtp_header_;
};
@@ -102,7 +104,7 @@
// all fields have to be zero.
TEST_F(AudioCodingModuleTest, DISABLED_ON_ANDROID(InitializedToZero)) {
AudioDecodingCallStats stats;
- acm2_->GetDecodingCallStatistics(&stats);
+ acm_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
@@ -117,15 +119,14 @@
AudioDecodingCallStats stats;
const int kInitialDelay = 100;
- acm2_->SetInitialPlayoutDelay(kInitialDelay);
+ acm_->SetInitialPlayoutDelay(kInitialDelay);
- AudioFrame audio_frame;
int num_calls = 0;
for (int time_ms = 0; time_ms < kInitialDelay;
time_ms += kFrameSizeMs, ++num_calls) {
InsertPacketAndPullAudio();
}
- acm2_->GetDecodingCallStatistics(&stats);
+ acm_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(0, stats.calls_to_neteq);
EXPECT_EQ(num_calls, stats.calls_to_silence_generator);
EXPECT_EQ(0, stats.decoded_normal);
@@ -141,11 +142,10 @@
AudioDecodingCallStats stats;
const int kNumNormalCalls = 10;
- AudioFrame audio_frame;
for (int num_calls = 0; num_calls < kNumNormalCalls; ++num_calls) {
InsertPacketAndPullAudio();
}
- acm2_->GetDecodingCallStatistics(&stats);
+ acm_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(kNumNormalCalls, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
@@ -160,7 +160,7 @@
for (int n = 0; n < kNumPlc + kNumPlcCng; ++n) {
JustPullAudio();
}
- acm2_->GetDecodingCallStatistics(&stats);
+ acm_->GetDecodingCallStatistics(&stats);
EXPECT_EQ(kNumNormalCalls + kNumPlc + kNumPlcCng, stats.calls_to_neteq);
EXPECT_EQ(0, stats.calls_to_silence_generator);
EXPECT_EQ(kNumNormalCalls, stats.decoded_normal);
@@ -169,4 +169,23 @@
EXPECT_EQ(kNumPlcCng, stats.decoded_plc_cng);
}
+TEST_F(AudioCodingModuleTest, VerifyOutputFrame) {
+ AudioFrame audio_frame;
+ const int kSampleRateHz = 32000;
+ EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame));
+ // The energy must be -1 in order to have the energy calculated later on in
+ // the AudioConferenceMixer module.
+ EXPECT_EQ(static_cast<uint32_t>(-1), audio_frame.energy_);
+ EXPECT_EQ(id_, audio_frame.id_);
+ EXPECT_EQ(0u, audio_frame.timestamp_);
+ EXPECT_GT(audio_frame.num_channels_, 0);
+ EXPECT_EQ(kSampleRateHz / 100, audio_frame.samples_per_channel_);
+ EXPECT_EQ(kSampleRateHz, audio_frame.sample_rate_hz_);
+}
+
+TEST_F(AudioCodingModuleTest, FailOnZeroDesiredFrequency) {
+ AudioFrame audio_frame;
+ EXPECT_EQ(-1, acm_->PlayoutData10Ms(0, &audio_frame));
+}
+
} // namespace webrtc
diff --git a/modules/audio_coding/main/acm2/nack_unittest.cc b/modules/audio_coding/main/acm2/nack_unittest.cc
index 8011d88..5837c31 100644
--- a/modules/audio_coding/main/acm2/nack_unittest.cc
+++ b/modules/audio_coding/main/acm2/nack_unittest.cc
@@ -398,7 +398,7 @@
// Packet lost more than NACK-list size limit.
uint16_t num_lost_packets = kNackThreshold + kNackListSize + 5;
- scoped_array<uint16_t> seq_num_lost(new uint16_t[num_lost_packets]);
+ scoped_ptr<uint16_t[]> seq_num_lost(new uint16_t[num_lost_packets]);
for (int n = 0; n < num_lost_packets; ++n) {
seq_num_lost[n] = ++seq_num;
}
diff --git a/modules/audio_coding/main/interface/audio_coding_module.h b/modules/audio_coding/main/interface/audio_coding_module.h
index db45add..9c7e562 100644
--- a/modules/audio_coding/main/interface/audio_coding_module.h
+++ b/modules/audio_coding/main/interface/audio_coding_module.h
@@ -15,7 +15,9 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
+#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
#include "webrtc/modules/interface/module.h"
+#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@@ -25,7 +27,6 @@
struct WebRtcRTPHeader;
class AudioFrame;
class RTPFragmentationHeader;
-class Clock;
#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
@@ -73,15 +74,22 @@
const uint16_t delayMS) = 0; // average delay in ms
};
-// Version string for testing, to distinguish instances of ACM1 from ACM2.
-extern const char kLegacyAcmVersion[];
-extern const char kExperimentalAcmVersion[];
-
class AudioCodingModule: public Module {
protected:
AudioCodingModule() {}
public:
+ struct Config {
+ Config()
+ : id(0),
+ neteq_config(),
+ clock(Clock::GetRealTimeClock()) {}
+
+ int id;
+ NetEq::Config neteq_config;
+ Clock* clock;
+ };
+
///////////////////////////////////////////////////////////////////////////
// Creation and destruction of a ACM.
//
@@ -178,11 +186,6 @@
//
static bool IsCodecValid(const CodecInst& codec);
- // Returns the version of ACM. This facilitates distinguishing instances of
- // ACM1 from ACM2 while testing. This API will be removed when ACM1 is
- // completely removed.
- virtual const char* Version() const = 0;
-
///////////////////////////////////////////////////////////////////////////
// Sender
//
@@ -936,20 +939,6 @@
AudioDecodingCallStats* call_stats) const = 0;
};
-struct AudioCodingModuleFactory {
- AudioCodingModuleFactory() {}
- virtual ~AudioCodingModuleFactory() {}
-
- virtual AudioCodingModule* Create(int id) const;
-};
-
-struct NewAudioCodingModuleFactory : AudioCodingModuleFactory {
- NewAudioCodingModuleFactory() {}
- virtual ~NewAudioCodingModuleFactory() {}
-
- virtual AudioCodingModule* Create(int id) const;
-};
-
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
diff --git a/modules/audio_coding/main/source/audio_coding_module_impl.cc b/modules/audio_coding/main/source/audio_coding_module_impl.cc
index b8060ce..5bdbd01 100644
--- a/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ b/modules/audio_coding/main/source/audio_coding_module_impl.cc
@@ -2250,18 +2250,10 @@
int decoded_seq_num;
uint32_t decoded_timestamp;
bool update_nack =
- neteq_.DecodedRtpInfo(&decoded_seq_num, &decoded_timestamp) &&
- nack_enabled_; // Update NACK only if it is enabled.
- audio_frame->num_channels_ = audio_frame_.num_channels_;
- audio_frame->vad_activity_ = audio_frame_.vad_activity_;
- audio_frame->speech_type_ = audio_frame_.speech_type_;
+ neteq_.DecodedRtpInfo(&decoded_seq_num, &decoded_timestamp);
- stereo_mode = (audio_frame_.num_channels_ > 1);
-
- // For stereo playout:
// Master and Slave samples are interleaved starting with Master.
- const uint16_t receive_freq =
- static_cast<uint16_t>(audio_frame_.sample_rate_hz_);
+ uint16_t receive_freq;
bool tone_detected = false;
int16_t last_detected_tone;
int16_t tone;
@@ -2270,10 +2262,17 @@
{
CriticalSectionScoped lock(acm_crit_sect_);
+ audio_frame->num_channels_ = audio_frame_.num_channels_;
+ audio_frame->vad_activity_ = audio_frame_.vad_activity_;
+ audio_frame->speech_type_ = audio_frame_.speech_type_;
+
+ stereo_mode = (audio_frame_.num_channels_ > 1);
+
+ receive_freq = static_cast<uint16_t>(audio_frame_.sample_rate_hz_);
// Update call statistics.
call_stats_.DecodedByNetEq(audio_frame->speech_type_);
- if (update_nack) {
+ if (nack_enabled_ && update_nack) {
assert(nack_.get());
nack_->UpdateLastDecodedPacket(decoded_seq_num, decoded_timestamp);
}
@@ -3029,10 +3028,6 @@
nack_enabled_ = false;
}
-const char* AudioCodingModuleImpl::Version() const {
- return kLegacyAcmVersion;
-}
-
void AudioCodingModuleImpl::GetDecodingCallStatistics(
AudioDecodingCallStats* call_stats) const {
CriticalSectionScoped lock(acm_crit_sect_);
diff --git a/modules/audio_coding/main/source/audio_coding_module_impl.h b/modules/audio_coding/main/source/audio_coding_module_impl.h
index f0b22f1..b99e5ac 100644
--- a/modules/audio_coding/main/source/audio_coding_module_impl.h
+++ b/modules/audio_coding/main/source/audio_coding_module_impl.h
@@ -44,8 +44,6 @@
AudioCodingModuleImpl(const int32_t id, Clock* clock);
~AudioCodingModuleImpl();
- virtual const char* Version() const;
-
// Change the unique identifier of this object.
virtual int32_t ChangeUniqueId(const int32_t id);
diff --git a/modules/audio_coding/main/test/opus_test.cc b/modules/audio_coding/main/test/opus_test.cc
index 230e9f1..261eb61 100644
--- a/modules/audio_coding/main/test/opus_test.cc
+++ b/modules/audio_coding/main/test/opus_test.cc
@@ -211,8 +211,9 @@
int frame_length, int percent_loss) {
AudioFrame audio_frame;
int32_t out_freq_hz_b = out_file_.SamplingFrequency();
- int16_t audio[480 * 12 * 2]; // Can hold 120 ms stereo audio.
- int16_t out_audio[480 * 12 * 2]; // Can hold 120 ms stereo audio.
+ const int kBufferSizeSamples = 480 * 12 * 2; // Can hold 120 ms stereo audio.
+ int16_t audio[kBufferSizeSamples];
+ int16_t out_audio[kBufferSizeSamples];
int16_t audio_type;
int written_samples = 0;
int read_samples = 0;
@@ -257,6 +258,7 @@
audio_frame.sample_rate_hz_,
48000,
channels,
+ kBufferSizeSamples - written_samples,
&audio[written_samples]));
written_samples += 480 * channels;
diff --git a/modules/audio_coding/neteq4/background_noise.h b/modules/audio_coding/neteq4/background_noise.h
index ac5446b..141d71d 100644
--- a/modules/audio_coding/neteq4/background_noise.h
+++ b/modules/audio_coding/neteq4/background_noise.h
@@ -126,7 +126,7 @@
int32_t residual_energy);
size_t num_channels_;
- scoped_array<ChannelParameters> channel_parameters_;
+ scoped_ptr<ChannelParameters[]> channel_parameters_;
bool initialized_;
NetEqBackgroundNoiseMode mode_;
diff --git a/modules/audio_coding/neteq4/decision_logic_unittest.cc b/modules/audio_coding/neteq4/decision_logic_unittest.cc
index d596c05..60a4b62 100644
--- a/modules/audio_coding/neteq4/decision_logic_unittest.cc
+++ b/modules/audio_coding/neteq4/decision_logic_unittest.cc
@@ -24,7 +24,7 @@
int fs_hz = 8000;
int output_size_samples = fs_hz / 100; // Samples per 10 ms.
DecoderDatabase decoder_database;
- PacketBuffer packet_buffer(10, 1000);
+ PacketBuffer packet_buffer(10);
DelayPeakDetector delay_peak_detector;
DelayManager delay_manager(240, &delay_peak_detector);
BufferLevelFilter buffer_level_filter;
diff --git a/modules/audio_coding/neteq4/expand.h b/modules/audio_coding/neteq4/expand.h
index aec1cd9..6962aa0 100644
--- a/modules/audio_coding/neteq4/expand.h
+++ b/modules/audio_coding/neteq4/expand.h
@@ -167,7 +167,7 @@
int lag_index_direction_;
int current_lag_index_;
bool stop_muting_;
- scoped_array<ChannelParameters> channel_parameters_;
+ scoped_ptr<ChannelParameters[]> channel_parameters_;
DISALLOW_COPY_AND_ASSIGN(Expand);
};
diff --git a/modules/audio_coding/neteq4/interface/neteq.h b/modules/audio_coding/neteq4/interface/neteq.h
index 8defe4f..64a23e8 100644
--- a/modules/audio_coding/neteq4/interface/neteq.h
+++ b/modules/audio_coding/neteq4/interface/neteq.h
@@ -70,10 +70,15 @@
struct Config {
Config()
: sample_rate_hz(16000),
- enable_audio_classifier(false) {}
+ enable_audio_classifier(false),
+ max_packets_in_buffer(50),
+ // |max_delay_ms| has the same effect as calling SetMaximumDelay().
+ max_delay_ms(2000) {}
int sample_rate_hz; // Initial vale. Will change with input data.
bool enable_audio_classifier;
+ int max_packets_in_buffer;
+ int max_delay_ms;
};
enum ReturnCodes {
@@ -107,13 +112,9 @@
kFrameSplitError,
kRedundancySplitError,
kPacketBufferCorruption,
- kOversizePacket,
kSyncPacketNotAccepted
};
- static const int kMaxNumPacketsInBuffer = 50; // TODO(hlundin): Remove.
- static const int kMaxBytesInBuffer = 113280; // TODO(hlundin): Remove.
-
// Creates a new NetEq object, with parameters set in |config|. The |config|
// object will only have to be valid for the duration of the call to this
// method.
@@ -179,7 +180,8 @@
// Sets a maximum delay in milliseconds for packet buffer. The latency will
// not exceed the given value, even required delay (given the channel
- // conditions) is higher.
+ // conditions) is higher. Calling this method has the same effect as setting
+ // the |max_delay_ms| value in the NetEq::Config struct.
virtual bool SetMaximumDelay(int delay_ms) = 0;
// The smallest latency required. This is computed bases on inter-arrival
@@ -249,9 +251,7 @@
// Current usage of packet-buffer and it's limits.
virtual void PacketBufferStatistics(int* current_num_packets,
- int* max_num_packets,
- int* current_memory_size_bytes,
- int* max_memory_size_bytes) const = 0;
+ int* max_num_packets) const = 0;
// Get sequence number and timestamp of the latest RTP.
// This method is to facilitate NACK.
diff --git a/modules/audio_coding/neteq4/mock/mock_packet_buffer.h b/modules/audio_coding/neteq4/mock/mock_packet_buffer.h
index 37fa90d..b1d8ef9 100644
--- a/modules/audio_coding/neteq4/mock/mock_packet_buffer.h
+++ b/modules/audio_coding/neteq4/mock/mock_packet_buffer.h
@@ -19,8 +19,8 @@
class MockPacketBuffer : public PacketBuffer {
public:
- MockPacketBuffer(size_t max_number_of_packets, size_t max_payload_memory)
- : PacketBuffer(max_number_of_packets, max_payload_memory) {}
+ MockPacketBuffer(size_t max_number_of_packets)
+ : PacketBuffer(max_number_of_packets) {}
virtual ~MockPacketBuffer() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_METHOD0(Flush,
diff --git a/modules/audio_coding/neteq4/neteq.cc b/modules/audio_coding/neteq4/neteq.cc
index bfcd86a..48dacea 100644
--- a/modules/audio_coding/neteq4/neteq.cc
+++ b/modules/audio_coding/neteq4/neteq.cc
@@ -32,12 +32,12 @@
BufferLevelFilter* buffer_level_filter = new BufferLevelFilter;
DecoderDatabase* decoder_database = new DecoderDatabase;
DelayPeakDetector* delay_peak_detector = new DelayPeakDetector;
- DelayManager* delay_manager = new DelayManager(kMaxNumPacketsInBuffer,
- delay_peak_detector);
+ DelayManager* delay_manager =
+ new DelayManager(config.max_packets_in_buffer, delay_peak_detector);
+ delay_manager->SetMaximumDelay(config.max_delay_ms);
DtmfBuffer* dtmf_buffer = new DtmfBuffer(config.sample_rate_hz);
DtmfToneGenerator* dtmf_tone_generator = new DtmfToneGenerator;
- PacketBuffer* packet_buffer = new PacketBuffer(kMaxNumPacketsInBuffer,
- kMaxBytesInBuffer);
+ PacketBuffer* packet_buffer = new PacketBuffer(config.max_packets_in_buffer);
PayloadSplitter* payload_splitter = new PayloadSplitter;
TimestampScaler* timestamp_scaler = new TimestampScaler(*decoder_database);
AccelerateFactory* accelerate_factory = new AccelerateFactory;
diff --git a/modules/audio_coding/neteq4/neteq.gypi b/modules/audio_coding/neteq4/neteq.gypi
index b041694..bd3b251 100644
--- a/modules/audio_coding/neteq4/neteq.gypi
+++ b/modules/audio_coding/neteq4/neteq.gypi
@@ -168,9 +168,6 @@
'target_name': 'neteq_unittest_tools',
'type': 'static_library',
'dependencies': [
- '<(DEPTH)/testing/gmock.gyp:gmock',
- '<(DEPTH)/testing/gtest.gyp:gtest',
- 'PCM16B', # Needed by neteq_performance_test.
'rtp_rtcp',
],
'direct_dependent_settings': {
@@ -186,8 +183,6 @@
'tools/audio_loop.h',
'tools/input_audio_file.cc',
'tools/input_audio_file.h',
- 'tools/neteq_performance_test.cc',
- 'tools/neteq_performance_test.h',
'tools/packet.cc',
'tools/packet.h',
'tools/packet_source.h',
@@ -195,8 +190,6 @@
'tools/rtp_file_source.h',
'tools/rtp_generator.cc',
'tools/rtp_generator.h',
- 'tools/neteq_quality_test.cc',
- 'tools/neteq_quality_test.h',
],
}, # neteq_unittest_tools
], # targets
diff --git a/modules/audio_coding/neteq4/neteq_impl.cc b/modules/audio_coding/neteq4/neteq_impl.cc
index dcf48ad..38a5456 100644
--- a/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/modules/audio_coding/neteq4/neteq_impl.cc
@@ -77,7 +77,6 @@
accelerate_factory_(accelerate_factory),
preemptive_expand_factory_(preemptive_expand_factory),
last_mode_(kModeNormal),
- mute_factor_array_(NULL),
decoded_buffer_length_(kMaxFrameSize),
decoded_buffer_(new int16_t[decoded_buffer_length_]),
playout_timestamp_(0),
@@ -365,12 +364,9 @@
}
void NetEqImpl::PacketBufferStatistics(int* current_num_packets,
- int* max_num_packets,
- int* current_memory_size_bytes,
- int* max_memory_size_bytes) const {
+ int* max_num_packets) const {
CriticalSectionScoped lock(crit_sect_.get());
- packet_buffer_->BufferStat(current_num_packets, max_num_packets,
- current_memory_size_bytes, max_memory_size_bytes);
+ packet_buffer_->BufferStat(current_num_packets, max_num_packets);
}
int NetEqImpl::DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const {
@@ -611,9 +607,6 @@
new_codec_ = true;
update_sample_rate_and_channels = true;
LOG_F(LS_WARNING) << "Packet buffer flushed";
- } else if (ret == PacketBuffer::kOversizePacket) {
- LOG_F(LS_WARNING) << "Packet larger than packet buffer";
- return kOversizePacket;
} else if (ret != PacketBuffer::kOK) {
LOG_FERR1(LS_WARNING, InsertPacketList, packet_list.size());
PacketBuffer::DeleteAllPackets(&packet_list);
diff --git a/modules/audio_coding/neteq4/neteq_impl.h b/modules/audio_coding/neteq4/neteq_impl.h
index 09a0049..5801d97 100644
--- a/modules/audio_coding/neteq4/neteq_impl.h
+++ b/modules/audio_coding/neteq4/neteq_impl.h
@@ -186,9 +186,7 @@
virtual void FlushBuffers();
virtual void PacketBufferStatistics(int* current_num_packets,
- int* max_num_packets,
- int* current_memory_size_bytes,
- int* max_memory_size_bytes) const;
+ int* max_num_packets) const;
// Get sequence number and timestamp of the latest RTP.
// This method is to facilitate NACK.
@@ -371,9 +369,9 @@
int output_size_samples_ GUARDED_BY(crit_sect_);
int decoder_frame_length_ GUARDED_BY(crit_sect_);
Modes last_mode_ GUARDED_BY(crit_sect_);
- scoped_array<int16_t> mute_factor_array_ GUARDED_BY(crit_sect_);
+ scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
- scoped_array<int16_t> decoded_buffer_ GUARDED_BY(crit_sect_);
+ scoped_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
bool new_codec_ GUARDED_BY(crit_sect_);
uint32_t timestamp_ GUARDED_BY(crit_sect_);
diff --git a/modules/audio_coding/neteq4/neteq_impl_unittest.cc b/modules/audio_coding/neteq4/neteq_impl_unittest.cc
index 15a1c8b..aedd8d5 100644
--- a/modules/audio_coding/neteq4/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq4/neteq_impl_unittest.cc
@@ -102,13 +102,13 @@
delay_peak_detector_ = new DelayPeakDetector;
}
if (use_mock_delay_manager_) {
- mock_delay_manager_ = new MockDelayManager(NetEq::kMaxNumPacketsInBuffer,
+ mock_delay_manager_ = new MockDelayManager(config_.max_packets_in_buffer,
delay_peak_detector_);
EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1);
delay_manager_ = mock_delay_manager_;
} else {
delay_manager_ =
- new DelayManager(NetEq::kMaxNumPacketsInBuffer, delay_peak_detector_);
+ new DelayManager(config_.max_packets_in_buffer, delay_peak_detector_);
}
if (use_mock_dtmf_buffer_) {
mock_dtmf_buffer_ = new MockDtmfBuffer(config_.sample_rate_hz);
@@ -123,12 +123,10 @@
dtmf_tone_generator_ = new DtmfToneGenerator;
}
if (use_mock_packet_buffer_) {
- mock_packet_buffer_ = new MockPacketBuffer(NetEq::kMaxNumPacketsInBuffer,
- NetEq::kMaxBytesInBuffer);
+ mock_packet_buffer_ = new MockPacketBuffer(config_.max_packets_in_buffer);
packet_buffer_ = mock_packet_buffer_;
} else {
- packet_buffer_ = new PacketBuffer(NetEq::kMaxNumPacketsInBuffer,
- NetEq::kMaxBytesInBuffer);
+ packet_buffer_ = new PacketBuffer(config_.max_packets_in_buffer);
}
if (use_mock_payload_splitter_) {
mock_payload_splitter_ = new MockPayloadSplitter;
@@ -381,7 +379,7 @@
neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType));
// Insert packets. The buffer should not flush.
- for (int i = 1; i <= NetEq::kMaxNumPacketsInBuffer; ++i) {
+ for (int i = 1; i <= config_.max_packets_in_buffer; ++i) {
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
diff --git a/modules/audio_coding/neteq4/neteq_tests.gypi b/modules/audio_coding/neteq4/neteq_tests.gypi
index 9751a1c..d19c6b2 100644
--- a/modules/audio_coding/neteq4/neteq_tests.gypi
+++ b/modules/audio_coding/neteq4/neteq_tests.gypi
@@ -154,12 +154,29 @@
},
{
+ 'target_name': 'neteq_test_support',
+ 'type': 'static_library',
+ 'dependencies': [
+ 'NetEq4',
+ 'PCM16B',
+ 'neteq_unittest_tools',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+ ],
+ 'sources': [
+ 'tools/neteq_performance_test.cc',
+ 'tools/neteq_performance_test.h',
+ 'tools/neteq_quality_test.cc',
+ 'tools/neteq_quality_test.h',
+ ],
+ }, # neteq_test_support
+
+ {
'target_name': 'neteq4_speed_test',
'type': 'executable',
'dependencies': [
'NetEq4',
- 'neteq_unittest_tools',
- 'PCM16B',
+ 'neteq_test_support',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/test/test.gyp:test_support_main',
],
@@ -173,7 +190,7 @@
'type': 'executable',
'dependencies': [
'NetEq4',
- 'neteq_unittest_tools',
+ 'neteq_test_support',
'webrtc_opus',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
diff --git a/modules/audio_coding/neteq4/neteq_unittest.cc b/modules/audio_coding/neteq4/neteq_unittest.cc
index 90dc717..93f2567 100644
--- a/modules/audio_coding/neteq4/neteq_unittest.cc
+++ b/modules/audio_coding/neteq4/neteq_unittest.cc
@@ -945,18 +945,6 @@
EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
}
-TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(OversizePacket)) {
- // Payload size is greater than packet buffer size
- const int kPayloadBytes = NetEq::kMaxBytesInBuffer + 1;
- uint8_t payload[kPayloadBytes] = {0};
- WebRtcRTPHeader rtp_info;
- PopulateRtpInfo(0, 0, &rtp_info);
- rtp_info.header.payloadType = 103; // iSAC, no packet splitting.
- EXPECT_EQ(NetEq::kFail,
- neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
- EXPECT_EQ(NetEq::kOversizePacket, neteq_->LastError());
-}
-
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
const int kPayloadBytes = 100;
uint8_t payload[kPayloadBytes] = {0};
diff --git a/modules/audio_coding/neteq4/packet_buffer.cc b/modules/audio_coding/neteq4/packet_buffer.cc
index 0cc0854..bcd0e7b 100644
--- a/modules/audio_coding/neteq4/packet_buffer.cc
+++ b/modules/audio_coding/neteq4/packet_buffer.cc
@@ -36,14 +36,8 @@
const Packet* new_packet_;
};
-// Constructor. The arguments define the maximum number of slots and maximum
-// payload memory (excluding RTP headers) that the buffer will accept.
-PacketBuffer::PacketBuffer(size_t max_number_of_packets,
- size_t max_memory_bytes)
- : max_number_of_packets_(max_number_of_packets),
- max_memory_bytes_(max_memory_bytes),
- current_memory_bytes_(0) {
-}
+PacketBuffer::PacketBuffer(size_t max_number_of_packets)
+ : max_number_of_packets_(max_number_of_packets) {}
// Destructor. All packets in the buffer will be destroyed.
PacketBuffer::~PacketBuffer() {
@@ -53,7 +47,6 @@
// Flush the buffer. All packets in the buffer will be destroyed.
void PacketBuffer::Flush() {
DeleteAllPackets(&buffer_);
- current_memory_bytes_ = 0;
}
int PacketBuffer::InsertPacket(Packet* packet) {
@@ -66,22 +59,10 @@
int return_val = kOK;
- if ((buffer_.size() >= max_number_of_packets_) ||
- (current_memory_bytes_ + packet->payload_length
- > static_cast<int>(max_memory_bytes_))) {
+ if (buffer_.size() >= max_number_of_packets_) {
// Buffer is full. Flush it.
Flush();
return_val = kFlushed;
- if ((buffer_.size() >= max_number_of_packets_) ||
- (current_memory_bytes_ + packet->payload_length
- > static_cast<int>(max_memory_bytes_))) {
- // Buffer is still too small for the packet. Either the buffer limits are
- // really small, or the packet is really large. Delete the packet and
- // return an error.
- delete [] packet->payload;
- delete packet;
- return kOversizePacket;
- }
}
// Get an iterator pointing to the place in the buffer where the new packet
@@ -91,7 +72,6 @@
buffer_.rbegin(), buffer_.rend(),
NewTimestampIsLarger(packet));
buffer_.insert(rit.base(), packet); // Insert the packet at that position.
- current_memory_bytes_ += packet->payload_length;
return return_val;
}
@@ -183,8 +163,6 @@
// Assert that the packet sanity checks in InsertPacket method works.
assert(packet && packet->payload);
buffer_.pop_front();
- current_memory_bytes_ -= packet->payload_length;
- assert(current_memory_bytes_ >= 0); // Assert bookkeeping is correct.
// Discard other packets with the same timestamp. These are duplicates or
// redundant payloads that should not be used.
if (discard_count) {
@@ -206,11 +184,9 @@
if (Empty()) {
return kBufferEmpty;
}
- Packet* temp_packet = buffer_.front();
// Assert that the packet sanity checks in InsertPacket method works.
- assert(temp_packet && temp_packet->payload);
- current_memory_bytes_ -= temp_packet->payload_length;
- assert(current_memory_bytes_ >= 0); // Assert bookkeeping is correct.
+ assert(buffer_.front());
+ assert(buffer_.front()->payload);
DeleteFirstPacket(&buffer_);
return kOK;
}
@@ -280,14 +256,9 @@
}
}
-void PacketBuffer::BufferStat(int* num_packets,
- int* max_num_packets,
- int* current_memory_bytes,
- int* max_memory_bytes) const {
+void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const {
*num_packets = static_cast<int>(buffer_.size());
*max_num_packets = static_cast<int>(max_number_of_packets_);
- *current_memory_bytes = current_memory_bytes_;
- *max_memory_bytes = static_cast<int>(max_memory_bytes_);
}
} // namespace webrtc
diff --git a/modules/audio_coding/neteq4/packet_buffer.h b/modules/audio_coding/neteq4/packet_buffer.h
index e964c28..3d22ad0 100644
--- a/modules/audio_coding/neteq4/packet_buffer.h
+++ b/modules/audio_coding/neteq4/packet_buffer.h
@@ -29,14 +29,12 @@
kNotFound,
kBufferEmpty,
kInvalidPacket,
- kInvalidPointer,
- kOversizePacket
+ kInvalidPointer
};
// Constructor creates a buffer which can hold a maximum of
- // |max_number_of_packets| packets and |max_payload_memory| bytes of payload,
- // excluding RTP headers.
- PacketBuffer(size_t max_number_of_packets, size_t max_payload_memory);
+ // |max_number_of_packets| packets.
+ PacketBuffer(size_t max_number_of_packets);
// Deletes all packets in the buffer before destroying the buffer.
virtual ~PacketBuffer();
@@ -116,12 +114,7 @@
// The default value for |inc| is 1.
virtual void IncrementWaitingTimes(int inc = 1);
- virtual void BufferStat(int* num_packets,
- int* max_num_packets,
- int* current_memory_bytes,
- int* max_memory_bytes) const;
-
- virtual int current_memory_bytes() const { return current_memory_bytes_; }
+ virtual void BufferStat(int* num_packets, int* max_num_packets) const;
// Static method that properly deletes the first packet, and its payload
// array, in |packet_list|. Returns false if |packet_list| already was empty,
@@ -134,8 +127,6 @@
private:
size_t max_number_of_packets_;
- size_t max_memory_bytes_;
- int current_memory_bytes_;
PacketList buffer_;
DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
};
diff --git a/modules/audio_coding/neteq4/packet_buffer_unittest.cc b/modules/audio_coding/neteq4/packet_buffer_unittest.cc
index c8109dc..387ca14 100644
--- a/modules/audio_coding/neteq4/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq4/packet_buffer_unittest.cc
@@ -70,13 +70,13 @@
// Start of test definitions.
TEST(PacketBuffer, CreateAndDestroy) {
- PacketBuffer* buffer = new PacketBuffer(10, 1000); // 10 packets, 1000 bytes.
+ PacketBuffer* buffer = new PacketBuffer(10); // 10 packets.
EXPECT_TRUE(buffer->Empty());
delete buffer;
}
TEST(PacketBuffer, InsertPacket) {
- PacketBuffer buffer(10, 1000); // 10 packets, 1000 bytes.
+ PacketBuffer buffer(10); // 10 packets.
PacketGenerator gen(17u, 4711u, 0, 10);
const int payload_len = 100;
@@ -88,7 +88,6 @@
EXPECT_EQ(4711u, next_ts);
EXPECT_FALSE(buffer.Empty());
EXPECT_EQ(1, buffer.NumPacketsInBuffer());
- EXPECT_EQ(payload_len, buffer.current_memory_bytes());
const RTPHeader* hdr = buffer.NextRtpHeader();
EXPECT_EQ(&(packet->header), hdr); // Compare pointer addresses.
@@ -98,7 +97,7 @@
// Test to flush buffer.
TEST(PacketBuffer, FlushBuffer) {
- PacketBuffer buffer(10, 1000); // 10 packets, 1000 bytes.
+ PacketBuffer buffer(10); // 10 packets.
PacketGenerator gen(0, 0, 0, 10);
const int payload_len = 10;
@@ -109,18 +108,16 @@
}
EXPECT_EQ(10, buffer.NumPacketsInBuffer());
EXPECT_FALSE(buffer.Empty());
- EXPECT_EQ(10 * payload_len, buffer.current_memory_bytes());
buffer.Flush();
// Buffer should delete the payloads itself.
EXPECT_EQ(0, buffer.NumPacketsInBuffer());
EXPECT_TRUE(buffer.Empty());
- EXPECT_EQ(0, buffer.current_memory_bytes());
}
// Test to fill the buffer over the limits, and verify that it flushes.
TEST(PacketBuffer, OverfillBuffer) {
- PacketBuffer buffer(10, 1000); // 10 packets, 1000 bytes.
+ PacketBuffer buffer(10); // 10 packets.
PacketGenerator gen(0, 0, 0, 10);
// Insert 10 small packets; should be ok.
@@ -131,7 +128,6 @@
EXPECT_EQ(PacketBuffer::kOK, buffer.InsertPacket(packet));
}
EXPECT_EQ(10, buffer.NumPacketsInBuffer());
- EXPECT_EQ(10 * payload_len, buffer.current_memory_bytes());
uint32_t next_ts;
EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
EXPECT_EQ(0u, next_ts); // Expect first inserted packet to be first in line.
@@ -140,30 +136,17 @@
Packet* packet = gen.NextPacket(payload_len);
EXPECT_EQ(PacketBuffer::kFlushed, buffer.InsertPacket(packet));
EXPECT_EQ(1, buffer.NumPacketsInBuffer());
- EXPECT_EQ(payload_len, buffer.current_memory_bytes());
EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
// Expect last inserted packet to be first in line.
EXPECT_EQ(packet->header.timestamp, next_ts);
- // Insert 2 large packets; expect to flush when inserting the second one.
- const int large_payload_len = 500;
- packet = gen.NextPacket(large_payload_len);
- EXPECT_EQ(PacketBuffer::kOK, buffer.InsertPacket(packet));
- EXPECT_EQ(2, buffer.NumPacketsInBuffer());
- EXPECT_EQ(payload_len + large_payload_len, buffer.current_memory_bytes());
-
- packet = gen.NextPacket(large_payload_len);
- EXPECT_EQ(PacketBuffer::kFlushed, buffer.InsertPacket(packet));
- EXPECT_EQ(1, buffer.NumPacketsInBuffer());
- EXPECT_EQ(large_payload_len, buffer.current_memory_bytes());
-
- // Flush buffer to delete remaining packets.
+ // Flush buffer to delete all packets.
buffer.Flush();
}
// Test inserting a list of packets.
TEST(PacketBuffer, InsertPacketList) {
- PacketBuffer buffer(10, 1000); // 10 packets, 1000 bytes.
+ PacketBuffer buffer(10); // 10 packets.
PacketGenerator gen(0, 0, 0, 10);
PacketList list;
const int payload_len = 10;
@@ -187,7 +170,6 @@
¤t_cng_pt));
EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list.
EXPECT_EQ(10, buffer.NumPacketsInBuffer());
- EXPECT_EQ(10 * payload_len, buffer.current_memory_bytes());
EXPECT_EQ(0, current_pt); // Current payload type changed to 0.
EXPECT_EQ(0xFF, current_cng_pt); // CNG payload type not changed.
@@ -200,7 +182,7 @@
// Expecting the buffer to flush.
// TODO(hlundin): Remove this test when legacy operation is no longer needed.
TEST(PacketBuffer, InsertPacketListChangePayloadType) {
- PacketBuffer buffer(10, 1000); // 10 packets, 1000 bytes.
+ PacketBuffer buffer(10); // 10 packets.
PacketGenerator gen(0, 0, 0, 10);
PacketList list;
const int payload_len = 10;
@@ -229,7 +211,6 @@
¤t_cng_pt));
EXPECT_TRUE(list.empty()); // The PacketBuffer should have depleted the list.
EXPECT_EQ(1, buffer.NumPacketsInBuffer()); // Only the last packet.
- EXPECT_EQ(1 * payload_len, buffer.current_memory_bytes());
EXPECT_EQ(1, current_pt); // Current payload type changed to 0.
EXPECT_EQ(0xFF, current_cng_pt); // CNG payload type not changed.
@@ -252,7 +233,7 @@
// 8 0x0005 0x00000028 0x0000001E
// 9 0x0006 0x00000032 0x00000028
TEST(PacketBuffer, ExtractOrderRedundancy) {
- PacketBuffer buffer(100, 1000); // 100 packets, 1000 bytes.
+ PacketBuffer buffer(100); // 100 packets.
const uint32_t ts_increment = 10; // Samples per packet.
const uint16_t start_seq_no = 0xFFFF - 2; // Wraps after 3 packets.
const uint32_t start_ts = 0xFFFFFFFF -
@@ -321,7 +302,7 @@
}
TEST(PacketBuffer, DiscardPackets) {
- PacketBuffer buffer(100, 1000); // 100 packets, 1000 bytes.
+ PacketBuffer buffer(100); // 100 packets.
const uint16_t start_seq_no = 17;
const uint32_t start_ts = 4711;
const uint32_t ts_increment = 10;
@@ -335,7 +316,6 @@
buffer.InsertPacket(packet);
}
EXPECT_EQ(10, buffer.NumPacketsInBuffer());
- EXPECT_EQ(10 * payload_len, buffer.current_memory_bytes());
// Discard them one by one and make sure that the right packets are at the
// front of the buffer.
@@ -351,7 +331,7 @@
}
TEST(PacketBuffer, Reordering) {
- PacketBuffer buffer(100, 1000); // 100 packets, 1000 bytes.
+ PacketBuffer buffer(100); // 100 packets.
const uint16_t start_seq_no = 17;
const uint32_t start_ts = 4711;
const uint32_t ts_increment = 10;
@@ -384,7 +364,6 @@
¤t_pt,
¤t_cng_pt));
EXPECT_EQ(10, buffer.NumPacketsInBuffer());
- EXPECT_EQ(10 * payload_len, buffer.current_memory_bytes());
// Extract them and make sure that come out in the right order.
uint32_t current_ts = start_ts;
@@ -408,18 +387,8 @@
int payload_len = 100;
PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment);
- PacketBuffer* buffer = new PacketBuffer(0, 1000); // 0 packets, 1000 bytes.
- Packet* packet = gen.NextPacket(payload_len);
- EXPECT_EQ(PacketBuffer::kOversizePacket, buffer->InsertPacket(packet));
- delete buffer;
-
- buffer = new PacketBuffer(100, 10); // 100 packets, 10 bytes.
- packet = gen.NextPacket(payload_len);
- EXPECT_EQ(PacketBuffer::kOversizePacket, buffer->InsertPacket(packet));
- delete buffer;
-
- buffer = new PacketBuffer(100, 10000); // 100 packets, 10000 bytes.
- packet = NULL;
+ PacketBuffer* buffer = new PacketBuffer(100); // 100 packets.
+ Packet* packet = NULL;
EXPECT_EQ(PacketBuffer::kInvalidPacket, buffer->InsertPacket(packet));
packet = gen.NextPacket(payload_len);
delete [] packet->payload;
@@ -448,7 +417,7 @@
// Insert packet list of three packets, where the second packet has an invalid
// payload. Expect first packet to be inserted, and the remaining two to be
// discarded.
- buffer = new PacketBuffer(100, 1000); // 100 packets, 1000 bytes.
+ buffer = new PacketBuffer(100); // 100 packets.
PacketList list;
list.push_back(gen.NextPacket(payload_len)); // Valid packet.
packet = gen.NextPacket(payload_len);
diff --git a/modules/audio_coding/neteq4/time_stretch.cc b/modules/audio_coding/neteq4/time_stretch.cc
index 5b6b3ba..5b246c1 100644
--- a/modules/audio_coding/neteq4/time_stretch.cc
+++ b/modules/audio_coding/neteq4/time_stretch.cc
@@ -29,7 +29,7 @@
int fs_mult_120 = fs_mult_ * 120; // Corresponds to 15 ms.
const int16_t* signal;
- scoped_array<int16_t> signal_array;
+ scoped_ptr<int16_t[]> signal_array;
size_t signal_len;
if (num_channels_ == 1) {
signal = input;
diff --git a/modules/audio_coding/neteq4/tools/audio_loop.h b/modules/audio_coding/neteq4/tools/audio_loop.h
index 038ca37..0499a75 100644
--- a/modules/audio_coding/neteq4/tools/audio_loop.h
+++ b/modules/audio_coding/neteq4/tools/audio_loop.h
@@ -27,8 +27,7 @@
AudioLoop()
: next_index_(0),
loop_length_samples_(0),
- block_length_samples_(0),
- audio_array_(NULL) {
+ block_length_samples_(0) {
}
virtual ~AudioLoop() {}
@@ -50,7 +49,7 @@
size_t next_index_;
size_t loop_length_samples_;
size_t block_length_samples_;
- scoped_array<int16_t> audio_array_;
+ scoped_ptr<int16_t[]> audio_array_;
DISALLOW_COPY_AND_ASSIGN(AudioLoop);
};
diff --git a/modules/audio_coding_module.target.darwin-arm.mk b/modules/audio_coding_module.target.darwin-arm.mk
index 683b823..a35c1ef 100644
--- a/modules/audio_coding_module.target.darwin-arm.mk
+++ b/modules/audio_coding_module.target.darwin-arm.mk
@@ -24,31 +24,34 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
# Flags passed to both C and C++ files.
@@ -150,11 +153,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -271,11 +276,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
diff --git a/modules/acm2.target.darwin-x86_64.mk b/modules/audio_coding_module.target.darwin-arm64.mk
similarity index 94%
rename from modules/acm2.target.darwin-x86_64.mk
rename to modules/audio_coding_module.target.darwin-arm64.mk
index e2c8787..3cd1529 100644
--- a/modules/acm2.target.darwin-x86_64.mk
+++ b/modules/audio_coding_module.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
+LOCAL_MODULE := third_party_webrtc_modules_audio_coding_module_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -56,7 +56,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,13 +75,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -151,7 +146,6 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -176,7 +170,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -196,13 +189,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -272,7 +261,6 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -307,8 +295,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -323,8 +309,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -347,10 +331,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
+gyp_all_modules: third_party_webrtc_modules_audio_coding_module_gyp
# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
+.PHONY: audio_coding_module
+audio_coding_module: third_party_webrtc_modules_audio_coding_module_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/audio_coding_module.target.darwin-mips.mk b/modules/audio_coding_module.target.darwin-mips.mk
index 9efe46e..f1a9f71 100644
--- a/modules/audio_coding_module.target.darwin-mips.mk
+++ b/modules/audio_coding_module.target.darwin-mips.mk
@@ -24,31 +24,34 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
# Flags passed to both C and C++ files.
@@ -150,11 +153,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -271,11 +276,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
diff --git a/modules/audio_coding_module.target.darwin-x86.mk b/modules/audio_coding_module.target.darwin-x86.mk
index 9040f19..dc9432a 100644
--- a/modules/audio_coding_module.target.darwin-x86.mk
+++ b/modules/audio_coding_module.target.darwin-x86.mk
@@ -24,31 +24,34 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
# Flags passed to both C and C++ files.
@@ -148,11 +151,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -266,11 +271,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
diff --git a/modules/audio_coding_module.target.darwin-x86_64.mk b/modules/audio_coding_module.target.darwin-x86_64.mk
index 6b90bee..f60e2f4 100644
--- a/modules/audio_coding_module.target.darwin-x86_64.mk
+++ b/modules/audio_coding_module.target.darwin-x86_64.mk
@@ -24,31 +24,34 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
# Flags passed to both C and C++ files.
@@ -148,11 +151,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -266,11 +271,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
diff --git a/modules/audio_coding_module.target.linux-arm.mk b/modules/audio_coding_module.target.linux-arm.mk
index 683b823..a35c1ef 100644
--- a/modules/audio_coding_module.target.linux-arm.mk
+++ b/modules/audio_coding_module.target.linux-arm.mk
@@ -24,31 +24,34 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
# Flags passed to both C and C++ files.
@@ -150,11 +153,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -271,11 +276,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
diff --git a/modules/acm2.target.darwin-x86_64.mk b/modules/audio_coding_module.target.linux-arm64.mk
similarity index 94%
copy from modules/acm2.target.darwin-x86_64.mk
copy to modules/audio_coding_module.target.linux-arm64.mk
index e2c8787..3cd1529 100644
--- a/modules/acm2.target.darwin-x86_64.mk
+++ b/modules/audio_coding_module.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_acm2_gyp
+LOCAL_MODULE := third_party_webrtc_modules_audio_coding_module_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -56,7 +56,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,13 +75,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -151,7 +146,6 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -176,7 +170,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -196,13 +189,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -272,7 +261,6 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -307,8 +295,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -323,8 +309,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -347,10 +331,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_acm2_gyp
+gyp_all_modules: third_party_webrtc_modules_audio_coding_module_gyp
# Alias gyp target name.
-.PHONY: acm2
-acm2: third_party_webrtc_modules_acm2_gyp
+.PHONY: audio_coding_module
+audio_coding_module: third_party_webrtc_modules_audio_coding_module_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/audio_coding_module.target.linux-mips.mk b/modules/audio_coding_module.target.linux-mips.mk
index 9efe46e..f1a9f71 100644
--- a/modules/audio_coding_module.target.linux-mips.mk
+++ b/modules/audio_coding_module.target.linux-mips.mk
@@ -24,31 +24,34 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
# Flags passed to both C and C++ files.
@@ -150,11 +153,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -271,11 +276,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
diff --git a/modules/audio_coding_module.target.linux-x86.mk b/modules/audio_coding_module.target.linux-x86.mk
index 9040f19..dc9432a 100644
--- a/modules/audio_coding_module.target.linux-x86.mk
+++ b/modules/audio_coding_module.target.linux-x86.mk
@@ -24,31 +24,34 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
# Flags passed to both C and C++ files.
@@ -148,11 +151,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -266,11 +271,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
diff --git a/modules/audio_coding_module.target.linux-x86_64.mk b/modules/audio_coding_module.target.linux-x86_64.mk
index 6b90bee..f60e2f4 100644
--- a/modules/audio_coding_module.target.linux-x86_64.mk
+++ b/modules/audio_coding_module.target.linux-x86_64.mk
@@ -24,31 +24,34 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/main/source/acm_amr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_amrwb.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_celt.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_cng.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_codec_database.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_detection.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_dtmf_playout.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g722.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7221c.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g729.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_g7291.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_generic_codec.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_gsmfr.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_ilbc.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_isac.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_neteq.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_opus.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_speex.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcm16b.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcma.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_pcmu.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_red.cc \
- third_party/webrtc/modules/audio_coding/main/source/acm_resampler.cc \
- third_party/webrtc/modules/audio_coding/main/source/audio_coding_module_impl.cc
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_amrwb.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_celt.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_cng.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_codec_database.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_dtmf_playout.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g722.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7221c.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g729.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_g7291.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_gsmfr.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_ilbc.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_isac.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_opus.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_speex.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcm16b.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcma.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_pcmu.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_red.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/acm_resampler.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/call_statistics.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc \
+ third_party/webrtc/modules/audio_coding/main/acm2/nack.cc
# Flags passed to both C and C++ files.
@@ -148,11 +151,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -266,11 +271,13 @@
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/pcm16b/include \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/opus/src/celt \
+ $(LOCAL_PATH)/third_party/opus/src/src \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/audio_conference_mixer.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/audio_conference_mixer.target.darwin-arm64.mk
index c84869c..97ce017 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/audio_conference_mixer.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_audio_conference_mixer_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,20 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc \
+ third_party/webrtc/modules/audio_conference_mixer/source/level_indicator.cc \
+ third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc \
+ third_party/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +51,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +93,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +111,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_conference_mixer/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +131,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +150,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +192,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +211,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_conference_mixer/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +241,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +255,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +277,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_audio_conference_mixer_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: audio_conference_mixer
+audio_conference_mixer: third_party_webrtc_modules_audio_conference_mixer_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/audio_conference_mixer.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/audio_conference_mixer.target.linux-arm64.mk
index c84869c..97ce017 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/audio_conference_mixer.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_audio_conference_mixer_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,20 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc \
+ third_party/webrtc/modules/audio_conference_mixer/source/level_indicator.cc \
+ third_party/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc \
+ third_party/webrtc/modules/audio_conference_mixer/source/time_scheduler.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +51,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +93,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +111,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_conference_mixer/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +131,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +150,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +192,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +211,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_conference_mixer/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +241,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +255,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +277,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_audio_conference_mixer_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: audio_conference_mixer
+audio_conference_mixer: third_party_webrtc_modules_audio_conference_mixer_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/audio_device.target.darwin-arm64.mk
similarity index 69%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/audio_device.target.darwin-arm64.mk
index 3b05cd9..f9b3442 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/audio_device.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_audio_device_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,18 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_device/audio_device_buffer.cc \
+ third_party/webrtc/modules/audio_device/audio_device_generic.cc \
+ third_party/webrtc/modules/audio_device/audio_device_utility.cc \
+ third_party/webrtc/modules/audio_device/audio_device_impl.cc \
+ third_party/webrtc/modules/audio_device/dummy/audio_device_dummy.cc \
+ third_party/webrtc/modules/audio_device/dummy/audio_device_utility_dummy.cc
# Flags passed to both C and C++ files.
@@ -70,6 +44,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +53,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +64,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +95,7 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_DUMMY_AUDIO_BUILD' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +114,15 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/dummy \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/android \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +150,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +159,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +170,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +201,7 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_DUMMY_AUDIO_BUILD' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +221,15 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/dummy \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/android \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +257,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +271,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +293,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_audio_device_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: audio_device
+audio_device: third_party_webrtc_modules_audio_device_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/audio_device.target.linux-arm64.mk
similarity index 69%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/audio_device.target.linux-arm64.mk
index 3b05cd9..f9b3442 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/audio_device.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_audio_device_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,18 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_device/audio_device_buffer.cc \
+ third_party/webrtc/modules/audio_device/audio_device_generic.cc \
+ third_party/webrtc/modules/audio_device/audio_device_utility.cc \
+ third_party/webrtc/modules/audio_device/audio_device_impl.cc \
+ third_party/webrtc/modules/audio_device/dummy/audio_device_dummy.cc \
+ third_party/webrtc/modules/audio_device/dummy/audio_device_utility_dummy.cc
# Flags passed to both C and C++ files.
@@ -70,6 +44,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +53,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +64,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +95,7 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_DUMMY_AUDIO_BUILD' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +114,15 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/dummy \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/android \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +150,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +159,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +170,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +201,7 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_DUMMY_AUDIO_BUILD' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +221,15 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/dummy \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/android \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +257,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +271,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +293,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_audio_device_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: audio_device
+audio_device: third_party_webrtc_modules_audio_device_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/audio_device/android/fine_audio_buffer.h b/modules/audio_device/android/fine_audio_buffer.h
index 597b8aa..e577b72 100644
--- a/modules/audio_device/android/fine_audio_buffer.h
+++ b/modules/audio_device/android/fine_audio_buffer.h
@@ -56,7 +56,7 @@
int bytes_per_10_ms_;
// Storage for samples that are not yet asked for.
- scoped_array<int8_t> cache_buffer_;
+ scoped_ptr<int8_t[]> cache_buffer_;
int cached_buffer_start_; // Location of first unread sample.
int cached_bytes_; // Number of bytes stored in cache.
};
diff --git a/modules/audio_device/android/fine_audio_buffer_unittest.cc b/modules/audio_device/android/fine_audio_buffer_unittest.cc
index 69ba741..e1f03f8 100644
--- a/modules/audio_device/android/fine_audio_buffer_unittest.cc
+++ b/modules/audio_device/android/fine_audio_buffer_unittest.cc
@@ -80,7 +80,7 @@
FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
sample_rate);
- scoped_array<int8_t> out_buffer;
+ scoped_ptr<int8_t[]> out_buffer;
out_buffer.reset(
new int8_t[fine_buffer.RequiredBufferSizeBytes()]);
for (int i = 0; i < kNumberOfFrames; ++i) {
diff --git a/modules/audio_device/android/opensles_input.cc b/modules/audio_device/android/opensles_input.cc
index 6b600c9..f22d8bf 100644
--- a/modules/audio_device/android/opensles_input.cc
+++ b/modules/audio_device/android/opensles_input.cc
@@ -289,7 +289,7 @@
fifo_.reset(new SingleRwFifo(num_fifo_buffers_needed_));
// Allocate the memory area to be used.
- rec_buf_.reset(new scoped_array<int8_t>[TotalBuffersUsed()]);
+ rec_buf_.reset(new scoped_ptr<int8_t[]>[TotalBuffersUsed()]);
for (int i = 0; i < TotalBuffersUsed(); ++i) {
rec_buf_[i].reset(new int8_t[buffer_size_bytes()]);
}
diff --git a/modules/audio_device/android/opensles_input.h b/modules/audio_device/android/opensles_input.h
index 48e4fd2..d27d824 100644
--- a/modules/audio_device/android/opensles_input.h
+++ b/modules/audio_device/android/opensles_input.h
@@ -205,7 +205,7 @@
// Audio buffers
AudioDeviceBuffer* audio_buffer_;
// Holds all allocated memory such that it is deallocated properly.
- scoped_array<scoped_array<int8_t> > rec_buf_;
+ scoped_ptr<scoped_ptr<int8_t[]>[]> rec_buf_;
// Index in |rec_buf_| pointing to the audio buffer that will be ready the
// next time RecorderSimpleBufferQueueCallbackHandler is invoked.
// Ready means buffer contains audio data from the device.
diff --git a/modules/audio_device/android/opensles_output.cc b/modules/audio_device/android/opensles_output.cc
index 6185b2a..377789b 100644
--- a/modules/audio_device/android/opensles_output.cc
+++ b/modules/audio_device/android/opensles_output.cc
@@ -340,7 +340,7 @@
fifo_.reset(new SingleRwFifo(num_fifo_buffers_needed_));
// Allocate the memory area to be used.
- play_buf_.reset(new scoped_array<int8_t>[TotalBuffersUsed()]);
+ play_buf_.reset(new scoped_ptr<int8_t[]>[TotalBuffersUsed()]);
int required_buffer_size = fine_buffer_->RequiredBufferSizeBytes();
for (int i = 0; i < TotalBuffersUsed(); ++i) {
play_buf_[i].reset(new int8_t[required_buffer_size]);
diff --git a/modules/audio_device/android/opensles_output.h b/modules/audio_device/android/opensles_output.h
index 464a7e4..aa9b5bf 100644
--- a/modules/audio_device/android/opensles_output.h
+++ b/modules/audio_device/android/opensles_output.h
@@ -223,7 +223,7 @@
// Audio buffers
AudioDeviceBuffer* audio_buffer_;
scoped_ptr<FineAudioBuffer> fine_buffer_;
- scoped_array<scoped_array<int8_t> > play_buf_;
+ scoped_ptr<scoped_ptr<int8_t[]>[]> play_buf_;
// Index in |rec_buf_| pointing to the audio buffer that will be ready the
// next time PlayerSimpleBufferQueueCallbackHandler is invoked.
// Ready means buffer is ready to be played out to device.
diff --git a/modules/audio_device/android/single_rw_fifo.h b/modules/audio_device/android/single_rw_fifo.h
index a1fcfaa..092b1d5 100644
--- a/modules/audio_device/android/single_rw_fifo.h
+++ b/modules/audio_device/android/single_rw_fifo.h
@@ -35,7 +35,7 @@
int capacity() const { return capacity_; }
private:
- scoped_array<int8_t*> queue_;
+ scoped_ptr<int8_t*[]> queue_;
int capacity_;
Atomic32 size_;
diff --git a/modules/audio_device/android/single_rw_fifo_unittest.cc b/modules/audio_device/android/single_rw_fifo_unittest.cc
index c722c27..9925baa 100644
--- a/modules/audio_device/android/single_rw_fifo_unittest.cc
+++ b/modules/audio_device/android/single_rw_fifo_unittest.cc
@@ -90,7 +90,7 @@
protected:
SingleRwFifo fifo_;
// Memory area for proper de-allocation.
- scoped_array<int8_t> buffer_[kCapacity];
+ scoped_ptr<int8_t[]> buffer_[kCapacity];
std::list<int8_t*> memory_queue_;
int pushed_;
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/audio_processing.target.darwin-arm64.mk
similarity index 68%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/audio_processing.target.darwin-arm64.mk
index c84869c..6450d07 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/audio_processing.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_audio_processing_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -11,56 +11,50 @@
gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
+GYP_TARGET_DEPENDENCIES := \
+ $(call intermediates-dir-for,STATIC_LIBRARIES,third_party_webrtc_modules_audioproc_debug_proto_gyp,,,$(GYP_VAR_PREFIX))/third_party_webrtc_modules_audioproc_debug_proto_gyp.a
GYP_GENERATED_OUTPUTS :=
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_processing/aec/echo_cancellation.c \
+ third_party/webrtc/modules/audio_processing/aec/aec_core.c \
+ third_party/webrtc/modules/audio_processing/aec/aec_rdft.c \
+ third_party/webrtc/modules/audio_processing/aec/aec_resampler.c \
+ third_party/webrtc/modules/audio_processing/aecm/echo_control_mobile.c \
+ third_party/webrtc/modules/audio_processing/aecm/aecm_core.c \
+ third_party/webrtc/modules/audio_processing/agc/analog_agc.c \
+ third_party/webrtc/modules/audio_processing/agc/digital_agc.c \
+ third_party/webrtc/modules/audio_processing/audio_buffer.cc \
+ third_party/webrtc/modules/audio_processing/audio_processing_impl.cc \
+ third_party/webrtc/modules/audio_processing/echo_cancellation_impl.cc \
+ third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.cc \
+ third_party/webrtc/modules/audio_processing/gain_control_impl.cc \
+ third_party/webrtc/modules/audio_processing/high_pass_filter_impl.cc \
+ third_party/webrtc/modules/audio_processing/level_estimator_impl.cc \
+ third_party/webrtc/modules/audio_processing/noise_suppression_impl.cc \
+ third_party/webrtc/modules/audio_processing/processing_component.cc \
+ third_party/webrtc/modules/audio_processing/typing_detection.cc \
+ third_party/webrtc/modules/audio_processing/utility/delay_estimator.c \
+ third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c \
+ third_party/webrtc/modules/audio_processing/utility/fft4g.c \
+ third_party/webrtc/modules/audio_processing/utility/ring_buffer.c \
+ third_party/webrtc/modules/audio_processing/voice_detection_impl.cc \
+ third_party/webrtc/modules/audio_processing/ns/noise_suppression.c \
+ third_party/webrtc/modules/audio_processing/ns/ns_core.c \
+ third_party/webrtc/modules/audio_processing/aecm/aecm_core_c.c
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +74,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +116,11 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_AUDIOPROC_DEBUG_DUMP' \
+ '-DWEBRTC_NS_FLOAT' \
+ '-DPROTOBUF_USE_DLLS' \
+ '-DGOOGLE_PROTOBUF_NO_RTTI' \
+ '-DGOOGLE_PROTOBUF_NO_STATIC_INITIALIZER' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +139,13 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(gyp_shared_intermediate_dir)/protoc_out \
+ $(LOCAL_PATH)/third_party/protobuf \
+ $(LOCAL_PATH)/third_party/protobuf/src \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +163,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +182,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +224,11 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_AUDIOPROC_DEBUG_DUMP' \
+ '-DWEBRTC_NS_FLOAT' \
+ '-DPROTOBUF_USE_DLLS' \
+ '-DGOOGLE_PROTOBUF_NO_RTTI' \
+ '-DGOOGLE_PROTOBUF_NO_STATIC_INITIALIZER' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +248,13 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(gyp_shared_intermediate_dir)/protoc_out \
+ $(LOCAL_PATH)/third_party/protobuf \
+ $(LOCAL_PATH)/third_party/protobuf/src \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +282,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +296,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -318,7 +307,8 @@
LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-LOCAL_STATIC_LIBRARIES :=
+LOCAL_STATIC_LIBRARIES := \
+ third_party_webrtc_modules_audioproc_debug_proto_gyp
# Enable grouping to fix circular references
LOCAL_GROUP_STATIC_LIBRARIES := true
@@ -329,10 +319,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_audio_processing_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: audio_processing
+audio_processing: third_party_webrtc_modules_audio_processing_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/audio_processing.target.linux-arm64.mk
similarity index 68%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/audio_processing.target.linux-arm64.mk
index c84869c..6450d07 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/audio_processing.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_audio_processing_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -11,56 +11,50 @@
gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
# Make sure our deps are built first.
-GYP_TARGET_DEPENDENCIES :=
+GYP_TARGET_DEPENDENCIES := \
+ $(call intermediates-dir-for,STATIC_LIBRARIES,third_party_webrtc_modules_audioproc_debug_proto_gyp,,,$(GYP_VAR_PREFIX))/third_party_webrtc_modules_audioproc_debug_proto_gyp.a
GYP_GENERATED_OUTPUTS :=
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_processing/aec/echo_cancellation.c \
+ third_party/webrtc/modules/audio_processing/aec/aec_core.c \
+ third_party/webrtc/modules/audio_processing/aec/aec_rdft.c \
+ third_party/webrtc/modules/audio_processing/aec/aec_resampler.c \
+ third_party/webrtc/modules/audio_processing/aecm/echo_control_mobile.c \
+ third_party/webrtc/modules/audio_processing/aecm/aecm_core.c \
+ third_party/webrtc/modules/audio_processing/agc/analog_agc.c \
+ third_party/webrtc/modules/audio_processing/agc/digital_agc.c \
+ third_party/webrtc/modules/audio_processing/audio_buffer.cc \
+ third_party/webrtc/modules/audio_processing/audio_processing_impl.cc \
+ third_party/webrtc/modules/audio_processing/echo_cancellation_impl.cc \
+ third_party/webrtc/modules/audio_processing/echo_control_mobile_impl.cc \
+ third_party/webrtc/modules/audio_processing/gain_control_impl.cc \
+ third_party/webrtc/modules/audio_processing/high_pass_filter_impl.cc \
+ third_party/webrtc/modules/audio_processing/level_estimator_impl.cc \
+ third_party/webrtc/modules/audio_processing/noise_suppression_impl.cc \
+ third_party/webrtc/modules/audio_processing/processing_component.cc \
+ third_party/webrtc/modules/audio_processing/typing_detection.cc \
+ third_party/webrtc/modules/audio_processing/utility/delay_estimator.c \
+ third_party/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.c \
+ third_party/webrtc/modules/audio_processing/utility/fft4g.c \
+ third_party/webrtc/modules/audio_processing/utility/ring_buffer.c \
+ third_party/webrtc/modules/audio_processing/voice_detection_impl.cc \
+ third_party/webrtc/modules/audio_processing/ns/noise_suppression.c \
+ third_party/webrtc/modules/audio_processing/ns/ns_core.c \
+ third_party/webrtc/modules/audio_processing/aecm/aecm_core_c.c
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +74,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +116,11 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_AUDIOPROC_DEBUG_DUMP' \
+ '-DWEBRTC_NS_FLOAT' \
+ '-DPROTOBUF_USE_DLLS' \
+ '-DGOOGLE_PROTOBUF_NO_RTTI' \
+ '-DGOOGLE_PROTOBUF_NO_STATIC_INITIALIZER' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +139,13 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(gyp_shared_intermediate_dir)/protoc_out \
+ $(LOCAL_PATH)/third_party/protobuf \
+ $(LOCAL_PATH)/third_party/protobuf/src \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +163,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +182,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +224,11 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_AUDIOPROC_DEBUG_DUMP' \
+ '-DWEBRTC_NS_FLOAT' \
+ '-DPROTOBUF_USE_DLLS' \
+ '-DGOOGLE_PROTOBUF_NO_RTTI' \
+ '-DGOOGLE_PROTOBUF_NO_STATIC_INITIALIZER' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +248,13 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(gyp_shared_intermediate_dir)/protoc_out \
+ $(LOCAL_PATH)/third_party/protobuf \
+ $(LOCAL_PATH)/third_party/protobuf/src \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +282,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +296,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -318,7 +307,8 @@
LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-LOCAL_STATIC_LIBRARIES :=
+LOCAL_STATIC_LIBRARIES := \
+ third_party_webrtc_modules_audioproc_debug_proto_gyp
# Enable grouping to fix circular references
LOCAL_GROUP_STATIC_LIBRARIES := true
@@ -329,10 +319,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_audio_processing_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: audio_processing
+audio_processing: third_party_webrtc_modules_audio_processing_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/audio_processing/aec/aec_core.c b/modules/audio_processing/aec/aec_core.c
index 3f3d2c0..b1b0e11 100644
--- a/modules/audio_processing/aec/aec_core.c
+++ b/modules/audio_processing/aec/aec_core.c
@@ -473,6 +473,7 @@
aec->delay_logging_enabled = 0;
memset(aec->delay_histogram, 0, sizeof(aec->delay_histogram));
+ aec->reported_delay_enabled = 1;
aec->extended_filter_enabled = 0;
aec->num_partitions = kNormalNumPartitions;
@@ -785,6 +786,14 @@
}
}
+void WebRtcAec_enable_reported_delay(AecCore* self, int enable) {
+ self->reported_delay_enabled = enable;
+}
+
+int WebRtcAec_reported_delay_enabled(AecCore* self) {
+ return self->reported_delay_enabled;
+}
+
void WebRtcAec_enable_delay_correction(AecCore* self, int enable) {
self->extended_filter_enabled = enable;
self->num_partitions = enable ? kExtendedNumPartitions : kNormalNumPartitions;
diff --git a/modules/audio_processing/aec/aec_core.h b/modules/audio_processing/aec/aec_core.h
index e1f6f90..327a5a9 100644
--- a/modules/audio_processing/aec/aec_core.h
+++ b/modules/audio_processing/aec/aec_core.h
@@ -22,17 +22,6 @@
#define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients
#define PART_LEN2 (PART_LEN * 2) // Length of partition * 2
-// Delay estimator constants, used for logging.
-enum {
- kMaxDelayBlocks = 60
-};
-enum {
- kLookaheadBlocks = 15
-};
-enum {
- kHistorySizeBlocks = kMaxDelayBlocks + kLookaheadBlocks
-};
-
typedef float complex_t[2];
// For performance reasons, some arrays of complex numbers are replaced by twice
// as long arrays of float, all the real parts followed by all the imaginary
@@ -104,6 +93,12 @@
int metrics_mode,
int delay_logging);
+// Non-zero enables, zero disables.
+void WebRtcAec_enable_reported_delay(AecCore* self, int enable);
+
+// Returns non-zero if reported delay is enabled and zero if disabled.
+int WebRtcAec_reported_delay_enabled(AecCore* self);
+
// We now interpret delay correction to mean an extended filter length feature.
// We reuse the delay correction infrastructure to avoid changes through to
// libjingle. See details along with |DelayCorrection| in
diff --git a/modules/audio_processing/aec/aec_core_internal.h b/modules/audio_processing/aec/aec_core_internal.h
index c6b762a..1c560f9 100644
--- a/modules/audio_processing/aec/aec_core_internal.h
+++ b/modules/audio_processing/aec/aec_core_internal.h
@@ -26,6 +26,17 @@
};
static const int kNormalNumPartitions = 12;
+// Delay estimator constants, used for logging.
+enum {
+ kMaxDelayBlocks = 60
+};
+enum {
+ kLookaheadBlocks = 15
+};
+enum {
+ kHistorySizeBlocks = kMaxDelayBlocks + kLookaheadBlocks
+};
+
// Extended filter adaptation parameters.
// TODO(ajm): No narrowband tuning yet.
static const float kExtendedMu = 0.4f;
@@ -122,6 +133,7 @@
void* delay_estimator_farend;
void* delay_estimator;
+ int reported_delay_enabled; // 0 = disabled, otherwise enabled.
// 1 = extended filter mode enabled, 0 = disabled.
int extended_filter_enabled;
// Runtime selection of number of filter partitions.
diff --git a/modules/audio_processing/aec/echo_cancellation.c b/modules/audio_processing/aec/echo_cancellation.c
index bbdd5f6..c7f4a9c 100644
--- a/modules/audio_processing/aec/echo_cancellation.c
+++ b/modules/audio_processing/aec/echo_cancellation.c
@@ -254,7 +254,7 @@
aecpc->checkBuffSize = 1;
aecpc->firstVal = 0;
- aecpc->startup_phase = 1;
+ aecpc->startup_phase = WebRtcAec_reported_delay_enabled(aecpc->aec);
aecpc->bufSizeStart = 0;
aecpc->checkBufSizeCtr = 0;
aecpc->msInSndCardBuf = 0;
@@ -766,7 +766,9 @@
}
} else {
// AEC is enabled.
- EstBufDelayNormal(aecpc);
+ if (WebRtcAec_reported_delay_enabled(aecpc->aec)) {
+ EstBufDelayNormal(aecpc);
+ }
// Note that 1 frame is supported for NB and 2 frames for WB.
for (i = 0; i < nFrames; i++) {
@@ -842,7 +844,9 @@
self->startup_phase = 0;
}
- EstBufDelayExtended(self);
+ if (WebRtcAec_reported_delay_enabled(self->aec)) {
+ EstBufDelayExtended(self);
+ }
{
// |delay_diff_offset| gives us the option to manually rewind the delay on
diff --git a/modules/audio_processing/audio_buffer.cc b/modules/audio_processing/audio_buffer.cc
index 6936155..c53d4df 100644
--- a/modules/audio_processing/audio_buffer.cc
+++ b/modules/audio_processing/audio_buffer.cc
@@ -23,6 +23,35 @@
kSamplesPer32kHzChannel = 320
};
+bool HasKeyboardChannel(AudioProcessing::ChannelLayout layout) {
+ switch (layout) {
+ case AudioProcessing::kMono:
+ case AudioProcessing::kStereo:
+ return false;
+ case AudioProcessing::kMonoAndKeyboard:
+ case AudioProcessing::kStereoAndKeyboard:
+ return true;
+ }
+ assert(false);
+ return false;
+}
+
+int KeyboardChannelIndex(AudioProcessing::ChannelLayout layout) {
+ switch (layout) {
+ case AudioProcessing::kMono:
+ case AudioProcessing::kStereo:
+ assert(false);
+ return -1;
+ case AudioProcessing::kMonoAndKeyboard:
+ return 1;
+ case AudioProcessing::kStereoAndKeyboard:
+ return 2;
+ }
+ assert(false);
+ return -1;
+}
+
+
void StereoToMono(const float* left, const float* right, float* out,
int samples_per_channel) {
for (int i = 0; i < samples_per_channel; ++i) {
@@ -32,8 +61,9 @@
void StereoToMono(const int16_t* left, const int16_t* right, int16_t* out,
int samples_per_channel) {
- for (int i = 0; i < samples_per_channel; i++)
+ for (int i = 0; i < samples_per_channel; ++i) {
out[i] = (left[i] + right[i]) >> 1;
+ }
}
} // namespace
@@ -72,6 +102,7 @@
activity_(AudioFrame::kVadUnknown),
is_muted_(false),
data_(NULL),
+ keyboard_data_(NULL),
channels_(new ChannelBuffer<int16_t>(proc_samples_per_channel_,
num_proc_channels_)) {
assert(input_samples_per_channel_ > 0);
@@ -118,6 +149,8 @@
}
}
+AudioBuffer::~AudioBuffer() {}
+
void AudioBuffer::CopyFrom(const float* const* data,
int samples_per_channel,
AudioProcessing::ChannelLayout layout) {
@@ -125,6 +158,10 @@
assert(ChannelsFromLayout(layout) == num_input_channels_);
InitForNewData();
+ if (HasKeyboardChannel(layout)) {
+ keyboard_data_ = data[KeyboardChannelIndex(layout)];
+ }
+
// Downmix.
const float* const* data_ptr = data;
if (num_input_channels_ == 2 && num_proc_channels_ == 1) {
@@ -180,10 +217,9 @@
}
}
-AudioBuffer::~AudioBuffer() {}
-
void AudioBuffer::InitForNewData() {
data_ = NULL;
+ keyboard_data_ = NULL;
data_was_mixed_ = false;
num_mixed_channels_ = 0;
num_mixed_low_pass_channels_ = 0;
@@ -240,6 +276,10 @@
return low_pass_reference_channels_->channel(channel);
}
+const float* AudioBuffer::keyboard_data() const {
+ return keyboard_data_;
+}
+
SplitFilterStates* AudioBuffer::filter_states(int channel) const {
assert(channel >= 0 && channel < num_proc_channels_);
return &filter_states_[channel];
@@ -269,6 +309,11 @@
return samples_per_split_channel_;
}
+int AudioBuffer::samples_per_keyboard_channel() const {
+ // We don't resample the keyboard channel.
+ return input_samples_per_channel_;
+}
+
// TODO(andrew): Do deinterleaving and mixing in one step?
void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
assert(proc_samples_per_channel_ == input_samples_per_channel_);
diff --git a/modules/audio_processing/audio_buffer.h b/modules/audio_processing/audio_buffer.h
index 45e62a4..eaf53eb 100644
--- a/modules/audio_processing/audio_buffer.h
+++ b/modules/audio_processing/audio_buffer.h
@@ -53,6 +53,7 @@
int num_channels() const;
int samples_per_channel() const;
int samples_per_split_channel() const;
+ int samples_per_keyboard_channel() const;
int16_t* data(int channel) const;
int16_t* low_pass_split_data(int channel) const;
@@ -60,6 +61,7 @@
int16_t* mixed_data(int channel) const;
int16_t* mixed_low_pass_data(int channel) const;
int16_t* low_pass_reference(int channel) const;
+ const float* keyboard_data() const;
SplitFilterStates* filter_states(int channel) const;
@@ -106,6 +108,7 @@
bool is_muted_;
int16_t* data_;
+ const float* keyboard_data_;
scoped_ptr<ChannelBuffer<int16_t> > channels_;
scoped_ptr<SplitChannelBuffer> split_channels_;
scoped_ptr<SplitFilterStates[]> filter_states_;
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 147cb18..de387ed 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -468,48 +468,46 @@
}
#endif
+ AudioBuffer* ca = capture_audio_.get(); // For brevity.
bool data_processed = is_data_processed();
if (analysis_needed(data_processed)) {
for (int i = 0; i < fwd_proc_format_.num_channels(); i++) {
- SplitFilterStates* filter_states = capture_audio_->filter_states(i);
// Split into a low and high band.
- WebRtcSpl_AnalysisQMF(capture_audio_->data(i),
- capture_audio_->samples_per_channel(),
- capture_audio_->low_pass_split_data(i),
- capture_audio_->high_pass_split_data(i),
- filter_states->analysis_filter_state1,
- filter_states->analysis_filter_state2);
+ WebRtcSpl_AnalysisQMF(ca->data(i),
+ ca->samples_per_channel(),
+ ca->low_pass_split_data(i),
+ ca->high_pass_split_data(i),
+ ca->filter_states(i)->analysis_filter_state1,
+ ca->filter_states(i)->analysis_filter_state2);
}
}
- RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(capture_audio_.get()));
- RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(capture_audio_.get()));
- RETURN_ON_ERR(echo_cancellation_->ProcessCaptureAudio(capture_audio_.get()));
+ RETURN_ON_ERR(high_pass_filter_->ProcessCaptureAudio(ca));
+ RETURN_ON_ERR(gain_control_->AnalyzeCaptureAudio(ca));
+ RETURN_ON_ERR(echo_cancellation_->ProcessCaptureAudio(ca));
if (echo_control_mobile_->is_enabled() && noise_suppression_->is_enabled()) {
- capture_audio_->CopyLowPassToReference();
+ ca->CopyLowPassToReference();
}
- RETURN_ON_ERR(noise_suppression_->ProcessCaptureAudio(capture_audio_.get()));
- RETURN_ON_ERR(
- echo_control_mobile_->ProcessCaptureAudio(capture_audio_.get()));
- RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(capture_audio_.get()));
- RETURN_ON_ERR(gain_control_->ProcessCaptureAudio(capture_audio_.get()));
+ RETURN_ON_ERR(noise_suppression_->ProcessCaptureAudio(ca));
+ RETURN_ON_ERR(echo_control_mobile_->ProcessCaptureAudio(ca));
+ RETURN_ON_ERR(voice_detection_->ProcessCaptureAudio(ca));
+ RETURN_ON_ERR(gain_control_->ProcessCaptureAudio(ca));
if (synthesis_needed(data_processed)) {
for (int i = 0; i < fwd_proc_format_.num_channels(); i++) {
// Recombine low and high bands.
- SplitFilterStates* filter_states = capture_audio_->filter_states(i);
- WebRtcSpl_SynthesisQMF(capture_audio_->low_pass_split_data(i),
- capture_audio_->high_pass_split_data(i),
- capture_audio_->samples_per_split_channel(),
- capture_audio_->data(i),
- filter_states->synthesis_filter_state1,
- filter_states->synthesis_filter_state2);
+ WebRtcSpl_SynthesisQMF(ca->low_pass_split_data(i),
+ ca->high_pass_split_data(i),
+ ca->samples_per_split_channel(),
+ ca->data(i),
+ ca->filter_states(i)->synthesis_filter_state1,
+ ca->filter_states(i)->synthesis_filter_state2);
}
}
// The level estimator operates on the recombined data.
- RETURN_ON_ERR(level_estimator_->ProcessStream(capture_audio_.get()));
+ RETURN_ON_ERR(level_estimator_->ProcessStream(ca));
was_stream_delay_set_ = false;
return kNoError;
@@ -592,27 +590,23 @@
return AnalyzeReverseStreamLocked();
}
-// TODO(ajm): Have AnalyzeReverseStream accept sample rates not matching the
-// primary stream and convert ourselves rather than having the user manage it.
-// We can be smarter and use the splitting filter when appropriate. Similarly,
-// perform downmixing here.
int AudioProcessingImpl::AnalyzeReverseStreamLocked() {
+ AudioBuffer* ra = render_audio_.get(); // For brevity.
if (rev_proc_format_.rate() == kSampleRate32kHz) {
for (int i = 0; i < rev_proc_format_.num_channels(); i++) {
// Split into low and high band.
- SplitFilterStates* filter_states = render_audio_->filter_states(i);
- WebRtcSpl_AnalysisQMF(render_audio_->data(i),
- render_audio_->samples_per_channel(),
- render_audio_->low_pass_split_data(i),
- render_audio_->high_pass_split_data(i),
- filter_states->analysis_filter_state1,
- filter_states->analysis_filter_state2);
+ WebRtcSpl_AnalysisQMF(ra->data(i),
+ ra->samples_per_channel(),
+ ra->low_pass_split_data(i),
+ ra->high_pass_split_data(i),
+ ra->filter_states(i)->analysis_filter_state1,
+ ra->filter_states(i)->analysis_filter_state2);
}
}
- RETURN_ON_ERR(echo_cancellation_->ProcessRenderAudio(render_audio_.get()));
- RETURN_ON_ERR(echo_control_mobile_->ProcessRenderAudio(render_audio_.get()));
- RETURN_ON_ERR(gain_control_->ProcessRenderAudio(render_audio_.get()));
+ RETURN_ON_ERR(echo_cancellation_->ProcessRenderAudio(ra));
+ RETURN_ON_ERR(echo_control_mobile_->ProcessRenderAudio(ra));
+ RETURN_ON_ERR(gain_control_->ProcessRenderAudio(ra));
return kNoError;
}
diff --git a/modules/audio_processing/common.h b/modules/audio_processing/common.h
index e4ac6ee..42454df 100644
--- a/modules/audio_processing/common.h
+++ b/modules/audio_processing/common.h
@@ -11,6 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
+#include <assert.h>
#include <string.h>
#include "webrtc/modules/audio_processing/include/audio_processing.h"
diff --git a/modules/audio_processing/echo_cancellation_impl.cc b/modules/audio_processing/echo_cancellation_impl.cc
index f0bd95e..d4bd781 100644
--- a/modules/audio_processing/echo_cancellation_impl.cc
+++ b/modules/audio_processing/echo_cancellation_impl.cc
@@ -67,7 +67,8 @@
was_stream_drift_set_(false),
stream_has_echo_(false),
delay_logging_enabled_(false),
- delay_correction_enabled_(false) {}
+ delay_correction_enabled_(false),
+ reported_delay_enabled_(true) {}
EchoCancellationImpl::~EchoCancellationImpl() {}
@@ -361,6 +362,8 @@
WebRtcAec_enable_delay_correction(WebRtcAec_aec_core(
static_cast<Handle*>(handle)), delay_correction_enabled_ ? 1 : 0);
+ WebRtcAec_enable_reported_delay(WebRtcAec_aec_core(
+ static_cast<Handle*>(handle)), reported_delay_enabled_ ? 1 : 0);
return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
}
diff --git a/modules/audio_processing/echo_cancellation_impl.h b/modules/audio_processing/echo_cancellation_impl.h
index b364193..b9c116a 100644
--- a/modules/audio_processing/echo_cancellation_impl.h
+++ b/modules/audio_processing/echo_cancellation_impl.h
@@ -72,6 +72,7 @@
bool stream_has_echo_;
bool delay_logging_enabled_;
bool delay_correction_enabled_;
+ bool reported_delay_enabled_;
};
} // namespace webrtc
diff --git a/modules/audio_processing/gen_aecm_core_neon_offsets_h.target.darwin-arm.mk b/modules/audio_processing/gen_aecm_core_neon_offsets_h.target.darwin-arm.mk
index d81e0c1..3b2e3de 100644
--- a/modules/audio_processing/gen_aecm_core_neon_offsets_h.target.darwin-arm.mk
+++ b/modules/audio_processing/gen_aecm_core_neon_offsets_h.target.darwin-arm.mk
@@ -23,7 +23,7 @@
$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets/aecm_core_neon_offsets.o: export PATH := $(subst $(ANDROID_BUILD_PATHS),,$(PATH))
$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets/aecm_core_neon_offsets.o: $(LOCAL_PATH)/third_party/libvpx/unpack_lib_posix.sh $(GYP_TARGET_DEPENDENCIES)
@echo "Gyp action: third_party_webrtc_modules_audio_processing_gen_core_neon_offsets_chromium_gyp_gen_aecm_core_neon_offsets_h_target_unpack_lib_posix ($@)"
- $(hide)cd $(gyp_local_path)/third_party/webrtc/modules/audio_processing; mkdir -p $(gyp_shared_intermediate_dir)/audio_processing/asm_offsets; ../../../../third_party/libvpx/unpack_lib_posix.sh -d "$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets" -f aecm_core_neon_offsets.o -a "$(gyp_shared_intermediate_dir)/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(PWD)/$(call intermediates-dir-for, STATIC_LIBRARIES, lib_core_neon_offsets,,, $(GYP_VAR_PREFIX))/lib_core_neon_offsets.a" -r "$(PWD)/$($(GYP_VAR_PREFIX)TARGET_AR)"
+ $(hide)cd $(gyp_local_path)/third_party/webrtc/modules/audio_processing; mkdir -p $(gyp_shared_intermediate_dir)/audio_processing/asm_offsets; ../../../../third_party/libvpx/unpack_lib_posix.sh -d "$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets" -f aecm_core_neon_offsets.o -a "$(gyp_shared_intermediate_dir)/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(realpath $(call intermediates-dir-for, STATIC_LIBRARIES, lib_core_neon_offsets,,, $(GYP_VAR_PREFIX)))/lib_core_neon_offsets.a" -r "$(realpath $($(GYP_VAR_PREFIX)TARGET_AR))"
diff --git a/modules/audio_processing/gen_aecm_core_neon_offsets_h.target.linux-arm.mk b/modules/audio_processing/gen_aecm_core_neon_offsets_h.target.linux-arm.mk
index d81e0c1..3b2e3de 100644
--- a/modules/audio_processing/gen_aecm_core_neon_offsets_h.target.linux-arm.mk
+++ b/modules/audio_processing/gen_aecm_core_neon_offsets_h.target.linux-arm.mk
@@ -23,7 +23,7 @@
$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets/aecm_core_neon_offsets.o: export PATH := $(subst $(ANDROID_BUILD_PATHS),,$(PATH))
$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets/aecm_core_neon_offsets.o: $(LOCAL_PATH)/third_party/libvpx/unpack_lib_posix.sh $(GYP_TARGET_DEPENDENCIES)
@echo "Gyp action: third_party_webrtc_modules_audio_processing_gen_core_neon_offsets_chromium_gyp_gen_aecm_core_neon_offsets_h_target_unpack_lib_posix ($@)"
- $(hide)cd $(gyp_local_path)/third_party/webrtc/modules/audio_processing; mkdir -p $(gyp_shared_intermediate_dir)/audio_processing/asm_offsets; ../../../../third_party/libvpx/unpack_lib_posix.sh -d "$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets" -f aecm_core_neon_offsets.o -a "$(gyp_shared_intermediate_dir)/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(PWD)/$(call intermediates-dir-for, STATIC_LIBRARIES, lib_core_neon_offsets,,, $(GYP_VAR_PREFIX))/lib_core_neon_offsets.a" -r "$(PWD)/$($(GYP_VAR_PREFIX)TARGET_AR)"
+ $(hide)cd $(gyp_local_path)/third_party/webrtc/modules/audio_processing; mkdir -p $(gyp_shared_intermediate_dir)/audio_processing/asm_offsets; ../../../../third_party/libvpx/unpack_lib_posix.sh -d "$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets" -f aecm_core_neon_offsets.o -a "$(gyp_shared_intermediate_dir)/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(realpath $(call intermediates-dir-for, STATIC_LIBRARIES, lib_core_neon_offsets,,, $(GYP_VAR_PREFIX)))/lib_core_neon_offsets.a" -r "$(realpath $($(GYP_VAR_PREFIX)TARGET_AR))"
diff --git a/modules/audio_processing/gen_nsx_core_neon_offsets_h.target.darwin-arm.mk b/modules/audio_processing/gen_nsx_core_neon_offsets_h.target.darwin-arm.mk
index 380d72d..d82ef39 100644
--- a/modules/audio_processing/gen_nsx_core_neon_offsets_h.target.darwin-arm.mk
+++ b/modules/audio_processing/gen_nsx_core_neon_offsets_h.target.darwin-arm.mk
@@ -23,7 +23,7 @@
$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets/nsx_core_neon_offsets.o: export PATH := $(subst $(ANDROID_BUILD_PATHS),,$(PATH))
$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets/nsx_core_neon_offsets.o: $(LOCAL_PATH)/third_party/libvpx/unpack_lib_posix.sh $(GYP_TARGET_DEPENDENCIES)
@echo "Gyp action: third_party_webrtc_modules_audio_processing_gen_core_neon_offsets_chromium_gyp_gen_nsx_core_neon_offsets_h_target_unpack_lib_posix ($@)"
- $(hide)cd $(gyp_local_path)/third_party/webrtc/modules/audio_processing; mkdir -p $(gyp_shared_intermediate_dir)/audio_processing/asm_offsets; ../../../../third_party/libvpx/unpack_lib_posix.sh -d "$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets" -f nsx_core_neon_offsets.o -a "$(gyp_shared_intermediate_dir)/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(PWD)/$(call intermediates-dir-for, STATIC_LIBRARIES, lib_core_neon_offsets,,, $(GYP_VAR_PREFIX))/lib_core_neon_offsets.a" -r "$(PWD)/$($(GYP_VAR_PREFIX)TARGET_AR)"
+ $(hide)cd $(gyp_local_path)/third_party/webrtc/modules/audio_processing; mkdir -p $(gyp_shared_intermediate_dir)/audio_processing/asm_offsets; ../../../../third_party/libvpx/unpack_lib_posix.sh -d "$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets" -f nsx_core_neon_offsets.o -a "$(gyp_shared_intermediate_dir)/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(realpath $(call intermediates-dir-for, STATIC_LIBRARIES, lib_core_neon_offsets,,, $(GYP_VAR_PREFIX)))/lib_core_neon_offsets.a" -r "$(realpath $($(GYP_VAR_PREFIX)TARGET_AR))"
diff --git a/modules/audio_processing/gen_nsx_core_neon_offsets_h.target.linux-arm.mk b/modules/audio_processing/gen_nsx_core_neon_offsets_h.target.linux-arm.mk
index 380d72d..d82ef39 100644
--- a/modules/audio_processing/gen_nsx_core_neon_offsets_h.target.linux-arm.mk
+++ b/modules/audio_processing/gen_nsx_core_neon_offsets_h.target.linux-arm.mk
@@ -23,7 +23,7 @@
$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets/nsx_core_neon_offsets.o: export PATH := $(subst $(ANDROID_BUILD_PATHS),,$(PATH))
$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets/nsx_core_neon_offsets.o: $(LOCAL_PATH)/third_party/libvpx/unpack_lib_posix.sh $(GYP_TARGET_DEPENDENCIES)
@echo "Gyp action: third_party_webrtc_modules_audio_processing_gen_core_neon_offsets_chromium_gyp_gen_nsx_core_neon_offsets_h_target_unpack_lib_posix ($@)"
- $(hide)cd $(gyp_local_path)/third_party/webrtc/modules/audio_processing; mkdir -p $(gyp_shared_intermediate_dir)/audio_processing/asm_offsets; ../../../../third_party/libvpx/unpack_lib_posix.sh -d "$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets" -f nsx_core_neon_offsets.o -a "$(gyp_shared_intermediate_dir)/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(PWD)/$(call intermediates-dir-for, STATIC_LIBRARIES, lib_core_neon_offsets,,, $(GYP_VAR_PREFIX))/lib_core_neon_offsets.a" -r "$(PWD)/$($(GYP_VAR_PREFIX)TARGET_AR)"
+ $(hide)cd $(gyp_local_path)/third_party/webrtc/modules/audio_processing; mkdir -p $(gyp_shared_intermediate_dir)/audio_processing/asm_offsets; ../../../../third_party/libvpx/unpack_lib_posix.sh -d "$(gyp_shared_intermediate_dir)/audio_processing/asm_offsets" -f nsx_core_neon_offsets.o -a "$(gyp_shared_intermediate_dir)/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(obj).$(TOOLSET)/third_party/webrtc/modules/audio_processing/lib_core_neon_offsets.a" -a "$(realpath $(call intermediates-dir-for, STATIC_LIBRARIES, lib_core_neon_offsets,,, $(GYP_VAR_PREFIX)))/lib_core_neon_offsets.a" -r "$(realpath $($(GYP_VAR_PREFIX)TARGET_AR))"
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 6db1d12..68b679d 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -61,7 +61,12 @@
bool enabled;
};
+#if defined(ANDROID) || defined(IOS)
+// AECM only supports 8kHz & 16kHz.
+static const int kAudioProcMaxNativeSampleRateHz = 16000;
+#else
static const int kAudioProcMaxNativeSampleRateHz = 32000;
+#endif
// The Audio Processing Module (APM) provides a collection of voice processing
// components designed for real-time communications software.
diff --git a/modules/audio_processing/lib_core_neon_offsets.gypi b/modules/audio_processing/lib_core_neon_offsets.gypi
index a4d6b74..ce257a8 100644
--- a/modules/audio_processing/lib_core_neon_offsets.gypi
+++ b/modules/audio_processing/lib_core_neon_offsets.gypi
@@ -14,7 +14,7 @@
'lib_intermediate_name': '',
'conditions' : [
['android_webview_build==1', {
- 'lib_intermediate_name' : '<(android_src)/$(call intermediates-dir-for, STATIC_LIBRARIES, lib_core_neon_offsets,,, $(GYP_VAR_PREFIX))/lib_core_neon_offsets.a',
+ 'lib_intermediate_name' : '$(realpath $(call intermediates-dir-for, STATIC_LIBRARIES, lib_core_neon_offsets,,, $(GYP_VAR_PREFIX)))/lib_core_neon_offsets.a',
}],
],
},
diff --git a/modules/audio_processing/test/audio_processing_unittest.cc b/modules/audio_processing/test/audio_processing_unittest.cc
index 257c05e..0c5b67d 100644
--- a/modules/audio_processing/test/audio_processing_unittest.cc
+++ b/modules/audio_processing/test/audio_processing_unittest.cc
@@ -81,6 +81,21 @@
ConvertToFloat(frame.data_, cb);
}
+// Number of channels including the keyboard channel.
+int TotalChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
+ switch (layout) {
+ case AudioProcessing::kMono:
+ return 1;
+ case AudioProcessing::kMonoAndKeyboard:
+ case AudioProcessing::kStereo:
+ return 2;
+ case AudioProcessing::kStereoAndKeyboard:
+ return 3;
+ }
+ assert(false);
+ return -1;
+}
+
int TruncateToMultipleOf10(int value) {
return (value / 10) * 10;
}
@@ -229,13 +244,15 @@
}
std::string OutputFilePath(std::string name,
- int sample_rate_hz,
+ int input_rate,
+ int output_rate,
+ int reverse_rate,
int num_input_channels,
int num_output_channels,
int num_reverse_channels) {
std::ostringstream ss;
- ss << name << sample_rate_hz / 1000 << "_" << num_reverse_channels << "r" <<
- num_input_channels << "i" << "_";
+ ss << name << "_i" << num_input_channels << "_" << input_rate / 1000
+ << "_r" << num_reverse_channels << "_" << reverse_rate / 1000 << "_";
if (num_output_channels == 1) {
ss << "mono";
} else if (num_output_channels == 2) {
@@ -243,7 +260,7 @@
} else {
assert(false);
}
- ss << ".pcm";
+ ss << output_rate / 1000 << ".pcm";
return test::OutputPath() + ss.str();
}
@@ -427,8 +444,13 @@
if (out_file_) {
ASSERT_EQ(0, fclose(out_file_));
}
- filename = OutputFilePath("out", sample_rate_hz, num_input_channels,
- num_output_channels, num_reverse_channels);
+ filename = OutputFilePath("out",
+ sample_rate_hz,
+ output_sample_rate_hz,
+ reverse_sample_rate_hz,
+ num_input_channels,
+ num_output_channels,
+ num_reverse_channels);
out_file_ = fopen(filename.c_str(), "wb");
ASSERT_TRUE(out_file_ != NULL) << "Could not open file " <<
filename << "\n";
@@ -1909,6 +1931,43 @@
#endif // WEBRTC_AUDIOPROC_BIT_EXACT
+TEST_F(ApmTest, NoErrorsWithKeyboardChannel) {
+ struct ChannelFormat {
+ AudioProcessing::ChannelLayout in_layout;
+ AudioProcessing::ChannelLayout out_layout;
+ };
+ ChannelFormat cf[] = {
+ {AudioProcessing::kMonoAndKeyboard, AudioProcessing::kMono},
+ {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kMono},
+ {AudioProcessing::kStereoAndKeyboard, AudioProcessing::kStereo},
+ };
+ size_t channel_format_size = sizeof(cf) / sizeof(*cf);
+
+ scoped_ptr<AudioProcessing> ap(AudioProcessing::Create());
+ // Enable one component just to ensure some processing takes place.
+ ap->noise_suppression()->Enable(true);
+ for (size_t i = 0; i < channel_format_size; ++i) {
+ const int in_rate = 44100;
+ const int out_rate = 48000;
+ ChannelBuffer<float> in_cb(SamplesFromRate(in_rate),
+ TotalChannelsFromLayout(cf[i].in_layout));
+ ChannelBuffer<float> out_cb(SamplesFromRate(out_rate),
+ ChannelsFromLayout(cf[i].out_layout));
+
+ // Run over a few chunks.
+ for (int j = 0; j < 10; ++j) {
+ EXPECT_NOERR(ap->ProcessStream(
+ in_cb.channels(),
+ in_cb.samples_per_channel(),
+ in_rate,
+ cf[i].in_layout,
+ out_rate,
+ cf[i].out_layout,
+ out_cb.channels()));
+ }
+ }
+}
+
// Reads a 10 ms chunk of int16 interleaved audio from the given (assumed
// stereo) file, converts to deinterleaved float (optionally downmixing) and
// returns the result in |cb|. Returns false if the file ended (or on error) and
@@ -2046,7 +2105,9 @@
FILE* far_file = fopen(ResourceFilePath("far", reverse_rate).c_str(), "rb");
FILE* near_file = fopen(ResourceFilePath("near", input_rate).c_str(), "rb");
FILE* out_file = fopen(OutputFilePath(output_file_prefix,
+ input_rate,
output_rate,
+ reverse_rate,
num_input_channels,
num_output_channels,
num_reverse_channels).c_str(), "wb");
@@ -2152,13 +2213,17 @@
#endif
FILE* out_file = fopen(OutputFilePath("out",
+ input_rate_,
output_rate_,
+ reverse_rate_,
cf[i].num_input,
cf[i].num_output,
cf[i].num_reverse).c_str(), "rb");
// The reference files always have matching input and output channels.
FILE* ref_file = fopen(OutputFilePath("ref",
ref_rate,
+ ref_rate,
+ ref_rate,
cf[i].num_output,
cf[i].num_output,
cf[i].num_reverse).c_str(), "rb");
diff --git a/modules/audio_processing/test/process_test.cc b/modules/audio_processing/test/process_test.cc
index 95984f5..01f9dce 100644
--- a/modules/audio_processing/test/process_test.cc
+++ b/modules/audio_processing/test/process_test.cc
@@ -518,7 +518,7 @@
const size_t path_size =
apm->echo_control_mobile()->echo_path_size_bytes();
- scoped_array<char> echo_path(new char[path_size]);
+ scoped_ptr<char[]> echo_path(new char[path_size]);
ASSERT_EQ(path_size, fread(echo_path.get(),
sizeof(char),
path_size,
@@ -1004,7 +1004,7 @@
if (aecm_echo_path_out_file != NULL) {
const size_t path_size =
apm->echo_control_mobile()->echo_path_size_bytes();
- scoped_array<char> echo_path(new char[path_size]);
+ scoped_ptr<char[]> echo_path(new char[path_size]);
apm->echo_control_mobile()->GetEchoPath(echo_path.get(), path_size);
ASSERT_EQ(path_size, fwrite(echo_path.get(),
sizeof(char),
diff --git a/modules/audio_processing/utility/ring_buffer_unittest.cc b/modules/audio_processing/utility/ring_buffer_unittest.cc
index bd68f94..5dacf0b 100644
--- a/modules/audio_processing/utility/ring_buffer_unittest.cc
+++ b/modules/audio_processing/utility/ring_buffer_unittest.cc
@@ -61,8 +61,8 @@
srand(seed);
for (int i = 0; i < kNumTests; i++) {
const int buffer_size = std::max(rand() % kMaxBufferSize, 1);
- scoped_array<int> write_data(new int[buffer_size]);
- scoped_array<int> read_data(new int[buffer_size]);
+ scoped_ptr<int[]> write_data(new int[buffer_size]);
+ scoped_ptr<int[]> read_data(new int[buffer_size]);
scoped_ring_buffer buffer(WebRtc_CreateBuffer(buffer_size, sizeof(int)));
ASSERT_TRUE(buffer.get() != NULL);
ASSERT_EQ(0, WebRtc_InitBuffer(buffer.get()));
diff --git a/modules/audioproc_debug_proto.target.darwin-arm64.mk b/modules/audioproc_debug_proto.target.darwin-arm64.mk
new file mode 100644
index 0000000..d16a03a
--- /dev/null
+++ b/modules/audioproc_debug_proto.target.darwin-arm64.mk
@@ -0,0 +1,312 @@
+# This file is generated by gyp; do not edit.
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := third_party_webrtc_modules_audioproc_debug_proto_gyp
+LOCAL_MODULE_SUFFIX := .a
+LOCAL_MODULE_TAGS := optional
+LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
+gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
+gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
+
+# Make sure our deps are built first.
+GYP_TARGET_DEPENDENCIES := \
+ $(gyp_shared_intermediate_dir)/protoc
+
+
+### Generated for rule "third_party_webrtc_modules_modules_gyp_audioproc_debug_proto_target_genproto":
+# "{'inputs': ['../../../tools/protoc_wrapper/protoc_wrapper.py', '$(gyp_shared_intermediate_dir)/protoc'], 'process_outputs_as_sources': '1', 'extension': 'proto', 'outputs': ['$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/%(INPUT_ROOT)s_pb2.py', '$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/%(INPUT_ROOT)s.pb.cc', '$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/%(INPUT_ROOT)s.pb.h'], 'rule_name': 'genproto', 'rule_sources': ['audio_processing/debug.proto'], 'action': ['python', '../../../tools/protoc_wrapper/protoc_wrapper.py', '--include', '', '--protobuf', '$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/%(INPUT_ROOT)s.pb.h', '--proto-in-dir', 'audio_processing', '--proto-in-file', '%(INPUT_ROOT)s$(suffix $<)', '--use-system-protobuf=0', '--', '$(gyp_shared_intermediate_dir)/protoc', '--cpp_out', '$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing', '--python_out', '$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing'], 'message': 'Generating C++ and Python code from $(RULE_SOURCES)'}":
+$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py: gyp_local_path := $(LOCAL_PATH)
+$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py: gyp_intermediate_dir := $(abspath $(gyp_intermediate_dir))
+$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py: gyp_shared_intermediate_dir := $(abspath $(gyp_shared_intermediate_dir))
+$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py: export PATH := $(subst $(ANDROID_BUILD_PATHS),,$(PATH))
+$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py: $(LOCAL_PATH)/third_party/webrtc/modules/audio_processing/debug.proto $(LOCAL_PATH)/tools/protoc_wrapper/protoc_wrapper.py $(gyp_shared_intermediate_dir)/protoc $(GYP_TARGET_DEPENDENCIES)
+ mkdir -p $(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing; cd $(gyp_local_path)/third_party/webrtc/modules; python ../../../tools/protoc_wrapper/protoc_wrapper.py --include "" --protobuf "$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.h" --proto-in-dir audio_processing --proto-in-file "debug$(suffix $<)" "--use-system-protobuf=0" -- "$(gyp_shared_intermediate_dir)/protoc" --cpp_out "$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing" --python_out "$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing"
+
+$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.cc: $(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py ;
+$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.h: $(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py ;
+
+
+GYP_GENERATED_OUTPUTS := \
+ $(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py \
+ $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.cc \
+ $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.h
+
+# Make sure our deps and generated files are built first.
+LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+
+LOCAL_CPP_EXTENSION := .cc
+$(gyp_intermediate_dir)/debug.pb.cc: $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.cc
+ mkdir -p $(@D); cp $< $@
+LOCAL_GENERATED_SOURCES := \
+ $(gyp_intermediate_dir)/debug.pb.cc \
+ $(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py \
+ $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.h
+
+GYP_COPIED_SOURCE_ORIGIN_DIRS := \
+ $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing
+
+LOCAL_SRC_FILES :=
+
+
+# Flags passed to both C and C++ files.
+MY_CFLAGS_Debug := \
+ --param=ssp-buffer-size=4 \
+ -Werror \
+ -fno-exceptions \
+ -fno-strict-aliasing \
+ -Wall \
+ -Wno-unused-parameter \
+ -Wno-missing-field-initializers \
+ -fvisibility=hidden \
+ -pipe \
+ -fPIC \
+ -Wno-unused-local-typedefs \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -ffunction-sections \
+ -funwind-tables \
+ -g \
+ -fno-short-enums \
+ -finline-limit=64 \
+ -Wa,--noexecstack \
+ -U_FORTIFY_SOURCE \
+ -Wno-extra \
+ -Wno-ignored-qualifiers \
+ -Wno-type-limits \
+ -Wno-unused-but-set-variable \
+ -Os \
+ -g \
+ -fomit-frame-pointer \
+ -fdata-sections \
+ -ffunction-sections \
+ -funwind-tables
+
+MY_DEFS_Debug := \
+ '-DV8_DEPRECATION_WARNINGS' \
+ '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
+ '-D_FILE_OFFSET_BITS=64' \
+ '-DNO_TCMALLOC' \
+ '-DDISABLE_NACL' \
+ '-DCHROMIUM_BUILD' \
+ '-DUSE_LIBJPEG_TURBO=1' \
+ '-DENABLE_WEBRTC=1' \
+ '-DUSE_PROPRIETARY_CODECS' \
+ '-DENABLE_CONFIGURATION_POLICY' \
+ '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
+ '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
+ '-DENABLE_EGLIMAGE=1' \
+ '-DCLD_VERSION=1' \
+ '-DENABLE_PRINTING=1' \
+ '-DENABLE_MANAGED_USERS=1' \
+ '-DWEBRTC_RESTRICT_LOGGING' \
+ '-DWEBRTC_MODULE_UTILITY_VIDEO' \
+ '-DWEBRTC_CHROMIUM_BUILD' \
+ '-DWEBRTC_POSIX' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_ANDROID' \
+ '-DWEBRTC_ANDROID_OPENSLES' \
+ '-DPROTOBUF_USE_DLLS' \
+ '-DGOOGLE_PROTOBUF_NO_RTTI' \
+ '-DGOOGLE_PROTOBUF_NO_STATIC_INITIALIZER' \
+ '-DUSE_OPENSSL=1' \
+ '-DUSE_OPENSSL_CERTS=1' \
+ '-D__STDC_CONSTANT_MACROS' \
+ '-D__STDC_FORMAT_MACROS' \
+ '-DANDROID' \
+ '-D__GNU_SOURCE=1' \
+ '-DUSE_STLPORT=1' \
+ '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
+ '-DCHROME_BUILD_ID=""' \
+ '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
+ '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
+ '-D_DEBUG'
+
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES_Debug := \
+ $(LOCAL_PATH)/third_party \
+ $(LOCAL_PATH) \
+ $(gyp_shared_intermediate_dir)/protoc_out \
+ $(LOCAL_PATH)/third_party/protobuf \
+ $(LOCAL_PATH)/third_party/protobuf/src \
+ $(PWD)/frameworks/wilhelm/include \
+ $(PWD)/bionic \
+ $(PWD)/external/stlport/stlport
+
+
+# Flags passed to only C++ (and not C) files.
+LOCAL_CPPFLAGS_Debug := \
+ -fno-rtti \
+ -fno-threadsafe-statics \
+ -fvisibility-inlines-hidden \
+ -Wsign-compare \
+ -Wno-non-virtual-dtor \
+ -Wno-sign-promo
+
+
+# Flags passed to both C and C++ files.
+MY_CFLAGS_Release := \
+ --param=ssp-buffer-size=4 \
+ -Werror \
+ -fno-exceptions \
+ -fno-strict-aliasing \
+ -Wall \
+ -Wno-unused-parameter \
+ -Wno-missing-field-initializers \
+ -fvisibility=hidden \
+ -pipe \
+ -fPIC \
+ -Wno-unused-local-typedefs \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -ffunction-sections \
+ -funwind-tables \
+ -g \
+ -fno-short-enums \
+ -finline-limit=64 \
+ -Wa,--noexecstack \
+ -U_FORTIFY_SOURCE \
+ -Wno-extra \
+ -Wno-ignored-qualifiers \
+ -Wno-type-limits \
+ -Wno-unused-but-set-variable \
+ -Os \
+ -fno-ident \
+ -fdata-sections \
+ -ffunction-sections \
+ -fomit-frame-pointer \
+ -funwind-tables
+
+MY_DEFS_Release := \
+ '-DV8_DEPRECATION_WARNINGS' \
+ '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
+ '-D_FILE_OFFSET_BITS=64' \
+ '-DNO_TCMALLOC' \
+ '-DDISABLE_NACL' \
+ '-DCHROMIUM_BUILD' \
+ '-DUSE_LIBJPEG_TURBO=1' \
+ '-DENABLE_WEBRTC=1' \
+ '-DUSE_PROPRIETARY_CODECS' \
+ '-DENABLE_CONFIGURATION_POLICY' \
+ '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
+ '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
+ '-DENABLE_EGLIMAGE=1' \
+ '-DCLD_VERSION=1' \
+ '-DENABLE_PRINTING=1' \
+ '-DENABLE_MANAGED_USERS=1' \
+ '-DWEBRTC_RESTRICT_LOGGING' \
+ '-DWEBRTC_MODULE_UTILITY_VIDEO' \
+ '-DWEBRTC_CHROMIUM_BUILD' \
+ '-DWEBRTC_POSIX' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_ANDROID' \
+ '-DWEBRTC_ANDROID_OPENSLES' \
+ '-DPROTOBUF_USE_DLLS' \
+ '-DGOOGLE_PROTOBUF_NO_RTTI' \
+ '-DGOOGLE_PROTOBUF_NO_STATIC_INITIALIZER' \
+ '-DUSE_OPENSSL=1' \
+ '-DUSE_OPENSSL_CERTS=1' \
+ '-D__STDC_CONSTANT_MACROS' \
+ '-D__STDC_FORMAT_MACROS' \
+ '-DANDROID' \
+ '-D__GNU_SOURCE=1' \
+ '-DUSE_STLPORT=1' \
+ '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
+ '-DCHROME_BUILD_ID=""' \
+ '-DNDEBUG' \
+ '-DNVALGRIND' \
+ '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
+ '-D_FORTIFY_SOURCE=2'
+
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES_Release := \
+ $(LOCAL_PATH)/third_party \
+ $(LOCAL_PATH) \
+ $(gyp_shared_intermediate_dir)/protoc_out \
+ $(LOCAL_PATH)/third_party/protobuf \
+ $(LOCAL_PATH)/third_party/protobuf/src \
+ $(PWD)/frameworks/wilhelm/include \
+ $(PWD)/bionic \
+ $(PWD)/external/stlport/stlport
+
+
+# Flags passed to only C++ (and not C) files.
+LOCAL_CPPFLAGS_Release := \
+ -fno-rtti \
+ -fno-threadsafe-statics \
+ -fvisibility-inlines-hidden \
+ -Wsign-compare \
+ -Wno-non-virtual-dtor \
+ -Wno-sign-promo
+
+
+LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
+LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
+LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
+LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
+### Rules for final target.
+
+LOCAL_LDFLAGS_Debug := \
+ -Wl,-z,now \
+ -Wl,-z,relro \
+ -Wl,--fatal-warnings \
+ -Wl,-z,noexecstack \
+ -fPIC \
+ -nostdlib \
+ -Wl,--no-undefined \
+ -Wl,--exclude-libs=ALL \
+ -Wl,--warn-shared-textrel \
+ -Wl,-O1 \
+ -Wl,--as-needed
+
+
+LOCAL_LDFLAGS_Release := \
+ -Wl,-z,now \
+ -Wl,-z,relro \
+ -Wl,--fatal-warnings \
+ -Wl,-z,noexecstack \
+ -fPIC \
+ -nostdlib \
+ -Wl,--no-undefined \
+ -Wl,--exclude-libs=ALL \
+ -Wl,-O1 \
+ -Wl,--as-needed \
+ -Wl,--gc-sections \
+ -Wl,--warn-shared-textrel
+
+
+LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
+
+LOCAL_STATIC_LIBRARIES :=
+
+# Enable grouping to fix circular references
+LOCAL_GROUP_STATIC_LIBRARIES := true
+
+LOCAL_SHARED_LIBRARIES := \
+ libstlport \
+ libdl
+
+# Add target alias to "gyp_all_modules" target.
+.PHONY: gyp_all_modules
+gyp_all_modules: third_party_webrtc_modules_audioproc_debug_proto_gyp
+
+# Alias gyp target name.
+.PHONY: audioproc_debug_proto
+audioproc_debug_proto: third_party_webrtc_modules_audioproc_debug_proto_gyp
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/audioproc_debug_proto.target.linux-arm64.mk b/modules/audioproc_debug_proto.target.linux-arm64.mk
new file mode 100644
index 0000000..d16a03a
--- /dev/null
+++ b/modules/audioproc_debug_proto.target.linux-arm64.mk
@@ -0,0 +1,312 @@
+# This file is generated by gyp; do not edit.
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := third_party_webrtc_modules_audioproc_debug_proto_gyp
+LOCAL_MODULE_SUFFIX := .a
+LOCAL_MODULE_TAGS := optional
+LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
+gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
+gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
+
+# Make sure our deps are built first.
+GYP_TARGET_DEPENDENCIES := \
+ $(gyp_shared_intermediate_dir)/protoc
+
+
+### Generated for rule "third_party_webrtc_modules_modules_gyp_audioproc_debug_proto_target_genproto":
+# "{'inputs': ['../../../tools/protoc_wrapper/protoc_wrapper.py', '$(gyp_shared_intermediate_dir)/protoc'], 'process_outputs_as_sources': '1', 'extension': 'proto', 'outputs': ['$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/%(INPUT_ROOT)s_pb2.py', '$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/%(INPUT_ROOT)s.pb.cc', '$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/%(INPUT_ROOT)s.pb.h'], 'rule_name': 'genproto', 'rule_sources': ['audio_processing/debug.proto'], 'action': ['python', '../../../tools/protoc_wrapper/protoc_wrapper.py', '--include', '', '--protobuf', '$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/%(INPUT_ROOT)s.pb.h', '--proto-in-dir', 'audio_processing', '--proto-in-file', '%(INPUT_ROOT)s$(suffix $<)', '--use-system-protobuf=0', '--', '$(gyp_shared_intermediate_dir)/protoc', '--cpp_out', '$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing', '--python_out', '$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing'], 'message': 'Generating C++ and Python code from $(RULE_SOURCES)'}":
+$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py: gyp_local_path := $(LOCAL_PATH)
+$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py: gyp_intermediate_dir := $(abspath $(gyp_intermediate_dir))
+$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py: gyp_shared_intermediate_dir := $(abspath $(gyp_shared_intermediate_dir))
+$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py: export PATH := $(subst $(ANDROID_BUILD_PATHS),,$(PATH))
+$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py: $(LOCAL_PATH)/third_party/webrtc/modules/audio_processing/debug.proto $(LOCAL_PATH)/tools/protoc_wrapper/protoc_wrapper.py $(gyp_shared_intermediate_dir)/protoc $(GYP_TARGET_DEPENDENCIES)
+ mkdir -p $(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing; cd $(gyp_local_path)/third_party/webrtc/modules; python ../../../tools/protoc_wrapper/protoc_wrapper.py --include "" --protobuf "$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.h" --proto-in-dir audio_processing --proto-in-file "debug$(suffix $<)" "--use-system-protobuf=0" -- "$(gyp_shared_intermediate_dir)/protoc" --cpp_out "$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing" --python_out "$(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing"
+
+$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.cc: $(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py ;
+$(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.h: $(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py ;
+
+
+GYP_GENERATED_OUTPUTS := \
+ $(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py \
+ $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.cc \
+ $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.h
+
+# Make sure our deps and generated files are built first.
+LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+
+LOCAL_CPP_EXTENSION := .cc
+$(gyp_intermediate_dir)/debug.pb.cc: $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.cc
+ mkdir -p $(@D); cp $< $@
+LOCAL_GENERATED_SOURCES := \
+ $(gyp_intermediate_dir)/debug.pb.cc \
+ $(gyp_shared_intermediate_dir)/pyproto/webrtc/audio_processing/debug_pb2.py \
+ $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing/debug.pb.h
+
+GYP_COPIED_SOURCE_ORIGIN_DIRS := \
+ $(gyp_shared_intermediate_dir)/protoc_out/webrtc/audio_processing
+
+LOCAL_SRC_FILES :=
+
+
+# Flags passed to both C and C++ files.
+MY_CFLAGS_Debug := \
+ --param=ssp-buffer-size=4 \
+ -Werror \
+ -fno-exceptions \
+ -fno-strict-aliasing \
+ -Wall \
+ -Wno-unused-parameter \
+ -Wno-missing-field-initializers \
+ -fvisibility=hidden \
+ -pipe \
+ -fPIC \
+ -Wno-unused-local-typedefs \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -ffunction-sections \
+ -funwind-tables \
+ -g \
+ -fno-short-enums \
+ -finline-limit=64 \
+ -Wa,--noexecstack \
+ -U_FORTIFY_SOURCE \
+ -Wno-extra \
+ -Wno-ignored-qualifiers \
+ -Wno-type-limits \
+ -Wno-unused-but-set-variable \
+ -Os \
+ -g \
+ -fomit-frame-pointer \
+ -fdata-sections \
+ -ffunction-sections \
+ -funwind-tables
+
+MY_DEFS_Debug := \
+ '-DV8_DEPRECATION_WARNINGS' \
+ '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
+ '-D_FILE_OFFSET_BITS=64' \
+ '-DNO_TCMALLOC' \
+ '-DDISABLE_NACL' \
+ '-DCHROMIUM_BUILD' \
+ '-DUSE_LIBJPEG_TURBO=1' \
+ '-DENABLE_WEBRTC=1' \
+ '-DUSE_PROPRIETARY_CODECS' \
+ '-DENABLE_CONFIGURATION_POLICY' \
+ '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
+ '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
+ '-DENABLE_EGLIMAGE=1' \
+ '-DCLD_VERSION=1' \
+ '-DENABLE_PRINTING=1' \
+ '-DENABLE_MANAGED_USERS=1' \
+ '-DWEBRTC_RESTRICT_LOGGING' \
+ '-DWEBRTC_MODULE_UTILITY_VIDEO' \
+ '-DWEBRTC_CHROMIUM_BUILD' \
+ '-DWEBRTC_POSIX' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_ANDROID' \
+ '-DWEBRTC_ANDROID_OPENSLES' \
+ '-DPROTOBUF_USE_DLLS' \
+ '-DGOOGLE_PROTOBUF_NO_RTTI' \
+ '-DGOOGLE_PROTOBUF_NO_STATIC_INITIALIZER' \
+ '-DUSE_OPENSSL=1' \
+ '-DUSE_OPENSSL_CERTS=1' \
+ '-D__STDC_CONSTANT_MACROS' \
+ '-D__STDC_FORMAT_MACROS' \
+ '-DANDROID' \
+ '-D__GNU_SOURCE=1' \
+ '-DUSE_STLPORT=1' \
+ '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
+ '-DCHROME_BUILD_ID=""' \
+ '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
+ '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
+ '-D_DEBUG'
+
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES_Debug := \
+ $(LOCAL_PATH)/third_party \
+ $(LOCAL_PATH) \
+ $(gyp_shared_intermediate_dir)/protoc_out \
+ $(LOCAL_PATH)/third_party/protobuf \
+ $(LOCAL_PATH)/third_party/protobuf/src \
+ $(PWD)/frameworks/wilhelm/include \
+ $(PWD)/bionic \
+ $(PWD)/external/stlport/stlport
+
+
+# Flags passed to only C++ (and not C) files.
+LOCAL_CPPFLAGS_Debug := \
+ -fno-rtti \
+ -fno-threadsafe-statics \
+ -fvisibility-inlines-hidden \
+ -Wsign-compare \
+ -Wno-non-virtual-dtor \
+ -Wno-sign-promo
+
+
+# Flags passed to both C and C++ files.
+MY_CFLAGS_Release := \
+ --param=ssp-buffer-size=4 \
+ -Werror \
+ -fno-exceptions \
+ -fno-strict-aliasing \
+ -Wall \
+ -Wno-unused-parameter \
+ -Wno-missing-field-initializers \
+ -fvisibility=hidden \
+ -pipe \
+ -fPIC \
+ -Wno-unused-local-typedefs \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -ffunction-sections \
+ -funwind-tables \
+ -g \
+ -fno-short-enums \
+ -finline-limit=64 \
+ -Wa,--noexecstack \
+ -U_FORTIFY_SOURCE \
+ -Wno-extra \
+ -Wno-ignored-qualifiers \
+ -Wno-type-limits \
+ -Wno-unused-but-set-variable \
+ -Os \
+ -fno-ident \
+ -fdata-sections \
+ -ffunction-sections \
+ -fomit-frame-pointer \
+ -funwind-tables
+
+MY_DEFS_Release := \
+ '-DV8_DEPRECATION_WARNINGS' \
+ '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
+ '-D_FILE_OFFSET_BITS=64' \
+ '-DNO_TCMALLOC' \
+ '-DDISABLE_NACL' \
+ '-DCHROMIUM_BUILD' \
+ '-DUSE_LIBJPEG_TURBO=1' \
+ '-DENABLE_WEBRTC=1' \
+ '-DUSE_PROPRIETARY_CODECS' \
+ '-DENABLE_CONFIGURATION_POLICY' \
+ '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
+ '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
+ '-DENABLE_EGLIMAGE=1' \
+ '-DCLD_VERSION=1' \
+ '-DENABLE_PRINTING=1' \
+ '-DENABLE_MANAGED_USERS=1' \
+ '-DWEBRTC_RESTRICT_LOGGING' \
+ '-DWEBRTC_MODULE_UTILITY_VIDEO' \
+ '-DWEBRTC_CHROMIUM_BUILD' \
+ '-DWEBRTC_POSIX' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_ANDROID' \
+ '-DWEBRTC_ANDROID_OPENSLES' \
+ '-DPROTOBUF_USE_DLLS' \
+ '-DGOOGLE_PROTOBUF_NO_RTTI' \
+ '-DGOOGLE_PROTOBUF_NO_STATIC_INITIALIZER' \
+ '-DUSE_OPENSSL=1' \
+ '-DUSE_OPENSSL_CERTS=1' \
+ '-D__STDC_CONSTANT_MACROS' \
+ '-D__STDC_FORMAT_MACROS' \
+ '-DANDROID' \
+ '-D__GNU_SOURCE=1' \
+ '-DUSE_STLPORT=1' \
+ '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
+ '-DCHROME_BUILD_ID=""' \
+ '-DNDEBUG' \
+ '-DNVALGRIND' \
+ '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
+ '-D_FORTIFY_SOURCE=2'
+
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES_Release := \
+ $(LOCAL_PATH)/third_party \
+ $(LOCAL_PATH) \
+ $(gyp_shared_intermediate_dir)/protoc_out \
+ $(LOCAL_PATH)/third_party/protobuf \
+ $(LOCAL_PATH)/third_party/protobuf/src \
+ $(PWD)/frameworks/wilhelm/include \
+ $(PWD)/bionic \
+ $(PWD)/external/stlport/stlport
+
+
+# Flags passed to only C++ (and not C) files.
+LOCAL_CPPFLAGS_Release := \
+ -fno-rtti \
+ -fno-threadsafe-statics \
+ -fvisibility-inlines-hidden \
+ -Wsign-compare \
+ -Wno-non-virtual-dtor \
+ -Wno-sign-promo
+
+
+LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
+LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
+LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
+LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
+### Rules for final target.
+
+LOCAL_LDFLAGS_Debug := \
+ -Wl,-z,now \
+ -Wl,-z,relro \
+ -Wl,--fatal-warnings \
+ -Wl,-z,noexecstack \
+ -fPIC \
+ -nostdlib \
+ -Wl,--no-undefined \
+ -Wl,--exclude-libs=ALL \
+ -Wl,--warn-shared-textrel \
+ -Wl,-O1 \
+ -Wl,--as-needed
+
+
+LOCAL_LDFLAGS_Release := \
+ -Wl,-z,now \
+ -Wl,-z,relro \
+ -Wl,--fatal-warnings \
+ -Wl,-z,noexecstack \
+ -fPIC \
+ -nostdlib \
+ -Wl,--no-undefined \
+ -Wl,--exclude-libs=ALL \
+ -Wl,-O1 \
+ -Wl,--as-needed \
+ -Wl,--gc-sections \
+ -Wl,--warn-shared-textrel
+
+
+LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
+
+LOCAL_STATIC_LIBRARIES :=
+
+# Enable grouping to fix circular references
+LOCAL_GROUP_STATIC_LIBRARIES := true
+
+LOCAL_SHARED_LIBRARIES := \
+ libstlport \
+ libdl
+
+# Add target alias to "gyp_all_modules" target.
+.PHONY: gyp_all_modules
+gyp_all_modules: third_party_webrtc_modules_audioproc_debug_proto_gyp
+
+# Alias gyp target name.
+.PHONY: audioproc_debug_proto
+audioproc_debug_proto: third_party_webrtc_modules_audioproc_debug_proto_gyp
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/bitrate_controller.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/bitrate_controller.target.darwin-arm64.mk
index c84869c..9ab4c78 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/bitrate_controller.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_bitrate_controller_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,18 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc \
+ third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +49,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +91,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +109,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +127,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +146,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +188,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +207,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +235,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +249,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +271,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_bitrate_controller_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: bitrate_controller
+bitrate_controller: third_party_webrtc_modules_bitrate_controller_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/bitrate_controller.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/bitrate_controller.target.linux-arm64.mk
index c84869c..9ab4c78 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/bitrate_controller.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_bitrate_controller_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,18 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc \
+ third_party/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +49,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +91,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +109,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +127,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +146,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +188,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +207,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +235,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +249,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +271,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_bitrate_controller_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: bitrate_controller
+bitrate_controller: third_party_webrtc_modules_bitrate_controller_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/desktop_capture/differ.h b/modules/desktop_capture/differ.h
index 8edce80..0b419d2 100644
--- a/modules/desktop_capture/differ.h
+++ b/modules/desktop_capture/differ.h
@@ -76,7 +76,7 @@
int bytes_per_row_;
// Diff information for each block in the image.
- scoped_array<DiffInfo> diff_info_;
+ scoped_ptr<DiffInfo[]> diff_info_;
// Dimensions and total size of diff info array.
int diff_info_width_;
diff --git a/modules/desktop_capture/differ_unittest.cc b/modules/desktop_capture/differ_unittest.cc
index 40fde4d..da1a214 100644
--- a/modules/desktop_capture/differ_unittest.cc
+++ b/modules/desktop_capture/differ_unittest.cc
@@ -200,8 +200,8 @@
int buffer_size_;
// Previous and current screen buffers.
- scoped_array<uint8_t> prev_;
- scoped_array<uint8_t> curr_;
+ scoped_ptr<uint8_t[]> prev_;
+ scoped_ptr<uint8_t[]> curr_;
private:
DISALLOW_COPY_AND_ASSIGN(DifferTest);
diff --git a/modules/desktop_capture/mouse_cursor.cc b/modules/desktop_capture/mouse_cursor.cc
index 07c89f0..22a9c0e 100644
--- a/modules/desktop_capture/mouse_cursor.cc
+++ b/modules/desktop_capture/mouse_cursor.cc
@@ -10,6 +10,8 @@
#include "webrtc/modules/desktop_capture/mouse_cursor.h"
+#include <assert.h>
+
#include "webrtc/modules/desktop_capture/desktop_frame.h"
namespace webrtc {
diff --git a/modules/desktop_capture/screen_capture_frame_queue.cc b/modules/desktop_capture/screen_capture_frame_queue.cc
index b045f05..45a3507 100644
--- a/modules/desktop_capture/screen_capture_frame_queue.cc
+++ b/modules/desktop_capture/screen_capture_frame_queue.cc
@@ -10,6 +10,7 @@
#include "webrtc/modules/desktop_capture/screen_capture_frame_queue.h"
+#include <assert.h>
#include <algorithm>
#include "webrtc/modules/desktop_capture/desktop_frame.h"
diff --git a/modules/desktop_capture/screen_capturer_helper.cc b/modules/desktop_capture/screen_capturer_helper.cc
index 75af043..86761c1 100644
--- a/modules/desktop_capture/screen_capturer_helper.cc
+++ b/modules/desktop_capture/screen_capturer_helper.cc
@@ -10,6 +10,7 @@
#include "webrtc/modules/desktop_capture/screen_capturer_helper.h"
+#include <assert.h>
#include <algorithm>
#include "webrtc/system_wrappers/interface/logging.h"
diff --git a/modules/desktop_capture/win/cursor.cc b/modules/desktop_capture/win/cursor.cc
index 11bb2db..00055c4 100644
--- a/modules/desktop_capture/win/cursor.cc
+++ b/modules/desktop_capture/win/cursor.cc
@@ -137,7 +137,7 @@
int width = bitmap_info.bmWidth;
int height = bitmap_info.bmHeight;
- scoped_array<uint32_t> mask_data(new uint32_t[width * height]);
+ scoped_ptr<uint32_t[]> mask_data(new uint32_t[width * height]);
// Get pixel data from |scoped_mask| converting it to 32bpp along the way.
// GetDIBits() sets the alpha component of every pixel to 0.
diff --git a/modules/desktop_capture/win/cursor_unittest.cc b/modules/desktop_capture/win/cursor_unittest.cc
index 9d23874..b046ace 100644
--- a/modules/desktop_capture/win/cursor_unittest.cc
+++ b/modules/desktop_capture/win/cursor_unittest.cc
@@ -62,7 +62,7 @@
// Get the pixels from |scoped_color|.
int size = width * height;
- scoped_array<uint32_t> data(new uint32_t[size]);
+ scoped_ptr<uint32_t[]> data(new uint32_t[size]);
EXPECT_TRUE(GetBitmapBits(scoped_color, size * sizeof(uint32_t), data.get()));
// Compare the 32bpp image in |mouse_shape| with the one loaded from |right|.
diff --git a/modules/desktop_capture/window_capturer_mac.mm b/modules/desktop_capture/window_capturer_mac.mm
index 3ceae31..d177fc4 100644
--- a/modules/desktop_capture/window_capturer_mac.mm
+++ b/modules/desktop_capture/window_capturer_mac.mm
@@ -42,6 +42,18 @@
return true;
}
+bool IsWindowValid(CGWindowID id) {
+ CFArrayRef window_id_array =
+ CFArrayCreate(NULL, reinterpret_cast<const void **>(&id), 1, NULL);
+ CFArrayRef window_array =
+ CGWindowListCreateDescriptionFromArray(window_id_array);
+ bool valid = window_array && CFArrayGetCount(window_array);
+ CFRelease(window_id_array);
+ CFRelease(window_array);
+
+ return valid;
+}
+
class WindowCapturerMac : public WindowCapturer {
public:
WindowCapturerMac();
@@ -115,22 +127,8 @@
}
bool WindowCapturerMac::SelectWindow(WindowId id) {
- // Request description for the specified window to make sure |id| is valid.
- CGWindowID ids[1];
- ids[0] = id;
- CFArrayRef window_id_array =
- CFArrayCreate(NULL, reinterpret_cast<const void **>(&ids), 1, NULL);
- CFArrayRef window_array =
- CGWindowListCreateDescriptionFromArray(window_id_array);
- int results_count = window_array ? CFArrayGetCount(window_array) : 0;
- CFRelease(window_id_array);
- CFRelease(window_array);
-
- if (results_count == 0) {
- // Could not find the window. It might have been closed.
+ if (!IsWindowValid(id))
return false;
- }
-
window_id_ = id;
return true;
}
@@ -180,6 +178,11 @@
}
void WindowCapturerMac::Capture(const DesktopRegion& region) {
+ if (!IsWindowValid(window_id_)) {
+ callback_->OnCaptureCompleted(NULL);
+ return;
+ }
+
CGImageRef window_image = CGWindowListCreateImage(
CGRectNull, kCGWindowListOptionIncludingWindow,
window_id_, kCGWindowImageBoundsIgnoreFraming);
diff --git a/modules/desktop_capture/window_capturer_win.cc b/modules/desktop_capture/window_capturer_win.cc
index b9764b3..a002185 100644
--- a/modules/desktop_capture/window_capturer_win.cc
+++ b/modules/desktop_capture/window_capturer_win.cc
@@ -182,8 +182,8 @@
return;
}
- // Stop capturing if the window has been minimized or hidden.
- if (IsIconic(window_) || !IsWindowVisible(window_)) {
+ // Stop capturing if the window has been closed or hidden.
+ if (!IsWindow(window_) || !IsWindowVisible(window_)) {
callback_->OnCaptureCompleted(NULL);
return;
}
diff --git a/modules/desktop_capture/window_capturer_x11.cc b/modules/desktop_capture/window_capturer_x11.cc
index baeb894..b641c93 100755
--- a/modules/desktop_capture/window_capturer_x11.cc
+++ b/modules/desktop_capture/window_capturer_x11.cc
@@ -278,6 +278,12 @@
}
void WindowCapturerLinux::Capture(const DesktopRegion& region) {
+ if (!x_server_pixel_buffer_.IsWindowValid()) {
+ LOG(LS_INFO) << "The window is no longer valid.";
+ callback_->OnCaptureCompleted(NULL);
+ return;
+ }
+
x_display_->ProcessPendingXEvents();
if (!has_composite_extension_) {
diff --git a/modules/desktop_capture/x11/x_server_pixel_buffer.cc b/modules/desktop_capture/x11/x_server_pixel_buffer.cc
index 6983a6d..be00fa7 100644
--- a/modules/desktop_capture/x11/x_server_pixel_buffer.cc
+++ b/modules/desktop_capture/x11/x_server_pixel_buffer.cc
@@ -213,6 +213,18 @@
return true;
}
+bool XServerPixelBuffer::IsWindowValid() const {
+ XWindowAttributes attributes;
+ {
+ XErrorTrap error_trap(display_);
+ if (!XGetWindowAttributes(display_, window_, &attributes) ||
+ error_trap.GetLastErrorAndDisable() != 0) {
+ return false;
+ }
+ }
+ return true;
+}
+
void XServerPixelBuffer::Synchronize() {
if (shm_segment_info_ && !shm_pixmap_) {
// XShmGetImage can fail if the display is being reconfigured.
diff --git a/modules/desktop_capture/x11/x_server_pixel_buffer.h b/modules/desktop_capture/x11/x_server_pixel_buffer.h
index b81096c..98f263f 100644
--- a/modules/desktop_capture/x11/x_server_pixel_buffer.h
+++ b/modules/desktop_capture/x11/x_server_pixel_buffer.h
@@ -40,6 +40,9 @@
// Returns the size of the window the buffer was initialized for.
const DesktopSize& window_size() { return window_size_; }
+ // Returns true if the window can be found.
+ bool IsWindowValid() const;
+
// If shared memory is being used without pixmaps, synchronize this pixel
// buffer with the root window contents (otherwise, this is a no-op).
// This is to avoid doing a full-screen capture for each individual
diff --git a/modules/iLBC.target.darwin-arm64.mk b/modules/iLBC.target.darwin-arm64.mk
new file mode 100644
index 0000000..48a373e
--- /dev/null
+++ b/modules/iLBC.target.darwin-arm64.mk
@@ -0,0 +1,354 @@
+# This file is generated by gyp; do not edit.
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := third_party_webrtc_modules_iLBC_gyp
+LOCAL_MODULE_SUFFIX := .a
+LOCAL_MODULE_TAGS := optional
+LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
+gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
+gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
+
+# Make sure our deps are built first.
+GYP_TARGET_DEPENDENCIES :=
+
+GYP_GENERATED_OUTPUTS :=
+
+# Make sure our deps and generated files are built first.
+LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+
+LOCAL_GENERATED_SOURCES :=
+
+GYP_COPIED_SOURCE_ORIGIN_DIRS :=
+
+LOCAL_SRC_FILES := \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/constants.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/decode.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/interpolate.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/refiner.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/smooth.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/split_vq.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/state_search.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/vq3.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/vq4.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
+
+
+# Flags passed to both C and C++ files.
+MY_CFLAGS_Debug := \
+ --param=ssp-buffer-size=4 \
+ -Werror \
+ -fno-exceptions \
+ -fno-strict-aliasing \
+ -Wall \
+ -Wno-unused-parameter \
+ -Wno-missing-field-initializers \
+ -fvisibility=hidden \
+ -pipe \
+ -fPIC \
+ -Wno-unused-local-typedefs \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -ffunction-sections \
+ -funwind-tables \
+ -g \
+ -fno-short-enums \
+ -finline-limit=64 \
+ -Wa,--noexecstack \
+ -U_FORTIFY_SOURCE \
+ -Wno-extra \
+ -Wno-ignored-qualifiers \
+ -Wno-type-limits \
+ -Wno-unused-but-set-variable \
+ -Os \
+ -g \
+ -fomit-frame-pointer \
+ -fdata-sections \
+ -ffunction-sections \
+ -funwind-tables
+
+MY_DEFS_Debug := \
+ '-DV8_DEPRECATION_WARNINGS' \
+ '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
+ '-D_FILE_OFFSET_BITS=64' \
+ '-DNO_TCMALLOC' \
+ '-DDISABLE_NACL' \
+ '-DCHROMIUM_BUILD' \
+ '-DUSE_LIBJPEG_TURBO=1' \
+ '-DENABLE_WEBRTC=1' \
+ '-DUSE_PROPRIETARY_CODECS' \
+ '-DENABLE_CONFIGURATION_POLICY' \
+ '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
+ '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
+ '-DENABLE_EGLIMAGE=1' \
+ '-DCLD_VERSION=1' \
+ '-DENABLE_PRINTING=1' \
+ '-DENABLE_MANAGED_USERS=1' \
+ '-DWEBRTC_RESTRICT_LOGGING' \
+ '-DWEBRTC_MODULE_UTILITY_VIDEO' \
+ '-DWEBRTC_CHROMIUM_BUILD' \
+ '-DWEBRTC_POSIX' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_ANDROID' \
+ '-DWEBRTC_ANDROID_OPENSLES' \
+ '-DUSE_OPENSSL=1' \
+ '-DUSE_OPENSSL_CERTS=1' \
+ '-D__STDC_CONSTANT_MACROS' \
+ '-D__STDC_FORMAT_MACROS' \
+ '-DANDROID' \
+ '-D__GNU_SOURCE=1' \
+ '-DUSE_STLPORT=1' \
+ '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
+ '-DCHROME_BUILD_ID=""' \
+ '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
+ '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
+ '-D_DEBUG'
+
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES_Debug := \
+ $(LOCAL_PATH)/third_party \
+ $(LOCAL_PATH) \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
+ $(LOCAL_PATH)/third_party/webrtc \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(PWD)/frameworks/wilhelm/include \
+ $(PWD)/bionic \
+ $(PWD)/external/stlport/stlport
+
+
+# Flags passed to only C++ (and not C) files.
+LOCAL_CPPFLAGS_Debug := \
+ -fno-rtti \
+ -fno-threadsafe-statics \
+ -fvisibility-inlines-hidden \
+ -Wsign-compare \
+ -Wno-non-virtual-dtor \
+ -Wno-sign-promo
+
+
+# Flags passed to both C and C++ files.
+MY_CFLAGS_Release := \
+ --param=ssp-buffer-size=4 \
+ -Werror \
+ -fno-exceptions \
+ -fno-strict-aliasing \
+ -Wall \
+ -Wno-unused-parameter \
+ -Wno-missing-field-initializers \
+ -fvisibility=hidden \
+ -pipe \
+ -fPIC \
+ -Wno-unused-local-typedefs \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -ffunction-sections \
+ -funwind-tables \
+ -g \
+ -fno-short-enums \
+ -finline-limit=64 \
+ -Wa,--noexecstack \
+ -U_FORTIFY_SOURCE \
+ -Wno-extra \
+ -Wno-ignored-qualifiers \
+ -Wno-type-limits \
+ -Wno-unused-but-set-variable \
+ -Os \
+ -fno-ident \
+ -fdata-sections \
+ -ffunction-sections \
+ -fomit-frame-pointer \
+ -funwind-tables
+
+MY_DEFS_Release := \
+ '-DV8_DEPRECATION_WARNINGS' \
+ '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
+ '-D_FILE_OFFSET_BITS=64' \
+ '-DNO_TCMALLOC' \
+ '-DDISABLE_NACL' \
+ '-DCHROMIUM_BUILD' \
+ '-DUSE_LIBJPEG_TURBO=1' \
+ '-DENABLE_WEBRTC=1' \
+ '-DUSE_PROPRIETARY_CODECS' \
+ '-DENABLE_CONFIGURATION_POLICY' \
+ '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
+ '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
+ '-DENABLE_EGLIMAGE=1' \
+ '-DCLD_VERSION=1' \
+ '-DENABLE_PRINTING=1' \
+ '-DENABLE_MANAGED_USERS=1' \
+ '-DWEBRTC_RESTRICT_LOGGING' \
+ '-DWEBRTC_MODULE_UTILITY_VIDEO' \
+ '-DWEBRTC_CHROMIUM_BUILD' \
+ '-DWEBRTC_POSIX' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_ANDROID' \
+ '-DWEBRTC_ANDROID_OPENSLES' \
+ '-DUSE_OPENSSL=1' \
+ '-DUSE_OPENSSL_CERTS=1' \
+ '-D__STDC_CONSTANT_MACROS' \
+ '-D__STDC_FORMAT_MACROS' \
+ '-DANDROID' \
+ '-D__GNU_SOURCE=1' \
+ '-DUSE_STLPORT=1' \
+ '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
+ '-DCHROME_BUILD_ID=""' \
+ '-DNDEBUG' \
+ '-DNVALGRIND' \
+ '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
+ '-D_FORTIFY_SOURCE=2'
+
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES_Release := \
+ $(LOCAL_PATH)/third_party \
+ $(LOCAL_PATH) \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
+ $(LOCAL_PATH)/third_party/webrtc \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(PWD)/frameworks/wilhelm/include \
+ $(PWD)/bionic \
+ $(PWD)/external/stlport/stlport
+
+
+# Flags passed to only C++ (and not C) files.
+LOCAL_CPPFLAGS_Release := \
+ -fno-rtti \
+ -fno-threadsafe-statics \
+ -fvisibility-inlines-hidden \
+ -Wsign-compare \
+ -Wno-non-virtual-dtor \
+ -Wno-sign-promo
+
+
+LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
+LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
+LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
+LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
+### Rules for final target.
+
+LOCAL_LDFLAGS_Debug := \
+ -Wl,-z,now \
+ -Wl,-z,relro \
+ -Wl,--fatal-warnings \
+ -Wl,-z,noexecstack \
+ -fPIC \
+ -nostdlib \
+ -Wl,--no-undefined \
+ -Wl,--exclude-libs=ALL \
+ -Wl,--warn-shared-textrel \
+ -Wl,-O1 \
+ -Wl,--as-needed
+
+
+LOCAL_LDFLAGS_Release := \
+ -Wl,-z,now \
+ -Wl,-z,relro \
+ -Wl,--fatal-warnings \
+ -Wl,-z,noexecstack \
+ -fPIC \
+ -nostdlib \
+ -Wl,--no-undefined \
+ -Wl,--exclude-libs=ALL \
+ -Wl,-O1 \
+ -Wl,--as-needed \
+ -Wl,--gc-sections \
+ -Wl,--warn-shared-textrel
+
+
+LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
+
+LOCAL_STATIC_LIBRARIES :=
+
+# Enable grouping to fix circular references
+LOCAL_GROUP_STATIC_LIBRARIES := true
+
+LOCAL_SHARED_LIBRARIES := \
+ libstlport \
+ libdl
+
+# Add target alias to "gyp_all_modules" target.
+.PHONY: gyp_all_modules
+gyp_all_modules: third_party_webrtc_modules_iLBC_gyp
+
+# Alias gyp target name.
+.PHONY: iLBC
+iLBC: third_party_webrtc_modules_iLBC_gyp
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/iLBC.target.linux-arm64.mk b/modules/iLBC.target.linux-arm64.mk
new file mode 100644
index 0000000..48a373e
--- /dev/null
+++ b/modules/iLBC.target.linux-arm64.mk
@@ -0,0 +1,354 @@
+# This file is generated by gyp; do not edit.
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE_CLASS := STATIC_LIBRARIES
+LOCAL_MODULE := third_party_webrtc_modules_iLBC_gyp
+LOCAL_MODULE_SUFFIX := .a
+LOCAL_MODULE_TAGS := optional
+LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
+gyp_intermediate_dir := $(call local-intermediates-dir,,$(GYP_VAR_PREFIX))
+gyp_shared_intermediate_dir := $(call intermediates-dir-for,GYP,shared,,,$(GYP_VAR_PREFIX))
+
+# Make sure our deps are built first.
+GYP_TARGET_DEPENDENCIES :=
+
+GYP_GENERATED_OUTPUTS :=
+
+# Make sure our deps and generated files are built first.
+LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+
+LOCAL_GENERATED_SOURCES :=
+
+GYP_COPIED_SOURCE_ORIGIN_DIRS :=
+
+LOCAL_SRC_FILES := \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/constants.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/decode.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/interpolate.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/refiner.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/smooth.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/split_vq.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/state_search.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/vq3.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/vq4.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c \
+ third_party/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
+
+
+# Flags passed to both C and C++ files.
+MY_CFLAGS_Debug := \
+ --param=ssp-buffer-size=4 \
+ -Werror \
+ -fno-exceptions \
+ -fno-strict-aliasing \
+ -Wall \
+ -Wno-unused-parameter \
+ -Wno-missing-field-initializers \
+ -fvisibility=hidden \
+ -pipe \
+ -fPIC \
+ -Wno-unused-local-typedefs \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -ffunction-sections \
+ -funwind-tables \
+ -g \
+ -fno-short-enums \
+ -finline-limit=64 \
+ -Wa,--noexecstack \
+ -U_FORTIFY_SOURCE \
+ -Wno-extra \
+ -Wno-ignored-qualifiers \
+ -Wno-type-limits \
+ -Wno-unused-but-set-variable \
+ -Os \
+ -g \
+ -fomit-frame-pointer \
+ -fdata-sections \
+ -ffunction-sections \
+ -funwind-tables
+
+MY_DEFS_Debug := \
+ '-DV8_DEPRECATION_WARNINGS' \
+ '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
+ '-D_FILE_OFFSET_BITS=64' \
+ '-DNO_TCMALLOC' \
+ '-DDISABLE_NACL' \
+ '-DCHROMIUM_BUILD' \
+ '-DUSE_LIBJPEG_TURBO=1' \
+ '-DENABLE_WEBRTC=1' \
+ '-DUSE_PROPRIETARY_CODECS' \
+ '-DENABLE_CONFIGURATION_POLICY' \
+ '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
+ '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
+ '-DENABLE_EGLIMAGE=1' \
+ '-DCLD_VERSION=1' \
+ '-DENABLE_PRINTING=1' \
+ '-DENABLE_MANAGED_USERS=1' \
+ '-DWEBRTC_RESTRICT_LOGGING' \
+ '-DWEBRTC_MODULE_UTILITY_VIDEO' \
+ '-DWEBRTC_CHROMIUM_BUILD' \
+ '-DWEBRTC_POSIX' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_ANDROID' \
+ '-DWEBRTC_ANDROID_OPENSLES' \
+ '-DUSE_OPENSSL=1' \
+ '-DUSE_OPENSSL_CERTS=1' \
+ '-D__STDC_CONSTANT_MACROS' \
+ '-D__STDC_FORMAT_MACROS' \
+ '-DANDROID' \
+ '-D__GNU_SOURCE=1' \
+ '-DUSE_STLPORT=1' \
+ '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
+ '-DCHROME_BUILD_ID=""' \
+ '-DDYNAMIC_ANNOTATIONS_ENABLED=1' \
+ '-DWTF_USE_DYNAMIC_ANNOTATIONS=1' \
+ '-D_DEBUG'
+
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES_Debug := \
+ $(LOCAL_PATH)/third_party \
+ $(LOCAL_PATH) \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
+ $(LOCAL_PATH)/third_party/webrtc \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(PWD)/frameworks/wilhelm/include \
+ $(PWD)/bionic \
+ $(PWD)/external/stlport/stlport
+
+
+# Flags passed to only C++ (and not C) files.
+LOCAL_CPPFLAGS_Debug := \
+ -fno-rtti \
+ -fno-threadsafe-statics \
+ -fvisibility-inlines-hidden \
+ -Wsign-compare \
+ -Wno-non-virtual-dtor \
+ -Wno-sign-promo
+
+
+# Flags passed to both C and C++ files.
+MY_CFLAGS_Release := \
+ --param=ssp-buffer-size=4 \
+ -Werror \
+ -fno-exceptions \
+ -fno-strict-aliasing \
+ -Wall \
+ -Wno-unused-parameter \
+ -Wno-missing-field-initializers \
+ -fvisibility=hidden \
+ -pipe \
+ -fPIC \
+ -Wno-unused-local-typedefs \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -fno-builtin-cos \
+ -fno-builtin-sin \
+ -fno-builtin-cosf \
+ -fno-builtin-sinf \
+ -ffunction-sections \
+ -funwind-tables \
+ -g \
+ -fno-short-enums \
+ -finline-limit=64 \
+ -Wa,--noexecstack \
+ -U_FORTIFY_SOURCE \
+ -Wno-extra \
+ -Wno-ignored-qualifiers \
+ -Wno-type-limits \
+ -Wno-unused-but-set-variable \
+ -Os \
+ -fno-ident \
+ -fdata-sections \
+ -ffunction-sections \
+ -fomit-frame-pointer \
+ -funwind-tables
+
+MY_DEFS_Release := \
+ '-DV8_DEPRECATION_WARNINGS' \
+ '-DBLINK_SCALE_FILTERS_AT_RECORD_TIME' \
+ '-D_FILE_OFFSET_BITS=64' \
+ '-DNO_TCMALLOC' \
+ '-DDISABLE_NACL' \
+ '-DCHROMIUM_BUILD' \
+ '-DUSE_LIBJPEG_TURBO=1' \
+ '-DENABLE_WEBRTC=1' \
+ '-DUSE_PROPRIETARY_CODECS' \
+ '-DENABLE_CONFIGURATION_POLICY' \
+ '-DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY' \
+ '-DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE' \
+ '-DENABLE_EGLIMAGE=1' \
+ '-DCLD_VERSION=1' \
+ '-DENABLE_PRINTING=1' \
+ '-DENABLE_MANAGED_USERS=1' \
+ '-DWEBRTC_RESTRICT_LOGGING' \
+ '-DWEBRTC_MODULE_UTILITY_VIDEO' \
+ '-DWEBRTC_CHROMIUM_BUILD' \
+ '-DWEBRTC_POSIX' \
+ '-DWEBRTC_LINUX' \
+ '-DWEBRTC_ANDROID' \
+ '-DWEBRTC_ANDROID_OPENSLES' \
+ '-DUSE_OPENSSL=1' \
+ '-DUSE_OPENSSL_CERTS=1' \
+ '-D__STDC_CONSTANT_MACROS' \
+ '-D__STDC_FORMAT_MACROS' \
+ '-DANDROID' \
+ '-D__GNU_SOURCE=1' \
+ '-DUSE_STLPORT=1' \
+ '-D_STLP_USE_PTR_SPECIALIZATIONS=1' \
+ '-DCHROME_BUILD_ID=""' \
+ '-DNDEBUG' \
+ '-DNVALGRIND' \
+ '-DDYNAMIC_ANNOTATIONS_ENABLED=0' \
+ '-D_FORTIFY_SOURCE=2'
+
+
+# Include paths placed before CFLAGS/CPPFLAGS
+LOCAL_C_INCLUDES_Release := \
+ $(LOCAL_PATH)/third_party \
+ $(LOCAL_PATH) \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/ilbc/interface \
+ $(LOCAL_PATH)/third_party/webrtc \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(PWD)/frameworks/wilhelm/include \
+ $(PWD)/bionic \
+ $(PWD)/external/stlport/stlport
+
+
+# Flags passed to only C++ (and not C) files.
+LOCAL_CPPFLAGS_Release := \
+ -fno-rtti \
+ -fno-threadsafe-statics \
+ -fvisibility-inlines-hidden \
+ -Wsign-compare \
+ -Wno-non-virtual-dtor \
+ -Wno-sign-promo
+
+
+LOCAL_CFLAGS := $(MY_CFLAGS_$(GYP_CONFIGURATION)) $(MY_DEFS_$(GYP_CONFIGURATION))
+LOCAL_C_INCLUDES := $(GYP_COPIED_SOURCE_ORIGIN_DIRS) $(LOCAL_C_INCLUDES_$(GYP_CONFIGURATION))
+LOCAL_CPPFLAGS := $(LOCAL_CPPFLAGS_$(GYP_CONFIGURATION))
+LOCAL_ASFLAGS := $(LOCAL_CFLAGS)
+### Rules for final target.
+
+LOCAL_LDFLAGS_Debug := \
+ -Wl,-z,now \
+ -Wl,-z,relro \
+ -Wl,--fatal-warnings \
+ -Wl,-z,noexecstack \
+ -fPIC \
+ -nostdlib \
+ -Wl,--no-undefined \
+ -Wl,--exclude-libs=ALL \
+ -Wl,--warn-shared-textrel \
+ -Wl,-O1 \
+ -Wl,--as-needed
+
+
+LOCAL_LDFLAGS_Release := \
+ -Wl,-z,now \
+ -Wl,-z,relro \
+ -Wl,--fatal-warnings \
+ -Wl,-z,noexecstack \
+ -fPIC \
+ -nostdlib \
+ -Wl,--no-undefined \
+ -Wl,--exclude-libs=ALL \
+ -Wl,-O1 \
+ -Wl,--as-needed \
+ -Wl,--gc-sections \
+ -Wl,--warn-shared-textrel
+
+
+LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
+
+LOCAL_STATIC_LIBRARIES :=
+
+# Enable grouping to fix circular references
+LOCAL_GROUP_STATIC_LIBRARIES := true
+
+LOCAL_SHARED_LIBRARIES := \
+ libstlport \
+ libdl
+
+# Add target alias to "gyp_all_modules" target.
+.PHONY: gyp_all_modules
+gyp_all_modules: third_party_webrtc_modules_iLBC_gyp
+
+# Alias gyp target name.
+.PHONY: iLBC
+iLBC: third_party_webrtc_modules_iLBC_gyp
+
+include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/iSAC.target.darwin-arm64.mk
similarity index 70%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/iSAC.target.darwin-arm64.mk
index c84869c..b0f1e93 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/iSAC.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_iSAC_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,44 +23,38 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/decode.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/fft.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/filterbank_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/filterbanks.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lattice.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/transform.c
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +74,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +116,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,9 +134,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -169,7 +156,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +175,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +217,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,9 +236,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -289,8 +268,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +282,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +304,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_iSAC_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: iSAC
+iSAC: third_party_webrtc_modules_iSAC_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/iSAC.target.linux-arm64.mk
similarity index 70%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/iSAC.target.linux-arm64.mk
index c84869c..b0f1e93 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/iSAC.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_iSAC_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,44 +23,38 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/decode.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/fft.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/filterbank_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/filterbanks.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lattice.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/main/source/transform.c
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +74,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +116,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,9 +134,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -169,7 +156,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +175,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +217,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,9 +236,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/main/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
@@ -289,8 +268,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +282,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +304,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_iSAC_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: iSAC
+iSAC: third_party_webrtc_modules_iSAC_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/iSACFix.target.darwin-arm64.mk
similarity index 70%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/iSACFix.target.darwin-arm64.mk
index c84869c..7275617 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/iSACFix.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_iSACFix_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,44 +23,36 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/fft.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filters.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/initialize.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/transform.c
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +72,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +114,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +132,12 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +155,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +174,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +216,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +235,12 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +268,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +282,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +304,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_iSACFix_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: iSACFix
+iSACFix: third_party_webrtc_modules_iSACFix_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/iSACFix.target.linux-arm64.mk
similarity index 70%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/iSACFix.target.linux-arm64.mk
index c84869c..7275617 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/iSACFix.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_iSACFix_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,44 +23,36 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/fft.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/filters.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/initialize.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c \
+ third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/transform.c
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +72,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +114,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +132,12 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +155,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +174,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +216,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +235,12 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/isac/fix/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +268,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +282,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +304,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_iSACFix_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: iSACFix
+iSACFix: third_party_webrtc_modules_iSACFix_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/media_file.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/media_file.target.darwin-arm64.mk
index c84869c..f663425 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/media_file.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_media_file_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,19 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/media_file/source/avi_file.cc \
+ third_party/webrtc/modules/media_file/source/media_file_impl.cc \
+ third_party/webrtc/modules/media_file/source/media_file_utility.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +50,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +92,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +110,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +130,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +149,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +191,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +210,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +240,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +254,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +276,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_media_file_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: media_file
+media_file: third_party_webrtc_modules_media_file_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/media_file.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/media_file.target.linux-arm64.mk
index c84869c..f663425 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/media_file.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_media_file_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,19 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/media_file/source/avi_file.cc \
+ third_party/webrtc/modules/media_file/source/media_file_impl.cc \
+ third_party/webrtc/modules/media_file/source/media_file_utility.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +50,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +92,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +110,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +130,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +149,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +191,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +210,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +240,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +254,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +276,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_media_file_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: media_file
+media_file: third_party_webrtc_modules_media_file_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/modules.gyp b/modules/modules.gyp
index 8fd0714..444c665 100644
--- a/modules/modules.gyp
+++ b/modules/modules.gyp
@@ -16,8 +16,7 @@
'audio_coding/codecs/isac/main/source/isac.gypi',
'audio_coding/codecs/isac/fix/source/isacfix.gypi',
'audio_coding/codecs/pcm16b/pcm16b.gypi',
- 'audio_coding/main/source/audio_coding_module.gypi',
- 'audio_coding/neteq/neteq.gypi',
+ 'audio_coding/main/acm2/audio_coding_module.gypi',
'audio_coding/neteq4/neteq.gypi',
'audio_conference_mixer/source/audio_conference_mixer.gypi',
'audio_device/audio_device.gypi',
@@ -76,7 +75,6 @@
'desktop_capture',
'iSACFix',
'media_file',
- 'NetEq',
'NetEq4',
'NetEq4TestTools',
'neteq_unittest_tools',
@@ -107,7 +105,6 @@
'audio_coding/main/acm2/call_statistics_unittest.cc',
'audio_coding/main/acm2/initial_delay_manager_unittest.cc',
'audio_coding/main/acm2/nack_unittest.cc',
- 'audio_coding/main/source/acm_neteq_unittest.cc',
'audio_coding/codecs/cng/cng_unittest.cc',
'audio_coding/codecs/isac/fix/source/filters_unittest.cc',
'audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc',
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/paced_sender.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/paced_sender.target.darwin-arm64.mk
index c84869c..7708ece 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/paced_sender.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_paced_sender_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,17 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/pacing/paced_sender.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +48,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +108,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +126,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +145,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +187,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +206,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +234,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +248,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +270,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_paced_sender_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: paced_sender
+paced_sender: third_party_webrtc_modules_paced_sender_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/paced_sender.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/paced_sender.target.linux-arm64.mk
index c84869c..7708ece 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/paced_sender.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_paced_sender_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,17 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/pacing/paced_sender.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +48,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +108,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +126,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +145,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +187,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +206,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +234,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +248,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +270,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_paced_sender_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: paced_sender
+paced_sender: third_party_webrtc_modules_paced_sender_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/remote_bitrate_estimator.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/remote_bitrate_estimator.target.darwin-arm64.mk
index c84869c..67fb35d 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/remote_bitrate_estimator.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_remote_bitrate_estimator_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,18 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/remote_bitrate_estimator/rate_statistics.cc \
+ third_party/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +49,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +91,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +109,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +127,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +146,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +188,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +207,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +235,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +249,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +271,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_remote_bitrate_estimator_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: remote_bitrate_estimator
+remote_bitrate_estimator: third_party_webrtc_modules_remote_bitrate_estimator_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/remote_bitrate_estimator.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/remote_bitrate_estimator.target.linux-arm64.mk
index c84869c..67fb35d 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/remote_bitrate_estimator.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_remote_bitrate_estimator_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,18 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/remote_bitrate_estimator/rate_statistics.cc \
+ third_party/webrtc/modules/remote_bitrate_estimator/rtp_to_ntp.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +49,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +91,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +109,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +127,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +146,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +188,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +207,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +235,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +249,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +271,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_remote_bitrate_estimator_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: remote_bitrate_estimator
+remote_bitrate_estimator: third_party_webrtc_modules_remote_bitrate_estimator_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/remote_bitrate_estimator/bwe_simulations.cc b/modules/remote_bitrate_estimator/bwe_simulations.cc
index e4c67b5..6b208e4 100644
--- a/modules/remote_bitrate_estimator/bwe_simulations.cc
+++ b/modules/remote_bitrate_estimator/bwe_simulations.cc
@@ -8,65 +8,63 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include "gtest/gtest.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h"
#include "webrtc/test/testsupport/fileutils.h"
+using std::string;
+
namespace webrtc {
namespace testing {
namespace bwe {
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
-std::vector<BweTestConfig::EstimatorConfig> SingleEstimatorConfig() {
+BweTestConfig::EstimatorConfig CreateEstimatorConfig(
+ int flow_id, bool plot_delay, bool plot_estimate) {
static const AbsoluteSendTimeRemoteBitrateEstimatorFactory factory =
AbsoluteSendTimeRemoteBitrateEstimatorFactory();
- std::vector<BweTestConfig::EstimatorConfig> result;
- result.push_back(BweTestConfig::EstimatorConfig("AST", &factory, kAimdControl,
- false));
+ return BweTestConfig::EstimatorConfig("AST", flow_id, &factory, kAimdControl,
+ plot_delay, plot_estimate);
+}
+
+BweTestConfig MakeAdaptiveBweTestConfig() {
+ BweTestConfig result;
+ result.estimator_configs.push_back(CreateEstimatorConfig(0, true, true));
return result;
}
-std::vector<const PacketSenderFactory*> AdaptiveVideoSenderFactories(
- uint32_t count) {
- static const AdaptiveVideoPacketSenderFactory factories[] = {
- AdaptiveVideoPacketSenderFactory(30.00f, 150, 0x1234, 0.13f),
- AdaptiveVideoPacketSenderFactory(30.00f, 300, 0x3456, 0.26f),
- AdaptiveVideoPacketSenderFactory(15.00f, 600, 0x4567, 0.39f),
- };
-
- assert(count <= sizeof(factories) / sizeof(factories[0]));
-
- std::vector<const PacketSenderFactory*> result;
- for (uint32_t i = 0; i < count; ++i) {
- result.push_back(&factories[i]);
+BweTestConfig MakeMultiFlowBweTestConfig(int flow_count) {
+ BweTestConfig result;
+ for (int i = 0; i < flow_count; ++i) {
+ result.estimator_configs.push_back(CreateEstimatorConfig(i, false, true));
}
return result;
}
-BweTestConfig MakeAdaptiveBweTestConfig(uint32_t sender_count) {
- BweTestConfig result = {
- AdaptiveVideoSenderFactories(sender_count), SingleEstimatorConfig()
- };
- return result;
-}
-
// This test fixture is used to instantiate tests running with adaptive video
// senders.
-class BweSimulation : public BweTest {
+class BweSimulation : public BweTest,
+ public ::testing::TestWithParam<BweTestConfig> {
public:
BweSimulation() : BweTest() {}
virtual ~BweSimulation() {}
+ virtual void SetUp() {
+ const BweTestConfig& config = GetParam();
+ SetupTestFromConfig(config);
+ }
+
private:
DISALLOW_COPY_AND_ASSIGN(BweSimulation);
};
INSTANTIATE_TEST_CASE_P(VideoSendersTest, BweSimulation,
- ::testing::Values(MakeAdaptiveBweTestConfig(1),
- MakeAdaptiveBweTestConfig(3)));
+ ::testing::Values(MakeAdaptiveBweTestConfig()));
TEST_P(BweSimulation, SprintUplinkTest) {
VerboseLogging(true);
+ AdaptiveVideoSender sender(0, this, 30, 300, 0, 0);
RateCounterFilter counter1(this, "sender_output");
TraceBasedDeliveryFilter filter(this, "link_capacity");
RateCounterFilter counter2(this, "receiver_input");
@@ -76,6 +74,7 @@
TEST_P(BweSimulation, Verizon4gDownlinkTest) {
VerboseLogging(true);
+ AdaptiveVideoSender sender(0, this, 30, 300, 0, 0);
RateCounterFilter counter1(this, "sender_output");
TraceBasedDeliveryFilter filter(this, "link_capacity");
RateCounterFilter counter2(this, "receiver_input");
@@ -85,6 +84,7 @@
TEST_P(BweSimulation, Choke1000kbps500kbps1000kbps) {
VerboseLogging(true);
+ AdaptiveVideoSender sender(0, this, 30, 300, 0, 0);
ChokeFilter filter(this);
RateCounterFilter counter(this, "receiver_input");
filter.SetCapacity(1000);
@@ -98,6 +98,7 @@
TEST_P(BweSimulation, Choke200kbps30kbps200kbps) {
VerboseLogging(true);
+ AdaptiveVideoSender sender(0, this, 30, 300, 0, 0);
ChokeFilter filter(this);
RateCounterFilter counter(this, "receiver_input");
filter.SetCapacity(200);
@@ -111,6 +112,7 @@
TEST_P(BweSimulation, GoogleWifiTrace3Mbps) {
VerboseLogging(true);
+ AdaptiveVideoSender sender(0, this, 30, 300, 0, 0);
RateCounterFilter counter1(this, "sender_output");
TraceBasedDeliveryFilter filter(this, "link_capacity");
filter.SetMaxDelay(500);
@@ -118,6 +120,42 @@
ASSERT_TRUE(filter.Init(test::ResourcePath("google-wifi-3mbps", "rx")));
RunFor(300 * 1000);
}
+
+class MultiFlowBweSimulation : public BweSimulation {
+ public:
+ MultiFlowBweSimulation() : BweSimulation() {}
+ virtual ~MultiFlowBweSimulation() {}
+
+ private:
+ DISALLOW_COPY_AND_ASSIGN(MultiFlowBweSimulation);
+};
+
+INSTANTIATE_TEST_CASE_P(VideoSendersTest, MultiFlowBweSimulation,
+ ::testing::Values(MakeMultiFlowBweTestConfig(3)));
+
+TEST_P(MultiFlowBweSimulation, SelfFairnessTest) {
+ VerboseLogging(true);
+ const int kAllFlowIds[] = {0, 1, 2};
+ const size_t kNumFlows = sizeof(kAllFlowIds) / sizeof(kAllFlowIds[0]);
+ scoped_ptr<AdaptiveVideoSender> senders[kNumFlows];
+ for (size_t i = 0; i < kNumFlows; ++i) {
+ senders[i].reset(new AdaptiveVideoSender(kAllFlowIds[i], this, 30, 300, 0,
+ 0));
+ }
+ // Second and third flow.
+ ChokeFilter choke(this, CreateFlowIds(&kAllFlowIds[1], 2));
+ choke.SetCapacity(1500);
+ // First flow.
+ ChokeFilter choke2(this, CreateFlowIds(&kAllFlowIds[0], 1));
+ choke2.SetCapacity(1000);
+
+ scoped_ptr<RateCounterFilter> rate_counters[kNumFlows];
+ for (size_t i = 0; i < kNumFlows; ++i) {
+ rate_counters[i].reset(new RateCounterFilter(
+ this, CreateFlowIds(&kAllFlowIds[i], 1), "receiver_input"));
+ }
+ RunFor(30 * 60 * 1000);
+}
#endif // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
} // namespace bwe
} // namespace testing
diff --git a/modules/remote_bitrate_estimator/include/rtp_to_ntp.h b/modules/remote_bitrate_estimator/include/rtp_to_ntp.h
index 7928abf..08a4d46 100644
--- a/modules/remote_bitrate_estimator/include/rtp_to_ntp.h
+++ b/modules/remote_bitrate_estimator/include/rtp_to_ntp.h
@@ -29,6 +29,15 @@
typedef std::list<RtcpMeasurement> RtcpList;
+// Updates |rtcp_list| with timestamps from the latest RTCP SR.
+// |new_rtcp_sr| will be set to true if these are the timestamps which have
+// never be added to |rtcp_list|.
+bool UpdateRtcpList(uint32_t ntp_secs,
+ uint32_t ntp_frac,
+ uint32_t rtp_timestamp,
+ RtcpList* rtcp_list,
+ bool* new_rtcp_sr);
+
// Converts an RTP timestamp to the NTP domain in milliseconds using two
// (RTP timestamp, NTP timestamp) pairs.
bool RtpToNtpMs(int64_t rtp_timestamp, const RtcpList& rtcp,
diff --git a/modules/remote_bitrate_estimator/rate_statistics.cc b/modules/remote_bitrate_estimator/rate_statistics.cc
index 4a9b448..48485ff 100644
--- a/modules/remote_bitrate_estimator/rate_statistics.cc
+++ b/modules/remote_bitrate_estimator/rate_statistics.cc
@@ -10,6 +10,8 @@
#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
+#include <assert.h>
+
namespace webrtc {
RateStatistics::RateStatistics(uint32_t window_size_ms, float scale)
diff --git a/modules/remote_bitrate_estimator/rate_statistics.h b/modules/remote_bitrate_estimator/rate_statistics.h
index 4296690..f97371b 100644
--- a/modules/remote_bitrate_estimator/rate_statistics.h
+++ b/modules/remote_bitrate_estimator/rate_statistics.h
@@ -34,7 +34,7 @@
// Counters are kept in buckets (circular buffer), with one bucket
// per millisecond.
const int num_buckets_;
- scoped_array<uint32_t> buckets_;
+ scoped_ptr<uint32_t[]> buckets_;
// Total count recorded in buckets.
uint32_t accumulated_count_;
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/remote_bitrate_estimator/rbe_components.target.darwin-arm64.mk
similarity index 63%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/remote_bitrate_estimator/rbe_components.target.darwin-arm64.mk
index c84869c..de6b173 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/remote_bitrate_estimator/rbe_components.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_remote_bitrate_estimator_rbe_components_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,19 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc \
+ third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc \
+ third_party/webrtc/modules/remote_bitrate_estimator/remote_rate_control.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,17 +46,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +88,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +106,6 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +123,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -185,17 +138,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +180,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +199,6 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +226,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +240,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +262,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_remote_bitrate_estimator_rbe_components_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: rbe_components
+rbe_components: third_party_webrtc_modules_remote_bitrate_estimator_rbe_components_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/remote_bitrate_estimator/rbe_components.target.linux-arm64.mk
similarity index 63%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/remote_bitrate_estimator/rbe_components.target.linux-arm64.mk
index c84869c..de6b173 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/remote_bitrate_estimator/rbe_components.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_remote_bitrate_estimator_rbe_components_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,19 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc \
+ third_party/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc \
+ third_party/webrtc/modules/remote_bitrate_estimator/remote_rate_control.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,17 +46,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +88,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +106,6 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +123,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -185,17 +138,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +180,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +199,6 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +226,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +240,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +262,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_remote_bitrate_estimator_rbe_components_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: rbe_components
+rbe_components: third_party_webrtc_modules_remote_bitrate_estimator_rbe_components_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc b/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
index e4fb4fa..67b6084 100644
--- a/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
+++ b/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc
@@ -8,72 +8,95 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <sstream>
+
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test.h"
#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/test/testsupport/perf_test.h"
+
+using std::string;
namespace webrtc {
namespace testing {
namespace bwe {
-std::vector<const PacketSenderFactory*> VideoSenderFactories(uint32_t count) {
- static const VideoPacketSenderFactory factories[] = {
- VideoPacketSenderFactory(30.00f, 150, 0x1234, 0.13f),
- VideoPacketSenderFactory(15.00f, 500, 0x2345, 0.16f),
- VideoPacketSenderFactory(30.00f, 1200, 0x3456, 0.26f),
- VideoPacketSenderFactory(7.49f, 150, 0x4567, 0.05f),
- VideoPacketSenderFactory(7.50f, 150, 0x5678, 0.15f),
- VideoPacketSenderFactory(7.51f, 150, 0x6789, 0.25f),
- VideoPacketSenderFactory(15.02f, 150, 0x7890, 0.27f),
- VideoPacketSenderFactory(15.03f, 150, 0x8901, 0.38f),
- VideoPacketSenderFactory(30.02f, 150, 0x9012, 0.39f),
- VideoPacketSenderFactory(30.03f, 150, 0x0123, 0.52f)
- };
+enum Estimator { kAbsSendTime, kTransmissionOffset };
- assert(count <= sizeof(factories) / sizeof(factories[0]));
-
- std::vector<const PacketSenderFactory*> result;
- for (uint32_t i = 0; i < count; ++i) {
- result.push_back(&factories[i]);
- }
-
- return result;
-}
-
-std::vector<BweTestConfig::EstimatorConfig> EstimatorConfigs() {
+BweTestConfig::EstimatorConfig EstimatorConfigs(Estimator estimator,
+ int flow_id) {
static const RemoteBitrateEstimatorFactory factories[] = {
RemoteBitrateEstimatorFactory(),
AbsoluteSendTimeRemoteBitrateEstimatorFactory()
};
+ switch (estimator) {
+ case kTransmissionOffset:
+ return BweTestConfig::EstimatorConfig("TOF", flow_id, &factories[0],
+ kMimdControl, false, false);
+ case kAbsSendTime:
+ return BweTestConfig::EstimatorConfig("AST", flow_id, &factories[1],
+ kMimdControl, false, false);
+ }
+ assert(false);
+ return BweTestConfig::EstimatorConfig();
+}
- std::vector<BweTestConfig::EstimatorConfig> result;
- result.push_back(BweTestConfig::EstimatorConfig("TOF", &factories[0]));
- result.push_back(BweTestConfig::EstimatorConfig("AST", &factories[1]));
+struct DefaultBweTestConfig {
+ BweTestConfig bwe_test_config;
+ size_t number_of_senders;
+};
+
+DefaultBweTestConfig MakeBweTestConfig(uint32_t sender_count,
+ Estimator estimator) {
+ DefaultBweTestConfig result;
+ result.bwe_test_config.estimator_configs.push_back(
+ EstimatorConfigs(estimator, 0));
+ result.number_of_senders = sender_count;
return result;
}
-BweTestConfig MakeBweTestConfig(uint32_t sender_count) {
- BweTestConfig result = {
- VideoSenderFactories(sender_count), EstimatorConfigs()
- };
- return result;
-}
+class DefaultBweTest : public BweTest,
+ public ::testing::TestWithParam<DefaultBweTestConfig> {
+ public:
+ DefaultBweTest() : packet_senders_() {}
+ virtual ~DefaultBweTest() {}
-INSTANTIATE_TEST_CASE_P(VideoSendersTest, BweTest,
- ::testing::Values(MakeBweTestConfig(1),
- MakeBweTestConfig(3)));
+ virtual void SetUp() {
+ const DefaultBweTestConfig& config = GetParam();
+ SetupTestFromConfig(config.bwe_test_config);
+ for (size_t i = 0; i < config.number_of_senders; ++i) {
+ packet_senders_.push_back(new VideoSender(0, this, 30, 300, 0, 0));
+ }
+ }
-TEST_P(BweTest, UnlimitedSpeed) {
+ virtual void TearDown() {
+ while (!packet_senders_.empty()) {
+ delete packet_senders_.front();
+ packet_senders_.pop_front();
+ }
+ }
+
+ protected:
+ std::list<PacketSender*> packet_senders_;
+};
+
+INSTANTIATE_TEST_CASE_P(VideoSendersTest, DefaultBweTest,
+ ::testing::Values(MakeBweTestConfig(1, kAbsSendTime),
+ MakeBweTestConfig(3, kAbsSendTime),
+ MakeBweTestConfig(1, kTransmissionOffset),
+ MakeBweTestConfig(3, kTransmissionOffset)));
+
+TEST_P(DefaultBweTest, UnlimitedSpeed) {
VerboseLogging(false);
RunFor(10 * 60 * 1000);
}
-TEST_P(BweTest, DISABLED_SteadyLoss) {
+TEST_P(DefaultBweTest, DISABLED_SteadyLoss) {
LossFilter loss(this);
loss.SetLoss(20.0);
RunFor(10 * 60 * 1000);
}
-TEST_P(BweTest, IncreasingLoss1) {
+TEST_P(DefaultBweTest, IncreasingLoss1) {
LossFilter loss(this);
for (int i = 0; i < 76; ++i) {
loss.SetLoss(i);
@@ -81,13 +104,13 @@
}
}
-TEST_P(BweTest, SteadyDelay) {
+TEST_P(DefaultBweTest, SteadyDelay) {
DelayFilter delay(this);
delay.SetDelay(1000);
RunFor(10 * 60 * 1000);
}
-TEST_P(BweTest, DISABLED_IncreasingDelay1) {
+TEST_P(DefaultBweTest, DISABLED_IncreasingDelay1) {
DelayFilter delay(this);
RunFor(10 * 60 * 1000);
for (int i = 0; i < 30 * 2; ++i) {
@@ -97,7 +120,7 @@
RunFor(10 * 60 * 1000);
}
-TEST_P(BweTest, IncreasingDelay2) {
+TEST_P(DefaultBweTest, IncreasingDelay2) {
DelayFilter delay(this);
RateCounterFilter counter(this);
RunFor(1 * 60 * 1000);
@@ -109,7 +132,7 @@
RunFor(10 * 60 * 1000);
}
-TEST_P(BweTest, JumpyDelay1) {
+TEST_P(DefaultBweTest, JumpyDelay1) {
DelayFilter delay(this);
RunFor(10 * 60 * 1000);
for (int i = 1; i < 200; ++i) {
@@ -122,14 +145,14 @@
RunFor(10 * 60 * 1000);
}
-TEST_P(BweTest, SteadyJitter) {
+TEST_P(DefaultBweTest, SteadyJitter) {
JitterFilter jitter(this);
RateCounterFilter counter(this);
jitter.SetJitter(20);
RunFor(2 * 60 * 1000);
}
-TEST_P(BweTest, IncreasingJitter1) {
+TEST_P(DefaultBweTest, IncreasingJitter1) {
JitterFilter jitter(this);
for (int i = 0; i < 2 * 60 * 2; ++i) {
jitter.SetJitter(i);
@@ -138,7 +161,7 @@
RunFor(10 * 60 * 1000);
}
-TEST_P(BweTest, IncreasingJitter2) {
+TEST_P(DefaultBweTest, IncreasingJitter2) {
JitterFilter jitter(this);
RunFor(30 * 1000);
for (int i = 1; i < 51; ++i) {
@@ -149,13 +172,13 @@
RunFor(10 * 60 * 1000);
}
-TEST_P(BweTest, SteadyReorder) {
+TEST_P(DefaultBweTest, SteadyReorder) {
ReorderFilter reorder(this);
reorder.SetReorder(20.0);
RunFor(10 * 60 * 1000);
}
-TEST_P(BweTest, IncreasingReorder1) {
+TEST_P(DefaultBweTest, IncreasingReorder1) {
ReorderFilter reorder(this);
for (int i = 0; i < 76; ++i) {
reorder.SetReorder(i);
@@ -163,13 +186,13 @@
}
}
-TEST_P(BweTest, DISABLED_SteadyChoke) {
+TEST_P(DefaultBweTest, DISABLED_SteadyChoke) {
ChokeFilter choke(this);
choke.SetCapacity(140);
RunFor(10 * 60 * 1000);
}
-TEST_P(BweTest, DISABLED_IncreasingChoke1) {
+TEST_P(DefaultBweTest, DISABLED_IncreasingChoke1) {
ChokeFilter choke(this);
for (int i = 1200; i >= 100; i -= 100) {
choke.SetCapacity(i);
@@ -177,7 +200,7 @@
}
}
-TEST_P(BweTest, DISABLED_IncreasingChoke2) {
+TEST_P(DefaultBweTest, DISABLED_IncreasingChoke2) {
ChokeFilter choke(this);
RunFor(60 * 1000);
for (int i = 1200; i >= 100; i -= 20) {
@@ -186,7 +209,7 @@
}
}
-TEST_P(BweTest, DISABLED_Multi1) {
+TEST_P(DefaultBweTest, DISABLED_Multi1) {
DelayFilter delay(this);
ChokeFilter choke(this);
RateCounterFilter counter(this);
@@ -201,7 +224,7 @@
RunFor(5 * 60 * 1000);
}
-TEST_P(BweTest, Multi2) {
+TEST_P(DefaultBweTest, Multi2) {
ChokeFilter choke(this);
JitterFilter jitter(this);
RateCounterFilter counter(this);
@@ -209,6 +232,108 @@
jitter.SetJitter(120);
RunFor(5 * 60 * 1000);
}
+
+// This test fixture is used to instantiate tests running with adaptive video
+// senders.
+class BweFeedbackTest : public BweTest,
+ public ::testing::TestWithParam<BweTestConfig> {
+ public:
+ BweFeedbackTest() : BweTest() {}
+ virtual ~BweFeedbackTest() {}
+
+ virtual void SetUp() {
+ BweTestConfig config;
+ config.estimator_configs.push_back(EstimatorConfigs(kAbsSendTime, 0));
+ SetupTestFromConfig(config);
+ }
+
+ void PrintResults(double max_throughput_kbps, Stats<double> throughput_kbps,
+ Stats<double> delay_ms) {
+ double utilization = throughput_kbps.GetMean() / max_throughput_kbps;
+ webrtc::test::PrintResult("BwePerformance",
+ GetTestName(),
+ "Utilization",
+ utilization * 100.0,
+ "%",
+ false);
+ std::stringstream ss;
+ ss << throughput_kbps.GetStdDev() / throughput_kbps.GetMean();
+ webrtc::test::PrintResult("BwePerformance",
+ GetTestName(),
+ "Utilization var coeff",
+ ss.str(),
+ "",
+ false);
+ webrtc::test::PrintResult("BwePerformance",
+ GetTestName(),
+ "Average delay",
+ delay_ms.AsString(),
+ "ms",
+ false);
+ }
+
+ private:
+ DISALLOW_COPY_AND_ASSIGN(BweFeedbackTest);
+};
+
+TEST_F(BweFeedbackTest, Choke1000kbps500kbps1000kbps) {
+ AdaptiveVideoSender sender(0, this, 30, 300, 0, 0);
+ ChokeFilter filter(this);
+ RateCounterFilter counter(this, "receiver_input");
+ const int kHighCapacityKbps = 1000;
+ const int kLowCapacityKbps = 500;
+ filter.SetCapacity(kHighCapacityKbps);
+ filter.SetMaxDelay(500);
+ RunFor(60 * 1000);
+ filter.SetCapacity(kLowCapacityKbps);
+ RunFor(60 * 1000);
+ filter.SetCapacity(kHighCapacityKbps);
+ RunFor(60 * 1000);
+ PrintResults((2 * kHighCapacityKbps + kLowCapacityKbps) / 3.0,
+ counter.GetBitrateStats(), filter.GetDelayStats());
+}
+
+TEST_F(BweFeedbackTest, Choke200kbps30kbps200kbps) {
+ AdaptiveVideoSender sender(0, this, 30, 300, 0, 0);
+ ChokeFilter filter(this);
+ RateCounterFilter counter(this, "receiver_input");
+ const int kHighCapacityKbps = 200;
+ const int kLowCapacityKbps = 30;
+ filter.SetCapacity(kHighCapacityKbps);
+ filter.SetMaxDelay(500);
+ RunFor(60 * 1000);
+ filter.SetCapacity(kLowCapacityKbps);
+ RunFor(60 * 1000);
+ filter.SetCapacity(kHighCapacityKbps);
+ RunFor(60 * 1000);
+
+ PrintResults((2 * kHighCapacityKbps + kLowCapacityKbps) / 3.0,
+ counter.GetBitrateStats(), filter.GetDelayStats());
+}
+
+TEST_F(BweFeedbackTest, Verizon4gDownlinkTest) {
+ AdaptiveVideoSender sender(0, this, 30, 300, 0, 0);
+ RateCounterFilter counter1(this, "sender_output");
+ TraceBasedDeliveryFilter filter(this, "link_capacity");
+ RateCounterFilter counter2(this, "receiver_input");
+ ASSERT_TRUE(filter.Init(test::ResourcePath("verizon4g-downlink", "rx")));
+ RunFor(22 * 60 * 1000);
+ PrintResults(filter.GetBitrateStats().GetMean(), counter2.GetBitrateStats(),
+ filter.GetDelayStats());
+}
+
+// webrtc:3277
+TEST_F(BweFeedbackTest, DISABLED_GoogleWifiTrace3Mbps) {
+ AdaptiveVideoSender sender(0, this, 30, 300, 0, 0);
+ RateCounterFilter counter1(this, "sender_output");
+ TraceBasedDeliveryFilter filter(this, "link_capacity");
+ filter.SetMaxDelay(500);
+ RateCounterFilter counter2(this, "receiver_input");
+ ASSERT_TRUE(filter.Init(test::ResourcePath("google-wifi-3mbps", "rx")));
+ RunFor(300 * 1000);
+ PrintResults(filter.GetBitrateStats().GetMean(), counter2.GetBitrateStats(),
+ filter.GetDelayStats());
+}
} // namespace bwe
} // namespace testing
} // namespace webrtc
diff --git a/modules/remote_bitrate_estimator/rtp_to_ntp.cc b/modules/remote_bitrate_estimator/rtp_to_ntp.cc
index 109edae..775cd0d 100644
--- a/modules/remote_bitrate_estimator/rtp_to_ntp.cc
+++ b/modules/remote_bitrate_estimator/rtp_to_ntp.cc
@@ -57,6 +57,40 @@
return true;
}
+bool UpdateRtcpList(uint32_t ntp_secs,
+ uint32_t ntp_frac,
+ uint32_t rtp_timestamp,
+ RtcpList* rtcp_list,
+ bool* new_rtcp_sr) {
+ *new_rtcp_sr = false;
+ if (ntp_secs == 0 && ntp_frac == 0) {
+ return false;
+ }
+
+ RtcpMeasurement measurement;
+ measurement.ntp_secs = ntp_secs;
+ measurement.ntp_frac = ntp_frac;
+ measurement.rtp_timestamp = rtp_timestamp;
+
+ for (RtcpList::iterator it = rtcp_list->begin();
+ it != rtcp_list->end(); ++it) {
+ if (measurement.ntp_secs == (*it).ntp_secs &&
+ measurement.ntp_frac == (*it).ntp_frac) {
+ // This RTCP has already been added to the list.
+ return true;
+ }
+ }
+
+ // We need two RTCP SR reports to map between RTP and NTP. More than two will
+ // not improve the mapping.
+ if (rtcp_list->size() == 2) {
+ rtcp_list->pop_back();
+ }
+ rtcp_list->push_front(measurement);
+ *new_rtcp_sr = true;
+ return true;
+}
+
// Converts |rtp_timestamp| to the NTP time base using the NTP and RTP timestamp
// pairs in |rtcp|. The converted timestamp is returned in
// |rtp_timestamp_in_ms|. This function compensates for wrap arounds in RTP
diff --git a/modules/remote_bitrate_estimator/test/bwe_test.cc b/modules/remote_bitrate_estimator/test/bwe_test.cc
index c500291..90ce6a3 100644
--- a/modules/remote_bitrate_estimator/test/bwe_test.cc
+++ b/modules/remote_bitrate_estimator/test/bwe_test.cc
@@ -23,44 +23,40 @@
namespace testing {
namespace bwe {
-namespace stl_helpers {
-template<typename T> void DeleteElements(T* container) {
- if (!container) return;
- for (typename T::iterator it = container->begin(); it != container->end();
- ++it) {
- delete *it;
- }
- container->clear();
-}
-} // namespace stl_helpers
-
-class BweTest::TestedEstimator : public RemoteBitrateObserver {
+class TestedEstimator : public RemoteBitrateObserver {
public:
static const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
+ static const int kDelayPlotIntervalMs = 100;
TestedEstimator(const string& test_name,
const BweTestConfig::EstimatorConfig& config)
: debug_name_(config.debug_name),
+ delay_log_prefix_(),
+ estimate_log_prefix_(),
+ last_delay_plot_ms_(0),
+ plot_delay_(config.plot_delay),
+ plot_estimate_(config.plot_estimate),
clock_(0),
stats_(),
- relative_estimator_stats_(),
latest_estimate_bps_(-1),
estimator_(config.estimator_factory->Create(
this, &clock_, config.control_type,
kRemoteBitrateEstimatorMinBitrateBps)),
- relative_estimator_(NULL),
baseline_(BaseLineFileInterface::Create(test_name + "_" + debug_name_,
config.update_baseline)) {
assert(estimator_.get());
assert(baseline_.get());
+ // Setup the prefix strings used when logging.
+ std::stringstream ss;
+ ss << "Delay_" << config.flow_id << "#2";
+ delay_log_prefix_ = ss.str();
+ ss.str("");
+ ss << "Estimate_" << config.flow_id << "#1";
+ estimate_log_prefix_ = ss.str();
// Default RTT in RemoteRateControl is 200 ms ; 50 ms is more realistic.
estimator_->OnRttUpdate(50);
}
- void SetRelativeEstimator(TestedEstimator* relative_estimator) {
- relative_estimator_ = relative_estimator;
- }
-
void EatPacket(const Packet& packet) {
BWE_TEST_LOGGING_CONTEXT(debug_name_);
@@ -70,9 +66,15 @@
// time once packet reaches the estimator.
int64_t packet_time_ms = (packet.send_time_us() + 500) / 1000;
BWE_TEST_LOGGING_TIME(packet_time_ms);
- BWE_TEST_LOGGING_PLOT("Delay_#2", clock_.TimeInMilliseconds(),
- packet_time_ms -
- (packet.creation_time_us() + 500) / 1000);
+ if (plot_delay_) {
+ if (clock_.TimeInMilliseconds() - last_delay_plot_ms_ >
+ kDelayPlotIntervalMs) {
+ BWE_TEST_LOGGING_PLOT(delay_log_prefix_, clock_.TimeInMilliseconds(),
+ packet_time_ms -
+ (packet.creation_time_us() + 500) / 1000);
+ last_delay_plot_ms_ = clock_.TimeInMilliseconds();
+ }
+ }
int64_t step_ms = estimator_->TimeUntilNextProcess();
while ((clock_.TimeInMilliseconds() + step_ms) < packet_time_ms) {
@@ -97,14 +99,9 @@
double estimated_kbps = static_cast<double>(estimated_bps) / 1000.0;
stats_.Push(estimated_kbps);
- BWE_TEST_LOGGING_PLOT("Estimate_#1", clock_.TimeInMilliseconds(),
- estimated_kbps);
- uint32_t relative_estimate_bps = 0;
- if (relative_estimator_ &&
- relative_estimator_->LatestEstimate(&relative_estimate_bps)) {
- double relative_estimate_kbps =
- static_cast<double>(relative_estimate_bps) / 1000.0;
- relative_estimator_stats_.Push(estimated_kbps - relative_estimate_kbps);
+ if (plot_estimate_) {
+ BWE_TEST_LOGGING_PLOT(estimate_log_prefix_, clock_.TimeInMilliseconds(),
+ estimated_kbps);
}
return true;
}
@@ -115,10 +112,6 @@
BWE_TEST_LOGGING_CONTEXT(debug_name_);
BWE_TEST_LOGGING_CONTEXT("Mean");
stats_.Log("kbps");
- if (relative_estimator_) {
- BWE_TEST_LOGGING_CONTEXT("Diff");
- relative_estimator_stats_.Log("kbps");
- }
}
void VerifyOrWriteBaseline() {
@@ -144,145 +137,217 @@
}
string debug_name_;
+ string delay_log_prefix_;
+ string estimate_log_prefix_;
+ int64_t last_delay_plot_ms_;
+ bool plot_delay_;
+ bool plot_estimate_;
SimulatedClock clock_;
Stats<double> stats_;
- Stats<double> relative_estimator_stats_;
int64_t latest_estimate_bps_;
scoped_ptr<RemoteBitrateEstimator> estimator_;
- TestedEstimator* relative_estimator_;
scoped_ptr<BaseLineFileInterface> baseline_;
DISALLOW_IMPLICIT_CONSTRUCTORS(TestedEstimator);
};
+class PacketProcessorRunner {
+ public:
+ explicit PacketProcessorRunner(PacketProcessor* processor)
+ : processor_(processor) {}
+
+ bool HasProcessor(const PacketProcessor* processor) const {
+ return processor == processor_;
+ }
+
+ void RunFor(int64_t time_ms, int64_t time_now_ms, Packets* in_out) {
+ Packets to_process;
+ FindPacketsToProcess(processor_->flow_ids(), in_out, &to_process);
+ processor_->RunFor(time_ms, &to_process);
+ QueuePackets(&to_process, time_now_ms * 1000);
+ if (!to_process.empty()) {
+ processor_->Plot((to_process.back().send_time_us() + 500) / 1000);
+ }
+ in_out->merge(to_process);
+ }
+
+ private:
+ void FindPacketsToProcess(const FlowIds& flow_ids, Packets* in,
+ Packets* out) {
+ assert(out->empty());
+ for (Packets::iterator it = in->begin(); it != in->end();) {
+ // TODO(holmer): Further optimize this by looking for consecutive flow ids
+ // in the packet list and only doing the binary search + splice once for a
+ // sequence.
+ if (std::binary_search(flow_ids.begin(), flow_ids.end(), it->flow_id())) {
+ Packets::iterator next = it;
+ ++next;
+ out->splice(out->end(), *in, it);
+ it = next;
+ } else {
+ ++it;
+ }
+ }
+ }
+
+ void QueuePackets(Packets* batch, int64_t end_of_batch_time_us) {
+ queue_.merge(*batch);
+ if (queue_.empty()) {
+ return;
+ }
+ Packets to_transfer;
+ Packets::iterator it = queue_.begin();
+ for (; it != queue_.end(); ++it) {
+ if (it->send_time_us() > end_of_batch_time_us) {
+ break;
+ }
+ }
+ to_transfer.splice(to_transfer.begin(), queue_, queue_.begin(), it);
+ batch->merge(to_transfer);
+ }
+
+ PacketProcessor* processor_;
+ Packets queue_;
+};
+
BweTest::BweTest()
: run_time_ms_(0),
+ time_now_ms_(-1),
simulation_interval_ms_(-1),
- previous_packets_(),
- packet_senders_(),
estimators_(),
processors_() {
}
BweTest::~BweTest() {
- stl_helpers::DeleteElements(&estimators_);
- stl_helpers::DeleteElements(&packet_senders_);
+ BWE_TEST_LOGGING_GLOBAL_ENABLE(true);
+ for (EstimatorMap::iterator it = estimators_.begin(); it != estimators_.end();
+ ++it) {
+ it->second->VerifyOrWriteBaseline();
+ it->second->LogStats();
+ }
+ BWE_TEST_LOGGING_GLOBAL_CONTEXT("");
+
+ for (EstimatorMap::iterator it = estimators_.begin();
+ it != estimators_.end(); ++it) {
+ delete it->second;
+ }
}
-void BweTest::SetUp() {
+void BweTest::SetupTestFromConfig(const BweTestConfig& config) {
const ::testing::TestInfo* const test_info =
::testing::UnitTest::GetInstance()->current_test_info();
string test_name =
string(test_info->test_case_name()) + "_" + string(test_info->name());
BWE_TEST_LOGGING_GLOBAL_CONTEXT(test_name);
-
- const BweTestConfig& config = GetParam();
-
- uint32_t total_capacity = 0;
- for (vector<const PacketSenderFactory*>::const_iterator it =
- config.sender_factories.begin(); it != config.sender_factories.end();
- ++it) {
- PacketSender* sender = (*it)->Create();
- assert(sender);
- total_capacity += sender->GetCapacityKbps();
- packet_senders_.push_back(sender);
- processors_.push_back(sender);
+ for (vector<BweTestConfig::EstimatorConfig>::const_iterator it =
+ config.estimator_configs.begin(); it != config.estimator_configs.end();
+ ++it) {
+ estimators_.insert(std::make_pair(it->flow_id, new TestedEstimator(
+ test_name, *it)));
}
- BWE_TEST_LOGGING_LOG1("RequiredLinkCapacity", "%d kbps", total_capacity)
-
- // Set simulation interval from first packet sender.
- if (packet_senders_.size() > 0) {
- simulation_interval_ms_ = packet_senders_[0]->GetFeedbackIntervalMs();
- }
-
- for (vector<BweTestConfig::EstimatorConfig>:: const_iterator it =
- config.estimator_configs.begin(); it != config.estimator_configs.end();
- ++it) {
- estimators_.push_back(new TestedEstimator(test_name, *it));
- }
- if (estimators_.size() > 1) {
- // Set all estimators as relative to the first one.
- for (uint32_t i = 1; i < estimators_.size(); ++i) {
- estimators_[i]->SetRelativeEstimator(estimators_[0]);
- }
- }
-
BWE_TEST_LOGGING_GLOBAL_ENABLE(false);
}
-void BweTest::TearDown() {
- BWE_TEST_LOGGING_GLOBAL_ENABLE(true);
-
- for (vector<TestedEstimator*>::iterator eit = estimators_.begin();
- eit != estimators_.end(); ++eit) {
- (*eit)->VerifyOrWriteBaseline();
- (*eit)->LogStats();
- }
-
- BWE_TEST_LOGGING_GLOBAL_CONTEXT("");
-}
-
-void BweTest::AddPacketProcessor(
- PacketProcessor* processor) {
+void BweTest::AddPacketProcessor(PacketProcessor* processor, bool is_sender) {
assert(processor);
- processors_.push_back(processor);
+ processors_.push_back(PacketProcessorRunner(processor));
+ if (is_sender) {
+ senders_.push_back(static_cast<PacketSender*>(processor));
+ }
+ const FlowIds& flow_ids = processor->flow_ids();
+ for (size_t i = 0; i < flow_ids.size(); ++i) {
+ assert(estimators_.count(flow_ids[i]) == 1);
+ }
}
void BweTest::RemovePacketProcessor(
PacketProcessor* processor) {
- vector<PacketProcessor*>::iterator it =
- std::find(processors_.begin(), processors_.end(), processor);
- assert(it != processors_.end());
- processors_.erase(it);
+ for (vector<PacketProcessorRunner>::iterator it = processors_.begin();
+ it != processors_.end(); ++it) {
+ if (it->HasProcessor(processor)) {
+ processors_.erase(it);
+ return;
+ }
+ }
+ assert(false);
}
void BweTest::VerboseLogging(bool enable) {
BWE_TEST_LOGGING_GLOBAL_ENABLE(enable);
}
+void BweTest::GiveFeedbackToAffectedSenders(int flow_id,
+ TestedEstimator* estimator) {
+ std::list<PacketSender*> affected_senders;
+ for (std::vector<PacketSender*>::iterator psit =
+ senders_.begin(); psit != senders_.end(); ++psit) {
+ const FlowIds& flow_ids = (*psit)->flow_ids();
+ if (std::binary_search(flow_ids.begin(), flow_ids.end(), flow_id)) {
+ affected_senders.push_back(*psit);
+ }
+ }
+ PacketSender::Feedback feedback = {0};
+ if (estimator->CheckEstimate(&feedback) && !affected_senders.empty()) {
+ // Allocate the bitrate evenly between the senders.
+ feedback.estimated_bps /= affected_senders.size();
+ for (std::list<PacketSender*>::iterator psit = affected_senders.begin();
+ psit != affected_senders.end(); ++psit) {
+ (*psit)->GiveFeedback(feedback);
+ }
+ }
+}
+
void BweTest::RunFor(int64_t time_ms) {
- for (run_time_ms_ += time_ms; run_time_ms_ >= simulation_interval_ms_;
- run_time_ms_ -= simulation_interval_ms_) {
+ // Set simulation interval from first packet sender.
+ // TODO(holmer): Support different feedback intervals for different flows.
+ if (!senders_.empty()) {
+ simulation_interval_ms_ = senders_[0]->GetFeedbackIntervalMs();
+ }
+ assert(simulation_interval_ms_ > 0);
+ if (time_now_ms_ == -1) {
+ time_now_ms_ = simulation_interval_ms_;
+ }
+ for (run_time_ms_ += time_ms;
+ time_now_ms_ <= run_time_ms_ - simulation_interval_ms_;
+ time_now_ms_ += simulation_interval_ms_) {
Packets packets;
- for (vector<PacketProcessor*>::const_iterator it =
+ for (vector<PacketProcessorRunner>::iterator it =
processors_.begin(); it != processors_.end(); ++it) {
- (*it)->RunFor(simulation_interval_ms_, &packets);
- if (!packets.empty()) {
- (*it)->Plot((packets.back().send_time_us() + 500) / 1000);
- }
+ it->RunFor(simulation_interval_ms_, time_now_ms_, &packets);
}
// Verify packets are in order between batches.
- if (!packets.empty() && !previous_packets_.empty()) {
- packets.splice(packets.begin(), previous_packets_,
- --previous_packets_.end());
- ASSERT_TRUE(IsTimeSorted(packets));
- packets.erase(packets.begin());
+ if (!packets.empty()) {
+ if (!previous_packets_.empty()) {
+ packets.splice(packets.begin(), previous_packets_,
+ --previous_packets_.end());
+ ASSERT_TRUE(IsTimeSorted(packets));
+ packets.erase(packets.begin());
+ }
+ ASSERT_LE(packets.front().send_time_us(), time_now_ms_ * 1000);
+ ASSERT_LE(packets.back().send_time_us(), time_now_ms_ * 1000);
} else {
ASSERT_TRUE(IsTimeSorted(packets));
}
- for (PacketsConstIt pit = packets.begin(); pit != packets.end(); ++pit) {
- for (vector<TestedEstimator*>::iterator eit = estimators_.begin();
- eit != estimators_.end(); ++eit) {
- (*eit)->EatPacket(*pit);
- }
+ for (PacketsConstIt it = packets.begin(); it != packets.end(); ++it) {
+ EstimatorMap::iterator est_it = estimators_.find(it->flow_id());
+ ASSERT_TRUE(est_it != estimators_.end());
+ est_it->second->EatPacket(*it);
}
- previous_packets_.swap(packets);
-
- for (vector<TestedEstimator*>::iterator eit = estimators_.begin();
- eit != estimators_.end(); ++eit) {
- PacketSender::Feedback feedback = {0};
- if ((*eit)->CheckEstimate(&feedback)) {
- for (vector<PacketSender*>::iterator psit = packet_senders_.begin();
- psit != packet_senders_.end(); ++psit) {
- (*psit)->GiveFeedback(feedback);
- }
- }
+ for (EstimatorMap::iterator est_it = estimators_.begin();
+ est_it != estimators_.end(); ++est_it) {
+ GiveFeedbackToAffectedSenders(est_it->first, est_it->second);
}
}
}
+
+string BweTest::GetTestName() const {
+ const ::testing::TestInfo* const test_info =
+ ::testing::UnitTest::GetInstance()->current_test_info();
+ return string(test_info->name());
+}
} // namespace bwe
} // namespace testing
} // namespace webrtc
diff --git a/modules/remote_bitrate_estimator/test/bwe_test.h b/modules/remote_bitrate_estimator/test/bwe_test.h
index cb708f4..ef35179 100644
--- a/modules/remote_bitrate_estimator/test/bwe_test.h
+++ b/modules/remote_bitrate_estimator/test/bwe_test.h
@@ -8,6 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <map>
#include <string>
#include <vector>
#include "gtest/gtest.h"
@@ -26,60 +27,96 @@
struct EstimatorConfig {
EstimatorConfig()
: debug_name(),
+ flow_id(0),
estimator_factory(NULL),
control_type(kMimdControl),
- update_baseline(false) {
+ update_baseline(false),
+ plot_delay(true),
+ plot_estimate(true) {
}
EstimatorConfig(std::string debug_name,
- const RemoteBitrateEstimatorFactory* estimator_factory)
+ int flow_id,
+ const RemoteBitrateEstimatorFactory* estimator_factory,
+ bool plot_delay,
+ bool plot_estimate)
: debug_name(debug_name),
+ flow_id(flow_id),
estimator_factory(estimator_factory),
control_type(kMimdControl),
- update_baseline(false) {
+ update_baseline(false),
+ plot_delay(plot_delay),
+ plot_estimate(plot_estimate) {
}
EstimatorConfig(std::string debug_name,
+ int flow_id,
+ const RemoteBitrateEstimatorFactory* estimator_factory,
+ RateControlType control_type,
+ bool plot_delay,
+ bool plot_estimate)
+ : debug_name(debug_name),
+ flow_id(flow_id),
+ estimator_factory(estimator_factory),
+ control_type(control_type),
+ update_baseline(false),
+ plot_delay(plot_delay),
+ plot_estimate(plot_estimate) {
+ }
+ EstimatorConfig(std::string debug_name,
+ int flow_id,
const RemoteBitrateEstimatorFactory* estimator_factory,
RateControlType control_type,
bool update_baseline)
: debug_name(debug_name),
+ flow_id(flow_id),
estimator_factory(estimator_factory),
control_type(control_type),
- update_baseline(update_baseline) {
+ update_baseline(update_baseline),
+ plot_delay(false),
+ plot_estimate(false) {
}
std::string debug_name;
+ int flow_id;
const RemoteBitrateEstimatorFactory* estimator_factory;
RateControlType control_type;
bool update_baseline;
+ bool plot_delay;
+ bool plot_estimate;
};
- std::vector<const PacketSenderFactory*> sender_factories;
std::vector<EstimatorConfig> estimator_configs;
};
-class BweTest : public ::testing::TestWithParam<BweTestConfig>,
- public PacketProcessorListener {
+class TestedEstimator;
+class PacketProcessorRunner;
+
+class BweTest : public PacketProcessorListener {
public:
BweTest();
virtual ~BweTest();
- virtual void SetUp();
- virtual void TearDown();
- virtual void AddPacketProcessor(PacketProcessor* processor);
+ virtual void AddPacketProcessor(PacketProcessor* processor, bool is_sender);
virtual void RemovePacketProcessor(PacketProcessor* processor);
protected:
+ void SetupTestFromConfig(const BweTestConfig& config);
void VerboseLogging(bool enable);
void RunFor(int64_t time_ms);
+ std::string GetTestName() const;
private:
- class TestedEstimator;
+ typedef std::map<int, TestedEstimator*> EstimatorMap;
+
+ void FindPacketsToProcess(const FlowIds& flow_ids, Packets* in,
+ Packets* out);
+ void GiveFeedbackToAffectedSenders(int flow_id, TestedEstimator* estimator);
int64_t run_time_ms_;
+ int64_t time_now_ms_;
int64_t simulation_interval_ms_;
+ EstimatorMap estimators_;
Packets previous_packets_;
- std::vector<PacketSender*> packet_senders_;
- std::vector<TestedEstimator*> estimators_;
- std::vector<PacketProcessor*> processors_;
+ std::vector<PacketSender*> senders_;
+ std::vector<PacketProcessorRunner> processors_;
DISALLOW_COPY_AND_ASSIGN(BweTest);
};
diff --git a/modules/remote_bitrate_estimator/test/bwe_test_framework.cc b/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
index 893b6fe..b3cd7db 100644
--- a/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
+++ b/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
@@ -45,6 +45,11 @@
DISALLOW_COPY_AND_ASSIGN(DelayCapHelper);
};
+const FlowIds CreateFlowIds(const int *flow_ids_array, size_t num_flow_ids) {
+ FlowIds flow_ids(&flow_ids_array[0], flow_ids_array + num_flow_ids);
+ return flow_ids;
+}
+
class RateCounter {
public:
RateCounter()
@@ -115,26 +120,29 @@
}
Packet::Packet()
- : creation_time_us_(-1),
+ : flow_id_(0),
+ creation_time_us_(-1),
send_time_us_(-1),
payload_size_(0) {
- memset(&header_, 0, sizeof(header_));
+ memset(&header_, 0, sizeof(header_));
}
-Packet::Packet(int64_t send_time_us, uint32_t payload_size,
+Packet::Packet(int flow_id, int64_t send_time_us, uint32_t payload_size,
const RTPHeader& header)
- : creation_time_us_(send_time_us),
- send_time_us_(send_time_us),
- payload_size_(payload_size),
- header_(header) {
+ : flow_id_(flow_id),
+ creation_time_us_(send_time_us),
+ send_time_us_(send_time_us),
+ payload_size_(payload_size),
+ header_(header) {
}
Packet::Packet(int64_t send_time_us, uint32_t sequence_number)
- : creation_time_us_(send_time_us),
+ : flow_id_(0),
+ creation_time_us_(send_time_us),
send_time_us_(send_time_us),
payload_size_(0) {
- memset(&header_, 0, sizeof(header_));
- header_.sequenceNumber = sequence_number;
+ memset(&header_, 0, sizeof(header_));
+ header_.sequenceNumber = sequence_number;
}
bool Packet::operator<(const Packet& rhs) const {
@@ -157,10 +165,20 @@
return true;
}
-PacketProcessor::PacketProcessor(PacketProcessorListener* listener)
- : listener_(listener) {
+PacketProcessor::PacketProcessor(PacketProcessorListener* listener,
+ bool is_sender)
+ : listener_(listener), flow_ids_(1, 0) {
if (listener_) {
- listener_->AddPacketProcessor(this);
+ listener_->AddPacketProcessor(this, is_sender);
+ }
+}
+
+PacketProcessor::PacketProcessor(PacketProcessorListener* listener,
+ const FlowIds& flow_ids,
+ bool is_sender)
+ : listener_(listener), flow_ids_(flow_ids) {
+ if (listener_) {
+ listener_->AddPacketProcessor(this, is_sender);
}
}
@@ -171,20 +189,36 @@
}
RateCounterFilter::RateCounterFilter(PacketProcessorListener* listener)
- : PacketProcessor(listener),
+ : PacketProcessor(listener, false),
rate_counter_(new RateCounter()),
- pps_stats_(),
+ packets_per_second_stats_(),
kbps_stats_(),
name_("") {}
RateCounterFilter::RateCounterFilter(PacketProcessorListener* listener,
const std::string& name)
- : PacketProcessor(listener),
+ : PacketProcessor(listener, false),
rate_counter_(new RateCounter()),
- pps_stats_(),
+ packets_per_second_stats_(),
kbps_stats_(),
name_(name) {}
+RateCounterFilter::RateCounterFilter(PacketProcessorListener* listener,
+ const FlowIds& flow_ids,
+ const std::string& name)
+ : PacketProcessor(listener, flow_ids, false),
+ rate_counter_(new RateCounter()),
+ packets_per_second_stats_(),
+ kbps_stats_(),
+ name_(name) {
+ std::stringstream ss;
+ ss << name_ << "_";
+ for (size_t i = 0; i < flow_ids.size(); ++i) {
+ ss << flow_ids[i] << ",";
+ }
+ name_ = ss.str();
+}
+
RateCounterFilter::~RateCounterFilter() {
LogStats();
}
@@ -199,7 +233,7 @@
void RateCounterFilter::LogStats() {
BWE_TEST_LOGGING_CONTEXT("RateCounterFilter");
- pps_stats_.Log("pps");
+ packets_per_second_stats_.Log("pps");
kbps_stats_.Log("kbps");
}
@@ -218,12 +252,12 @@
for (PacketsConstIt it = in_out->begin(); it != in_out->end(); ++it) {
rate_counter_->UpdateRates(it->send_time_us(), it->payload_size());
}
- pps_stats_.Push(rate_counter_->packets_per_second());
+ packets_per_second_stats_.Push(rate_counter_->packets_per_second());
kbps_stats_.Push(rate_counter_->bits_per_second() / 1000.0);
}
LossFilter::LossFilter(PacketProcessorListener* listener)
- : PacketProcessor(listener),
+ : PacketProcessor(listener, false),
random_(0x12345678),
loss_fraction_(0.0f) {
}
@@ -248,7 +282,7 @@
}
DelayFilter::DelayFilter(PacketProcessorListener* listener)
- : PacketProcessor(listener),
+ : PacketProcessor(listener, false),
delay_us_(0),
last_send_time_us_(0) {
}
@@ -270,7 +304,7 @@
}
JitterFilter::JitterFilter(PacketProcessorListener* listener)
- : PacketProcessor(listener),
+ : PacketProcessor(listener, false),
random_(0x89674523),
stddev_jitter_us_(0),
last_send_time_us_(0) {
@@ -295,7 +329,7 @@
}
ReorderFilter::ReorderFilter(PacketProcessorListener* listener)
- : PacketProcessor(listener),
+ : PacketProcessor(listener, false),
random_(0x27452389),
reorder_fraction_(0.0f) {
}
@@ -327,7 +361,15 @@
}
ChokeFilter::ChokeFilter(PacketProcessorListener* listener)
- : PacketProcessor(listener),
+ : PacketProcessor(listener, false),
+ kbps_(1200),
+ last_send_time_us_(0),
+ delay_cap_helper_(new DelayCapHelper()) {
+}
+
+ChokeFilter::ChokeFilter(PacketProcessorListener* listener,
+ const FlowIds& flow_ids)
+ : PacketProcessor(listener, flow_ids, false),
kbps_(1200),
last_send_time_us_(0),
delay_cap_helper_(new DelayCapHelper()) {
@@ -369,26 +411,30 @@
TraceBasedDeliveryFilter::TraceBasedDeliveryFilter(
PacketProcessorListener* listener)
- : PacketProcessor(listener),
+ : PacketProcessor(listener, false),
current_offset_us_(0),
delivery_times_us_(),
next_delivery_it_(),
local_time_us_(-1),
rate_counter_(new RateCounter),
name_(""),
- delay_cap_helper_(new DelayCapHelper()) {}
+ delay_cap_helper_(new DelayCapHelper()),
+ packets_per_second_stats_(),
+ kbps_stats_() {}
TraceBasedDeliveryFilter::TraceBasedDeliveryFilter(
PacketProcessorListener* listener,
const std::string& name)
- : PacketProcessor(listener),
+ : PacketProcessor(listener, false),
current_offset_us_(0),
delivery_times_us_(),
next_delivery_it_(),
local_time_us_(-1),
rate_counter_(new RateCounter),
name_(name),
- delay_cap_helper_(new DelayCapHelper()) {}
+ delay_cap_helper_(new DelayCapHelper()),
+ packets_per_second_stats_(),
+ kbps_stats_() {}
TraceBasedDeliveryFilter::~TraceBasedDeliveryFilter() {
}
@@ -449,6 +495,8 @@
}
++it;
}
+ packets_per_second_stats_.Push(rate_counter_->packets_per_second());
+ kbps_stats_.Push(rate_counter_->bits_per_second() / 1000.0);
}
void TraceBasedDeliveryFilter::SetMaxDelay(int max_delay_ms) {
@@ -459,6 +507,10 @@
return delay_cap_helper_->delay_stats();
}
+Stats<double> TraceBasedDeliveryFilter::GetBitrateStats() const {
+ return kbps_stats_;
+}
+
void TraceBasedDeliveryFilter::ProceedToNextSlot() {
if (*next_delivery_it_ <= local_time_us_) {
++next_delivery_it_;
@@ -478,12 +530,18 @@
}
PacketSender::PacketSender(PacketProcessorListener* listener)
- : PacketProcessor(listener) {
+ : PacketProcessor(listener, true) {}
+
+PacketSender::PacketSender(PacketProcessorListener* listener,
+ const FlowIds& flow_ids)
+ : PacketProcessor(listener, flow_ids, true) {
+
}
-VideoSender::VideoSender(PacketProcessorListener* listener, float fps,
- uint32_t kbps, uint32_t ssrc, float first_frame_offset)
- : PacketSender(listener),
+VideoSender::VideoSender(int flow_id, PacketProcessorListener* listener,
+ float fps, uint32_t kbps, uint32_t ssrc,
+ float first_frame_offset)
+ : PacketSender(listener, FlowIds(1, flow_id)),
kMaxPayloadSizeBytes(1200),
kTimestampBase(0xff80ff00ul),
frame_period_ms_(1000.0 / fps),
@@ -506,7 +564,7 @@
void VideoSender::RunFor(int64_t time_ms, Packets* in_out) {
assert(in_out);
now_ms_ += time_ms;
- Packets newPackets;
+ Packets new_packets;
while (now_ms_ >= next_frame_ms_) {
prototype_header_.sequenceNumber++;
prototype_header_.timestamp = kTimestampBase +
@@ -524,21 +582,23 @@
uint32_t payload_size = frame_size_bytes_;
while (payload_size > 0) {
uint32_t size = std::min(kMaxPayloadSizeBytes, payload_size);
- newPackets.push_back(Packet(send_time_us, size, prototype_header_));
+ new_packets.push_back(Packet(flow_ids()[0], send_time_us, size,
+ prototype_header_));
payload_size -= size;
}
next_frame_ms_ += frame_period_ms_;
}
- in_out->merge(newPackets);
+ in_out->merge(new_packets);
}
-AdaptiveVideoSender::AdaptiveVideoSender(PacketProcessorListener* listener,
+AdaptiveVideoSender::AdaptiveVideoSender(int flow_id,
+ PacketProcessorListener* listener,
float fps,
uint32_t kbps,
uint32_t ssrc,
float first_frame_offset)
- : VideoSender(listener, fps, kbps, ssrc, first_frame_offset) {}
+ : VideoSender(flow_id, listener, fps, kbps, ssrc, first_frame_offset) {}
void AdaptiveVideoSender::GiveFeedback(const PacketSender::Feedback& feedback) {
bytes_per_second_ = feedback.estimated_bps / 8;
diff --git a/modules/remote_bitrate_estimator/test/bwe_test_framework.h b/modules/remote_bitrate_estimator/test/bwe_test_framework.h
index 5302ba3..8af07f9 100644
--- a/modules/remote_bitrate_estimator/test/bwe_test_framework.h
+++ b/modules/remote_bitrate_estimator/test/bwe_test_framework.h
@@ -17,6 +17,7 @@
#include <algorithm>
#include <list>
#include <numeric>
+#include <sstream>
#include <string>
#include <vector>
@@ -31,6 +32,10 @@
class DelayCapHelper;
class RateCounter;
+
+typedef std::vector<int> FlowIds;
+const FlowIds CreateFlowIds(const int *flow_ids_array, size_t num_flow_ids);
+
template<typename T> class Stats {
public:
Stats()
@@ -84,6 +89,13 @@
return max_;
}
+ std::string AsString() {
+ std::stringstream ss;
+ ss << (GetMean() >= 0 ? GetMean() : -1) << ", " <<
+ (GetStdDev() >= 0 ? GetStdDev() : -1);
+ return ss.str();
+ }
+
void Log(const std::string& units) {
BWE_TEST_LOGGING_LOG5("", "%f %s\t+/-%f\t[%f,%f]",
GetMean(), units.c_str(), GetStdDev(), GetMin(), GetMax());
@@ -139,12 +151,13 @@
class Packet {
public:
Packet();
- Packet(int64_t send_time_us, uint32_t payload_size,
+ Packet(int flow_id, int64_t send_time_us, uint32_t payload_size,
const RTPHeader& header);
Packet(int64_t send_time_us, uint32_t sequence_number);
bool operator<(const Packet& rhs) const;
+ int flow_id() const { return flow_id_; }
int64_t creation_time_us() const { return creation_time_us_; }
void set_send_time_us(int64_t send_time_us);
int64_t send_time_us() const { return send_time_us_; }
@@ -152,6 +165,7 @@
const RTPHeader& header() const { return header_; }
private:
+ int flow_id_;
int64_t creation_time_us_; // Time when the packet was created.
int64_t send_time_us_; // Time the packet left last processor touching it.
uint32_t payload_size_; // Size of the (non-existent, simulated) payload.
@@ -170,13 +184,16 @@
public:
virtual ~PacketProcessorListener() {}
- virtual void AddPacketProcessor(PacketProcessor* processor) = 0;
+ virtual void AddPacketProcessor(PacketProcessor* processor,
+ bool is_sender) = 0;
virtual void RemovePacketProcessor(PacketProcessor* processor) = 0;
};
class PacketProcessor {
public:
- explicit PacketProcessor(PacketProcessorListener* listener);
+ PacketProcessor(PacketProcessorListener* listener, bool is_sender);
+ PacketProcessor(PacketProcessorListener* listener, const FlowIds& flow_ids,
+ bool is_sender);
virtual ~PacketProcessor();
// Called after each simulation batch to allow the processor to plot any
@@ -188,8 +205,11 @@
// |send_time_us_|. The simulation time |time_ms| is optional to use.
virtual void RunFor(int64_t time_ms, Packets* in_out) = 0;
+ const FlowIds& flow_ids() const { return flow_ids_; }
+
private:
PacketProcessorListener* listener_;
+ FlowIds flow_ids_;
DISALLOW_COPY_AND_ASSIGN(PacketProcessor);
};
@@ -199,6 +219,9 @@
explicit RateCounterFilter(PacketProcessorListener* listener);
RateCounterFilter(PacketProcessorListener* listener,
const std::string& name);
+ RateCounterFilter(PacketProcessorListener* listener,
+ const FlowIds& flow_ids,
+ const std::string& name);
virtual ~RateCounterFilter();
uint32_t packets_per_second() const;
@@ -211,7 +234,7 @@
private:
scoped_ptr<RateCounter> rate_counter_;
- Stats<double> pps_stats_;
+ Stats<double> packets_per_second_stats_;
Stats<double> kbps_stats_;
std::string name_;
@@ -283,6 +306,7 @@
class ChokeFilter : public PacketProcessor {
public:
explicit ChokeFilter(PacketProcessorListener* listener);
+ ChokeFilter(PacketProcessorListener* listener, const FlowIds& flow_ids);
virtual ~ChokeFilter();
void SetCapacity(uint32_t kbps);
@@ -315,6 +339,7 @@
void SetMaxDelay(int max_delay_ms);
Stats<double> GetDelayStats() const;
+ Stats<double> GetBitrateStats() const;
private:
void ProceedToNextSlot();
@@ -327,6 +352,8 @@
scoped_ptr<RateCounter> rate_counter_;
std::string name_;
scoped_ptr<DelayCapHelper> delay_cap_helper_;
+ Stats<double> packets_per_second_stats_;
+ Stats<double> kbps_stats_;
DISALLOW_COPY_AND_ASSIGN(TraceBasedDeliveryFilter);
};
@@ -338,6 +365,7 @@
};
explicit PacketSender(PacketProcessorListener* listener);
+ PacketSender(PacketProcessorListener* listener, const FlowIds& flow_ids);
virtual ~PacketSender() {}
virtual uint32_t GetCapacityKbps() const { return 0; }
@@ -347,23 +375,17 @@
// Note that changing the feedback interval affects the timing of when the
// output of the estimators is sampled and therefore the baseline files may
// have to be regenerated.
- virtual int64_t GetFeedbackIntervalMs() const { return 1000; }
+ virtual int GetFeedbackIntervalMs() const { return 1000; }
virtual void GiveFeedback(const Feedback& feedback) {}
private:
DISALLOW_COPY_AND_ASSIGN(PacketSender);
};
-struct PacketSenderFactory {
- PacketSenderFactory() {}
- virtual ~PacketSenderFactory() {}
- virtual PacketSender* Create() const = 0;
-};
-
class VideoSender : public PacketSender {
public:
- VideoSender(PacketProcessorListener* listener, float fps, uint32_t kbps,
- uint32_t ssrc, float first_frame_offset);
+ VideoSender(int flow_id, PacketProcessorListener* listener, float fps,
+ uint32_t kbps, uint32_t ssrc, float first_frame_offset);
virtual ~VideoSender() {}
uint32_t max_payload_size_bytes() const { return kMaxPayloadSizeBytes; }
@@ -390,48 +412,17 @@
class AdaptiveVideoSender : public VideoSender {
public:
- AdaptiveVideoSender(PacketProcessorListener* listener, float fps,
- uint32_t kbps, uint32_t ssrc, float first_frame_offset);
+ AdaptiveVideoSender(int flow_id, PacketProcessorListener* listener,
+ float fps, uint32_t kbps, uint32_t ssrc,
+ float first_frame_offset);
virtual ~AdaptiveVideoSender() {}
- virtual int64_t GetFeedbackIntervalMs() const { return 100; }
+ virtual int GetFeedbackIntervalMs() const { return 100; }
virtual void GiveFeedback(const Feedback& feedback);
private:
DISALLOW_IMPLICIT_CONSTRUCTORS(AdaptiveVideoSender);
};
-
-class VideoPacketSenderFactory : public PacketSenderFactory {
- public:
- VideoPacketSenderFactory(float fps, uint32_t kbps, uint32_t ssrc,
- float frame_offset)
- : fps_(fps),
- kbps_(kbps),
- ssrc_(ssrc),
- frame_offset_(frame_offset) {
- }
- virtual ~VideoPacketSenderFactory() {}
- virtual PacketSender* Create() const {
- return new VideoSender(NULL, fps_, kbps_, ssrc_, frame_offset_);
- }
- protected:
- float fps_;
- uint32_t kbps_;
- uint32_t ssrc_;
- float frame_offset_;
-};
-
-class AdaptiveVideoPacketSenderFactory : public VideoPacketSenderFactory {
- public:
- AdaptiveVideoPacketSenderFactory(float fps, uint32_t kbps, uint32_t ssrc,
- float frame_offset)
- : VideoPacketSenderFactory(fps, kbps, ssrc, frame_offset) {}
- virtual ~AdaptiveVideoPacketSenderFactory() {}
- virtual PacketSender* Create() const {
- return new AdaptiveVideoSender(NULL, fps_, kbps_, ssrc_, frame_offset_);
- }
-};
-
} // namespace bwe
} // namespace testing
} // namespace webrtc
diff --git a/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc b/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
index 56329b6..4ff35df 100644
--- a/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
+++ b/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
@@ -184,7 +184,7 @@
RTPHeader header;
// "Send" a packet every 10 ms.
for (int64_t i = 0; i < run_for_ms; i += 10, now_ms_ += 10) {
- packets.push_back(Packet(now_ms_ * 1000, payload_bits / 8, header));
+ packets.push_back(Packet(0, now_ms_ * 1000, payload_bits / 8, header));
}
filter_.RunFor(run_for_ms, &packets);
ASSERT_TRUE(IsTimeSorted(packets));
@@ -587,7 +587,7 @@
int64_t send_time_ms = now_ms_ + (i * run_for_ms) / packets_to_generate;
header.sequenceNumber = sequence_number_++;
// Payload is 1000 bits.
- packets.push_back(Packet(send_time_ms * 1000, 125, header));
+ packets.push_back(Packet(0, send_time_ms * 1000, 125, header));
send_times_us_.push_back(send_time_ms * 1000);
}
ASSERT_TRUE(IsTimeSorted(packets));
@@ -768,7 +768,7 @@
TEST(BweTestFramework_VideoSenderTest, Fps1Kbps80_1s) {
// 1 fps, 80 kbps
- VideoSender sender(NULL, 1.0f, 80, 0x1234, 0);
+ VideoSender sender(0, NULL, 1.0f, 80, 0x1234, 0);
EXPECT_EQ(10000u, sender.bytes_per_second());
// We're at 1 fps, so all packets should be generated on first call, giving 10
// packets of each 1000 bytes, total 10000 bytes.
@@ -785,7 +785,7 @@
TEST(BweTestFramework_VideoSenderTest, Fps1Kbps80_1s_Offset) {
// 1 fps, 80 kbps, offset 0.5 of a frame period, ==0.5s in this case.
- VideoSender sender(NULL, 1.0f, 80, 0x1234, 0.5f);
+ VideoSender sender(0, NULL, 1.0f, 80, 0x1234, 0.5f);
EXPECT_EQ(10000u, sender.bytes_per_second());
// 499ms, no output.
TestVideoSender(&sender, 499, 0, 0, 0);
@@ -805,7 +805,7 @@
TEST(BweTestFramework_VideoSenderTest, Fps50Kpbs80_11s) {
// 50 fps, 80 kbps.
- VideoSender sender(NULL, 50.0f, 80, 0x1234, 0);
+ VideoSender sender(0, NULL, 50.0f, 80, 0x1234, 0);
EXPECT_EQ(10000u, sender.bytes_per_second());
// 9998ms, should see 500 frames, 200 byte payloads, total 100000 bytes.
TestVideoSender(&sender, 9998, 500, 200, 100000);
@@ -821,7 +821,7 @@
TEST(BweTestFramework_VideoSenderTest, Fps10Kpbs120_1s) {
// 20 fps, 120 kbps.
- VideoSender sender(NULL, 20.0f, 120, 0x1234, 0);
+ VideoSender sender(0, NULL, 20.0f, 120, 0x1234, 0);
EXPECT_EQ(15000u, sender.bytes_per_second());
// 498ms, 10 frames with 750 byte payloads, total 7500 bytes.
TestVideoSender(&sender, 498, 10, 750, 7500);
@@ -837,7 +837,7 @@
TEST(BweTestFramework_VideoSenderTest, Fps30Kbps800_20s) {
// 20 fps, 820 kbps.
- VideoSender sender(NULL, 25.0f, 820, 0x1234, 0);
+ VideoSender sender(0, NULL, 25.0f, 820, 0x1234, 0);
EXPECT_EQ(102500u, sender.bytes_per_second());
// 9998ms, 250 frames. 820 kbps = 102500 bytes/s, so total should be 1025000.
// Each frame is 102500/25=4100 bytes, or 5 packets (4 @1000 bytes, 1 @100),
@@ -858,7 +858,7 @@
TEST(BweTestFramework_VideoSenderTest, TestAppendInOrder) {
// 1 fps, 80 kbps, 250ms offset.
- VideoSender sender1(NULL, 1.0f, 80, 0x1234, 0.25f);
+ VideoSender sender1(0, NULL, 1.0f, 80, 0x1234, 0.25f);
EXPECT_EQ(10000u, sender1.bytes_per_second());
Packets packets;
// Generate some packets, verify they are sorted.
@@ -873,7 +873,7 @@
EXPECT_EQ(18u, packets.size());
// Another sender, 2 fps, 160 kpbs, 150ms offset
- VideoSender sender2(NULL, 2.0f, 160, 0x2234, 0.30f);
+ VideoSender sender2(0, NULL, 2.0f, 160, 0x2234, 0.30f);
EXPECT_EQ(20000u, sender2.bytes_per_second());
// Generate some packets, verify that they are merged with the packets already
// on the list.
@@ -887,7 +887,7 @@
}
TEST(BweTestFramework_VideoSenderTest, FeedbackIneffective) {
- VideoSender sender(NULL, 25.0f, 820, 0x1234, 0);
+ VideoSender sender(0, NULL, 25.0f, 820, 0x1234, 0);
EXPECT_EQ(102500u, sender.bytes_per_second());
TestVideoSender(&sender, 9998, 1000, 500, 1025000);
@@ -899,7 +899,7 @@
}
TEST(BweTestFramework_AdaptiveVideoSenderTest, FeedbackChangesBitrate) {
- AdaptiveVideoSender sender(NULL, 25.0f, 820, 0x1234, 0);
+ AdaptiveVideoSender sender(0, NULL, 25.0f, 820, 0x1234, 0);
EXPECT_EQ(102500u, sender.bytes_per_second());
TestVideoSender(&sender, 9998, 1000, 500, 1025000);
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/rtp_rtcp.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/rtp_rtcp.target.darwin-arm64.mk
index 3b05cd9..987f969 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/rtp_rtcp.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_rtp_rtcp_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,41 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/rtp_rtcp/source/bitrate.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/ssrc_database.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/producer_fec.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
# Flags passed to both C and C++ files.
@@ -70,6 +67,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +76,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +87,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +118,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +136,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +164,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +173,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +184,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +215,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +234,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +262,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +276,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +298,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_rtp_rtcp_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: rtp_rtcp
+rtp_rtcp: third_party_webrtc_modules_rtp_rtcp_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/rtp_rtcp.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/rtp_rtcp.target.linux-arm64.mk
index 3b05cd9..987f969 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/rtp_rtcp.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_rtp_rtcp_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,41 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/rtp_rtcp/source/bitrate.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtcp_utility.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_sender.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_utility.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/ssrc_database.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/tmmbr_help.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/dtmf_queue.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/producer_fec.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc \
+ third_party/webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
# Flags passed to both C and C++ files.
@@ -70,6 +67,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +76,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +87,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +118,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +136,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +164,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +173,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +184,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +215,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +234,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +262,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +276,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +298,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_rtp_rtcp_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: rtp_rtcp
+rtp_rtcp: third_party_webrtc_modules_rtp_rtcp_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index dc65336..70a3443 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -733,7 +733,12 @@
uint16_t* avg_rtt,
uint16_t* min_rtt,
uint16_t* max_rtt) const {
- return rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
+ int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
+ if (rtt && *rtt == 0) {
+ // Try to get RTT from RtcpRttStats class.
+ *rtt = static_cast<uint16_t>(rtt_ms());
+ }
+ return ret;
}
// Reset RoundTripTime statistics.
diff --git a/modules/rtp_rtcp/source/rtp_utility.cc b/modules/rtp_rtcp/source/rtp_utility.cc
index 3f99845..c1f3c64 100644
--- a/modules/rtp_rtcp/source/rtp_utility.cc
+++ b/modules/rtp_rtcp/source/rtp_utility.cc
@@ -472,7 +472,8 @@
RTPExtensionType type;
if (ptrExtensionMap->GetType(id, &type) != 0) {
// If we encounter an unknown extension, just skip over it.
- LOG(LS_WARNING) << "Failed to find extension id: " << id;
+ LOG(LS_WARNING) << "Failed to find extension id: "
+ << static_cast<int>(id);
} else {
switch (type) {
case kRtpExtensionTransmissionTimeOffset: {
diff --git a/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc b/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
index cc1ee84..2fb0968 100644
--- a/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
+++ b/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
@@ -198,7 +198,7 @@
int RecoveredMediaPackets(int num_media_packets,
int num_fec_packets,
uint8_t* state) {
- scoped_array<uint8_t> state_tmp(
+ scoped_ptr<uint8_t[]> state_tmp(
new uint8_t[num_media_packets + num_fec_packets]);
memcpy(state_tmp.get(), state, num_media_packets + num_fec_packets);
int num_recovered_packets = 0;
@@ -392,7 +392,7 @@
// (which containes the code size parameters/protection length).
void ComputeMetricsForCode(CodeType code_type,
int code_index) {
- scoped_array<double> prob_weight(new double[kNumLossModels]);
+ scoped_ptr<double[]> prob_weight(new double[kNumLossModels]);
memset(prob_weight.get() , 0, sizeof(double) * kNumLossModels);
MetricsFecCode metrics_code;
SetMetricsZero(&metrics_code);
@@ -400,7 +400,7 @@
int num_media_packets = code_params_[code_index].num_media_packets;
int num_fec_packets = code_params_[code_index].num_fec_packets;
int tot_num_packets = num_media_packets + num_fec_packets;
- scoped_array<uint8_t> state(new uint8_t[tot_num_packets]);
+ scoped_ptr<uint8_t[]> state(new uint8_t[tot_num_packets]);
memset(state.get() , 0, tot_num_packets);
int num_loss_configurations = static_cast<int>(pow(2.0f, tot_num_packets));
diff --git a/modules/utility/source/video_frames_queue.cc b/modules/utility/source/video_frames_queue.cc
index fbcbd2c..63afbe9 100644
--- a/modules/utility/source/video_frames_queue.cc
+++ b/modules/utility/source/video_frames_queue.cc
@@ -80,8 +80,10 @@
// List is traversed beginning to end. If ptrRenderFrame is not
// NULL it must be the first, and thus oldest, VideoFrame in the
// queue. It can be recycled.
- ReturnFrame(ptrRenderFrame);
- iter = _incomingFrames.erase(iter);
+ if (ptrRenderFrame) {
+ ReturnFrame(ptrRenderFrame);
+ _incomingFrames.pop_front();
+ }
ptrRenderFrame = ptrOldestFrameInList;
} else {
// All VideoFrames following this one will be even newer. No match
diff --git a/modules/video_capture/test/video_capture_unittest.cc b/modules/video_capture/test/video_capture_unittest.cc
index db0c818..4c2263d 100644
--- a/modules/video_capture/test/video_capture_unittest.cc
+++ b/modules/video_capture/test/video_capture_unittest.cc
@@ -25,6 +25,7 @@
#include "webrtc/system_wrappers/interface/scoped_refptr.h"
#include "webrtc/system_wrappers/interface/sleep.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/test/testsupport/gtest_disable.h"
using webrtc::CriticalSectionWrapper;
using webrtc::CriticalSectionScoped;
@@ -473,7 +474,7 @@
unsigned int length = webrtc::CalcBufferSize(webrtc::kI420,
test_frame_.width(),
test_frame_.height());
- webrtc::scoped_array<uint8_t> test_buffer(new uint8_t[length]);
+ webrtc::scoped_ptr<uint8_t[]> test_buffer(new uint8_t[length]);
webrtc::ExtractBuffer(test_frame_, length, test_buffer.get());
EXPECT_EQ(0, capture_input_interface_->IncomingFrame(test_buffer.get(),
length, capture_callback_.capability(), 0));
@@ -550,7 +551,8 @@
}
// Test frame rate and no picture alarm.
-TEST_F(VideoCaptureExternalTest , FrameRate) {
+// Flaky on Win32, see webrtc:3270.
+TEST_F(VideoCaptureExternalTest, DISABLED_ON_WIN(FrameRate)) {
int64_t testTime = 3;
TickTime startTime = TickTime::Now();
@@ -558,7 +560,7 @@
unsigned int length = webrtc::CalcBufferSize(webrtc::kI420,
test_frame_.width(),
test_frame_.height());
- webrtc::scoped_array<uint8_t> test_buffer(new uint8_t[length]);
+ webrtc::scoped_ptr<uint8_t[]> test_buffer(new uint8_t[length]);
webrtc::ExtractBuffer(test_frame_, length, test_buffer.get());
EXPECT_EQ(0, capture_input_interface_->IncomingFrame(test_buffer.get(),
length, capture_callback_.capability(), 0));
@@ -574,7 +576,7 @@
unsigned int length = webrtc::CalcBufferSize(webrtc::kI420,
test_frame_.width(),
test_frame_.height());
- webrtc::scoped_array<uint8_t> test_buffer(new uint8_t[length]);
+ webrtc::scoped_ptr<uint8_t[]> test_buffer(new uint8_t[length]);
webrtc::ExtractBuffer(test_frame_, length, test_buffer.get());
EXPECT_EQ(0, capture_input_interface_->IncomingFrame(test_buffer.get(),
length, capture_callback_.capability(), 0));
@@ -592,7 +594,7 @@
unsigned int length = webrtc::CalcBufferSize(webrtc::kI420,
test_frame_.width(),
test_frame_.height());
- webrtc::scoped_array<uint8_t> test_buffer(new uint8_t[length]);
+ webrtc::scoped_ptr<uint8_t[]> test_buffer(new uint8_t[length]);
webrtc::ExtractBuffer(test_frame_, length, test_buffer.get());
EXPECT_EQ(0, capture_input_interface_->IncomingFrame(test_buffer.get(),
length, capture_callback_.capability(), 0));
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/video_capture_module.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/video_capture_module.target.darwin-arm64.mk
index c84869c..a260b9b 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/video_capture_module.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_video_capture_module_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,21 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_capture/device_info_impl.cc \
+ third_party/webrtc/modules/video_capture/video_capture_factory.cc \
+ third_party/webrtc/modules/video_capture/video_capture_impl.cc \
+ third_party/webrtc/modules/video_capture/external/device_info_external.cc \
+ third_party/webrtc/modules/video_capture/external/video_capture_external.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +52,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +94,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +112,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +132,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +151,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +193,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +212,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +242,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +256,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +278,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_video_capture_module_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: video_capture_module
+video_capture_module: third_party_webrtc_modules_video_capture_module_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/video_capture_module.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/video_capture_module.target.linux-arm64.mk
index c84869c..a260b9b 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/video_capture_module.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_video_capture_module_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,21 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_capture/device_info_impl.cc \
+ third_party/webrtc/modules/video_capture/video_capture_factory.cc \
+ third_party/webrtc/modules/video_capture/video_capture_impl.cc \
+ third_party/webrtc/modules/video_capture/external/device_info_external.cc \
+ third_party/webrtc/modules/video_capture/external/video_capture_external.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +52,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +94,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +112,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +132,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +151,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +193,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +212,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +242,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +256,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +278,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_video_capture_module_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: video_capture_module
+video_capture_module: third_party_webrtc_modules_video_capture_module_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/video_coding/codecs/test/videoprocessor.cc b/modules/video_coding/codecs/test/videoprocessor.cc
index 30ee6a8..93738ca 100644
--- a/modules/video_coding/codecs/test/videoprocessor.cc
+++ b/modules/video_coding/codecs/test/videoprocessor.cc
@@ -331,7 +331,7 @@
}
// TODO(mikhal): Extracting the buffer for now - need to update test.
int length = CalcBufferSize(kI420, up_image.width(), up_image.height());
- scoped_array<uint8_t> image_buffer(new uint8_t[length]);
+ scoped_ptr<uint8_t[]> image_buffer(new uint8_t[length]);
length = ExtractBuffer(up_image, length, image_buffer.get());
// Update our copy of the last successful frame:
memcpy(last_successful_frame_buffer_, image_buffer.get(), length);
@@ -344,7 +344,7 @@
// Update our copy of the last successful frame:
// TODO(mikhal): Add as a member function, so won't be allocated per frame.
int length = CalcBufferSize(kI420, image.width(), image.height());
- scoped_array<uint8_t> image_buffer(new uint8_t[length]);
+ scoped_ptr<uint8_t[]> image_buffer(new uint8_t[length]);
length = ExtractBuffer(image, length, image_buffer.get());
assert(length > 0);
memcpy(last_successful_frame_buffer_, image_buffer.get(), length);
diff --git a/modules/video_coding/codecs/test_framework/unit_test.cc b/modules/video_coding/codecs/test_framework/unit_test.cc
index 3b034e0..ec12a51 100644
--- a/modules/video_coding/codecs/test_framework/unit_test.cc
+++ b/modules/video_coding/codecs/test_framework/unit_test.cc
@@ -565,7 +565,7 @@
frameLength = WaitForDecodedFrame();
}
unsigned int length = CalcBufferSize(kI420, width, height);
- scoped_array<uint8_t> decoded_buffer(new uint8_t[length]);
+ scoped_ptr<uint8_t[]> decoded_buffer(new uint8_t[length]);
ExtractBuffer(_decodedVideoBuffer, _lengthSourceFrame,
decoded_buffer.get());
EXPECT_TRUE(CheckIfBitExact(decoded_buffer.get(), frameLength, _refDecFrame,
@@ -645,7 +645,7 @@
// check that decoded frame matches with reference
unsigned int length = CalcBufferSize(kI420, width, height);
- scoped_array<uint8_t> decoded_buffer(new uint8_t[length]);
+ scoped_ptr<uint8_t[]> decoded_buffer(new uint8_t[length]);
ExtractBuffer(_decodedVideoBuffer, length, decoded_buffer.get());
EXPECT_TRUE(CheckIfBitExact(decoded_buffer.get(), length,
_refDecFrame, _lengthSourceFrame) == true);
diff --git a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
index ff99ed2..3cc8bc3 100644
--- a/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
+++ b/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
@@ -181,7 +181,7 @@
scoped_ptr<Vp8UnitTestEncodeCompleteCallback> encode_complete_callback_;
scoped_ptr<Vp8UnitTestDecodeCompleteCallback> decode_complete_callback_;
- scoped_array<uint8_t> source_buffer_;
+ scoped_ptr<uint8_t[]> source_buffer_;
FILE* source_file_;
I420VideoFrame input_frame_;
scoped_ptr<VideoEncoder> encoder_;
diff --git a/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc b/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
index 1bd3e1a..ffa0bcc 100644
--- a/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
+++ b/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
@@ -142,7 +142,7 @@
EXPECT_EQ(0, decoder->InitDecode(&inst, 1));
webrtc::I420VideoFrame input_frame;
unsigned int length = webrtc::CalcBufferSize(webrtc::kI420, width, height);
- webrtc::scoped_array<uint8_t> frame_buffer(new uint8_t[length]);
+ webrtc::scoped_ptr<uint8_t[]> frame_buffer(new uint8_t[length]);
int half_width = (width + 1) / 2;
// Set and register callbacks.
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/video_coding/codecs/vp8/webrtc_vp8.target.darwin-arm64.mk
similarity index 63%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/video_coding/codecs/vp8/webrtc_vp8.target.darwin-arm64.mk
index c84869c..7cfaea8 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/video_coding/codecs/vp8/webrtc_vp8.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_video_coding_codecs_vp8_webrtc_vp8_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,21 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.cc \
+ third_party/webrtc/modules/video_coding/codecs/vp8/vp8_factory.cc \
+ third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc \
+ third_party/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc \
+ third_party/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,17 +48,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +108,10 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/libvpx/source/libvpx \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +129,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -185,17 +144,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +186,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +205,10 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/libvpx/source/libvpx \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +236,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +250,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +272,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_video_coding_codecs_vp8_webrtc_vp8_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: webrtc_vp8
+webrtc_vp8: third_party_webrtc_modules_video_coding_codecs_vp8_webrtc_vp8_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/video_coding/codecs/vp8/webrtc_vp8.target.linux-arm64.mk
similarity index 63%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/video_coding/codecs/vp8/webrtc_vp8.target.linux-arm64.mk
index c84869c..7cfaea8 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/video_coding/codecs/vp8/webrtc_vp8.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_video_coding_codecs_vp8_webrtc_vp8_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,21 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_coding/codecs/vp8/reference_picture_selection.cc \
+ third_party/webrtc/modules/video_coding/codecs/vp8/vp8_factory.cc \
+ third_party/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc \
+ third_party/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc \
+ third_party/webrtc/modules/video_coding/codecs/vp8/realtime_temporal_layers.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,17 +48,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +108,10 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/libvpx/source/libvpx \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +129,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -185,17 +144,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +186,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +205,10 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
+ $(LOCAL_PATH)/third_party/libvpx/source/libvpx \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +236,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +250,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +272,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_video_coding_codecs_vp8_webrtc_vp8_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: webrtc_vp8
+webrtc_vp8: third_party_webrtc_modules_video_coding_codecs_vp8_webrtc_vp8_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/video_coding/main/test/rtp_player.cc b/modules/video_coding/main/test/rtp_player.cc
index e314a7c..f02aebb 100644
--- a/modules/video_coding/main/test/rtp_player.cc
+++ b/modules/video_coding/main/test/rtp_player.cc
@@ -61,7 +61,7 @@
uint16_t seq_num() const { return seq_num_; }
private:
- scoped_array<uint8_t> data_;
+ scoped_ptr<uint8_t[]> data_;
uint32_t length_;
int64_t resend_time_ms_;
uint32_t ssrc_;
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/video_coding/utility/video_coding_utility.target.darwin-arm64.mk
similarity index 63%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/video_coding/utility/video_coding_utility.target.darwin-arm64.mk
index c84869c..59c6a59 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/video_coding/utility/video_coding_utility.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_video_coding_utility_video_coding_utility_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,18 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_coding/utility/exp_filter.cc \
+ third_party/webrtc/modules/video_coding/utility/frame_dropper.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,17 +45,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +87,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +105,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +123,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -185,17 +138,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +180,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +199,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +227,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +241,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +263,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_video_coding_utility_video_coding_utility_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: video_coding_utility
+video_coding_utility: third_party_webrtc_modules_video_coding_utility_video_coding_utility_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/video_coding/utility/video_coding_utility.target.linux-arm64.mk
similarity index 63%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/video_coding/utility/video_coding_utility.target.linux-arm64.mk
index c84869c..59c6a59 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/video_coding/utility/video_coding_utility.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_video_coding_utility_video_coding_utility_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,18 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_coding/utility/exp_filter.cc \
+ third_party/webrtc/modules/video_coding/utility/frame_dropper.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,17 +45,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +87,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +105,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +123,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -185,17 +138,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +180,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +199,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +227,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +241,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +263,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_video_coding_utility_video_coding_utility_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: video_coding_utility
+video_coding_utility: third_party_webrtc_modules_video_coding_utility_video_coding_utility_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/video_processing.target.darwin-arm64.mk
similarity index 69%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/video_processing.target.darwin-arm64.mk
index 3b05cd9..359aade 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/video_processing.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_video_processing_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,22 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_processing/main/source/brighten.cc \
+ third_party/webrtc/modules/video_processing/main/source/brightness_detection.cc \
+ third_party/webrtc/modules/video_processing/main/source/color_enhancement.cc \
+ third_party/webrtc/modules/video_processing/main/source/content_analysis.cc \
+ third_party/webrtc/modules/video_processing/main/source/deflickering.cc \
+ third_party/webrtc/modules/video_processing/main/source/denoising.cc \
+ third_party/webrtc/modules/video_processing/main/source/frame_preprocessor.cc \
+ third_party/webrtc/modules/video_processing/main/source/spatial_resampler.cc \
+ third_party/webrtc/modules/video_processing/main/source/video_decimator.cc \
+ third_party/webrtc/modules/video_processing/main/source/video_processing_impl.cc
# Flags passed to both C and C++ files.
@@ -70,6 +48,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +57,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +68,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +99,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +117,12 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +150,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +159,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +170,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +201,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +220,12 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +253,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +267,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +289,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_video_processing_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: video_processing
+video_processing: third_party_webrtc_modules_video_processing_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/video_processing.target.linux-arm64.mk
similarity index 69%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/video_processing.target.linux-arm64.mk
index 3b05cd9..359aade 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/video_processing.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_video_processing_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,22 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_processing/main/source/brighten.cc \
+ third_party/webrtc/modules/video_processing/main/source/brightness_detection.cc \
+ third_party/webrtc/modules/video_processing/main/source/color_enhancement.cc \
+ third_party/webrtc/modules/video_processing/main/source/content_analysis.cc \
+ third_party/webrtc/modules/video_processing/main/source/deflickering.cc \
+ third_party/webrtc/modules/video_processing/main/source/denoising.cc \
+ third_party/webrtc/modules/video_processing/main/source/frame_preprocessor.cc \
+ third_party/webrtc/modules/video_processing/main/source/spatial_resampler.cc \
+ third_party/webrtc/modules/video_processing/main/source/video_decimator.cc \
+ third_party/webrtc/modules/video_processing/main/source/video_processing_impl.cc
# Flags passed to both C and C++ files.
@@ -70,6 +48,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +57,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +68,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +99,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +117,12 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +150,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +159,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +170,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +201,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +220,12 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +253,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +267,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +289,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_video_processing_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: video_processing
+video_processing: third_party_webrtc_modules_video_processing_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/video_processing/main/test/unit_test/brightness_detection_test.cc b/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
index d7ac729..c53c1fb 100644
--- a/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
+++ b/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
@@ -19,7 +19,7 @@
uint32_t frameNum = 0;
int32_t brightnessWarning = 0;
uint32_t warningCount = 0;
- scoped_array<uint8_t> video_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
frame_length_)
{
diff --git a/modules/video_processing/main/test/unit_test/color_enhancement_test.cc b/modules/video_processing/main/test/unit_test/color_enhancement_test.cc
index fc560be..c1cd462 100644
--- a/modules/video_processing/main/test/unit_test/color_enhancement_test.cc
+++ b/modules/video_processing/main/test/unit_test/color_enhancement_test.cc
@@ -39,7 +39,7 @@
ASSERT_TRUE(modFile != NULL) << "Could not open output file.\n";
uint32_t frameNum = 0;
- scoped_array<uint8_t> video_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
frame_length_)
{
@@ -86,7 +86,7 @@
width_, half_width_, half_width_);
// Compare frame-by-frame.
- scoped_array<uint8_t> ref_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> ref_buffer(new uint8_t[frame_length_]);
while (fread(video_buffer.get(), 1, frame_length_, modFile) ==
frame_length_)
{
@@ -114,7 +114,7 @@
// Verify that all color pixels are enhanced, and no luminance values are
// altered.
- scoped_array<uint8_t> testFrame(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> testFrame(new uint8_t[frame_length_]);
// Use value 128 as probe value, since we know that this will be changed
// in the enhancement.
diff --git a/modules/video_processing/main/test/unit_test/content_metrics_test.cc b/modules/video_processing/main/test/unit_test/content_metrics_test.cc
index 36a1ad7..c0d1ab4 100644
--- a/modules/video_processing/main/test/unit_test/content_metrics_test.cc
+++ b/modules/video_processing/main/test/unit_test/content_metrics_test.cc
@@ -23,7 +23,7 @@
ca__c.Initialize(width_,height_);
ca__sse.Initialize(width_,height_);
- scoped_array<uint8_t> video_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
while (fread(video_buffer.get(), 1, frame_length_, source_file_)
== frame_length_) {
// Using ConvertToI420 to add stride to the image.
diff --git a/modules/video_processing/main/test/unit_test/deflickering_test.cc b/modules/video_processing/main/test/unit_test/deflickering_test.cc
index 0fa3f48..1bf53fc 100644
--- a/modules/video_processing/main/test/unit_test/deflickering_test.cc
+++ b/modules/video_processing/main/test/unit_test/deflickering_test.cc
@@ -43,7 +43,7 @@
"Could not open output file: " << output_file << "\n";
printf("\nRun time [us / frame]:\n");
- scoped_array<uint8_t> video_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
for (uint32_t run_idx = 0; run_idx < NumRuns; run_idx++)
{
TickTime t0;
diff --git a/modules/video_processing/main/test/unit_test/denoising_test.cc b/modules/video_processing/main/test/unit_test/denoising_test.cc
index 3023a2d..c00db6a 100644
--- a/modules/video_processing/main/test/unit_test/denoising_test.cc
+++ b/modules/video_processing/main/test/unit_test/denoising_test.cc
@@ -49,7 +49,7 @@
int32_t modifiedPixels = 0;
frameNum = 0;
- scoped_array<uint8_t> video_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
while (fread(video_buffer.get(), 1, frame_length_, source_file_) ==
frame_length_)
{
diff --git a/modules/video_processing/main/test/unit_test/video_processing_unittest.cc b/modules/video_processing/main/test/unit_test/video_processing_unittest.cc
index 6e54923..9d70e67 100644
--- a/modules/video_processing/main/test/unit_test/video_processing_unittest.cc
+++ b/modules/video_processing/main/test/unit_test/video_processing_unittest.cc
@@ -89,7 +89,7 @@
TEST_F(VideoProcessingModuleTest, HandleBadStats) {
VideoProcessingModule::FrameStats stats;
- scoped_array<uint8_t> video_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
source_file_));
EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0,
@@ -129,7 +129,7 @@
I420VideoFrame video_frame2;
VideoProcessingModule::FrameStats stats;
// Only testing non-static functions here.
- scoped_array<uint8_t> video_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
source_file_));
EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0,
@@ -172,7 +172,7 @@
TEST_F(VideoProcessingModuleTest, FrameStats) {
VideoProcessingModule::FrameStats stats;
- scoped_array<uint8_t> video_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
source_file_));
EXPECT_EQ(0, ConvertToI420(kI420, video_buffer.get(), 0, 0,
@@ -242,7 +242,7 @@
vpm_->EnableTemporalDecimation(false);
// Reading test frame
- scoped_array<uint8_t> video_buffer(new uint8_t[frame_length_]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[frame_length_]);
ASSERT_EQ(frame_length_, fread(video_buffer.get(), 1, frame_length_,
source_file_));
// Using ConvertToI420 to add stride to the image.
diff --git a/modules/video_render/video_render.gypi b/modules/video_render/video_render.gypi
index a3c0f6d..3cb6167 100644
--- a/modules/video_render/video_render.gypi
+++ b/modules/video_render/video_render.gypi
@@ -87,6 +87,12 @@
'android/video_render_android_surface_view.cc',
'android/video_render_opengles20.cc',
],
+ }, {
+ 'all_dependent_settings': {
+ 'libraries': [
+ '-lGLESv2',
+ ],
+ },
}],
['OS!="ios" or include_internal_video_render==0', {
'sources!': [
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/video_render_module.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/video_render_module.target.darwin-arm64.mk
index c84869c..4b1d803 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/video_render_module.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_video_render_module_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,20 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_render/external/video_render_external_impl.cc \
+ third_party/webrtc/modules/video_render/incoming_video_stream.cc \
+ third_party/webrtc/modules/video_render/video_render_frames.cc \
+ third_party/webrtc/modules/video_render/video_render_impl.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +51,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +93,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +111,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +131,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +150,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +192,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +211,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +241,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +255,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +277,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_video_render_module_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: video_render_module
+video_render_module: third_party_webrtc_modules_video_render_module_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/video_render_module.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/video_render_module.target.linux-arm64.mk
index c84869c..4b1d803 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/video_render_module.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_video_render_module_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,20 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_render/external/video_render_external_impl.cc \
+ third_party/webrtc/modules/video_render/incoming_video_stream.cc \
+ third_party/webrtc/modules/video_render/video_render_frames.cc \
+ third_party/webrtc/modules/video_render/video_render_impl.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +51,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +93,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +111,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +131,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +150,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +192,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +211,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +241,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +255,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +277,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_video_render_module_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: video_render_module
+video_render_module: third_party_webrtc_modules_video_render_module_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/webrtc_i420.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/webrtc_i420.target.darwin-arm64.mk
index c84869c..b234379 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/webrtc_i420.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_webrtc_i420_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,17 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_coding/codecs/i420/main/source/i420.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +48,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +108,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +126,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +145,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +187,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +206,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +234,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +248,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +270,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_webrtc_i420_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: webrtc_i420
+webrtc_i420: third_party_webrtc_modules_webrtc_i420_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/webrtc_i420.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/webrtc_i420.target.linux-arm64.mk
index c84869c..b234379 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/webrtc_i420.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_webrtc_i420_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,17 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_coding/codecs/i420/main/source/i420.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +48,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +90,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +108,7 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +126,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +145,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +187,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +206,7 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +234,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +248,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +270,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_webrtc_i420_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: webrtc_i420
+webrtc_i420: third_party_webrtc_modules_webrtc_i420_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/webrtc_opus.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/webrtc_opus.target.darwin-arm64.mk
index 3b05cd9..dd361fc 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/webrtc_opus.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_webrtc_opus_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,7 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
# Flags passed to both C and C++ files.
@@ -70,6 +38,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +47,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +58,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +89,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +107,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +136,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +145,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +156,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +187,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +206,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +235,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +249,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +271,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_webrtc_opus_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: webrtc_opus
+webrtc_opus: third_party_webrtc_modules_webrtc_opus_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/webrtc_opus.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/webrtc_opus.target.linux-arm64.mk
index 3b05cd9..dd361fc 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/webrtc_opus.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_webrtc_opus_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,7 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
# Flags passed to both C and C++ files.
@@ -70,6 +38,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +47,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +58,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +89,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +107,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +136,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +145,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +156,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +187,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +206,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/opus/src/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +235,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +249,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +271,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_webrtc_opus_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: webrtc_opus
+webrtc_opus: third_party_webrtc_modules_webrtc_opus_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/webrtc_utility.target.darwin-arm64.mk
similarity index 70%
rename from modules/NetEq.target.darwin-x86_64.mk
rename to modules/webrtc_utility.target.darwin-arm64.mk
index c84869c..ea90c75 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/webrtc_utility.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_webrtc_utility_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,26 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/utility/source/audio_frame_operations.cc \
+ third_party/webrtc/modules/utility/source/coder.cc \
+ third_party/webrtc/modules/utility/source/file_player_impl.cc \
+ third_party/webrtc/modules/utility/source/file_recorder_impl.cc \
+ third_party/webrtc/modules/utility/source/helpers_android.cc \
+ third_party/webrtc/modules/utility/source/process_thread_impl.cc \
+ third_party/webrtc/modules/utility/source/rtp_dump_impl.cc \
+ third_party/webrtc/modules/utility/source/frame_scaler.cc \
+ third_party/webrtc/modules/utility/source/video_coder.cc \
+ third_party/webrtc/modules/utility/source/video_frames_queue.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +57,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +99,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +117,14 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +142,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +161,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +203,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +222,14 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +257,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +271,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +293,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_webrtc_utility_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: webrtc_utility
+webrtc_utility: third_party_webrtc_modules_webrtc_utility_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/modules/webrtc_utility.target.linux-arm64.mk
similarity index 70%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to modules/webrtc_utility.target.linux-arm64.mk
index c84869c..ea90c75 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/modules/webrtc_utility.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_webrtc_utility_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,26 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/utility/source/audio_frame_operations.cc \
+ third_party/webrtc/modules/utility/source/coder.cc \
+ third_party/webrtc/modules/utility/source/file_player_impl.cc \
+ third_party/webrtc/modules/utility/source/file_recorder_impl.cc \
+ third_party/webrtc/modules/utility/source/helpers_android.cc \
+ third_party/webrtc/modules/utility/source/process_thread_impl.cc \
+ third_party/webrtc/modules/utility/source/rtp_dump_impl.cc \
+ third_party/webrtc/modules/utility/source/frame_scaler.cc \
+ third_party/webrtc/modules/utility/source/video_coder.cc \
+ third_party/webrtc/modules/utility/source/video_frames_queue.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -80,13 +57,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +99,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +117,14 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +142,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -189,13 +161,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +203,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +222,14 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
$(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +257,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +271,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +293,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_webrtc_utility_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: webrtc_utility
+webrtc_utility: third_party_webrtc_modules_webrtc_utility_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/webrtc_video_coding.target.darwin-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/webrtc_video_coding.target.darwin-arm64.mk
index 3b05cd9..0f62005 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/webrtc_video_coding.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_webrtc_video_coding_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,36 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_coding/main/source/codec_database.cc \
+ third_party/webrtc/modules/video_coding/main/source/codec_timer.cc \
+ third_party/webrtc/modules/video_coding/main/source/content_metrics_processing.cc \
+ third_party/webrtc/modules/video_coding/main/source/decoding_state.cc \
+ third_party/webrtc/modules/video_coding/main/source/encoded_frame.cc \
+ third_party/webrtc/modules/video_coding/main/source/frame_buffer.cc \
+ third_party/webrtc/modules/video_coding/main/source/generic_decoder.cc \
+ third_party/webrtc/modules/video_coding/main/source/generic_encoder.cc \
+ third_party/webrtc/modules/video_coding/main/source/inter_frame_delay.cc \
+ third_party/webrtc/modules/video_coding/main/source/jitter_buffer.cc \
+ third_party/webrtc/modules/video_coding/main/source/jitter_estimator.cc \
+ third_party/webrtc/modules/video_coding/main/source/media_opt_util.cc \
+ third_party/webrtc/modules/video_coding/main/source/media_optimization.cc \
+ third_party/webrtc/modules/video_coding/main/source/packet.cc \
+ third_party/webrtc/modules/video_coding/main/source/qm_select.cc \
+ third_party/webrtc/modules/video_coding/main/source/receiver.cc \
+ third_party/webrtc/modules/video_coding/main/source/rtt_filter.cc \
+ third_party/webrtc/modules/video_coding/main/source/session_info.cc \
+ third_party/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc \
+ third_party/webrtc/modules/video_coding/main/source/timestamp_map.cc \
+ third_party/webrtc/modules/video_coding/main/source/timing.cc \
+ third_party/webrtc/modules/video_coding/main/source/video_coding_impl.cc \
+ third_party/webrtc/modules/video_coding/main/source/video_sender.cc \
+ third_party/webrtc/modules/video_coding/main/source/video_receiver.cc
# Flags passed to both C and C++ files.
@@ -70,6 +62,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +71,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +82,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +113,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +131,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +161,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +170,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +181,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +212,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +231,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +261,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +275,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +297,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_webrtc_video_coding_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: webrtc_video_coding
+webrtc_video_coding: third_party_webrtc_modules_webrtc_video_coding_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/modules/webrtc_video_coding.target.linux-arm64.mk
similarity index 65%
copy from modules/NetEq.target.darwin-x86.mk
copy to modules/webrtc_video_coding.target.linux-arm64.mk
index 3b05cd9..0f62005 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/modules/webrtc_video_coding.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_modules_webrtc_video_coding_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,36 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/modules/video_coding/main/source/codec_database.cc \
+ third_party/webrtc/modules/video_coding/main/source/codec_timer.cc \
+ third_party/webrtc/modules/video_coding/main/source/content_metrics_processing.cc \
+ third_party/webrtc/modules/video_coding/main/source/decoding_state.cc \
+ third_party/webrtc/modules/video_coding/main/source/encoded_frame.cc \
+ third_party/webrtc/modules/video_coding/main/source/frame_buffer.cc \
+ third_party/webrtc/modules/video_coding/main/source/generic_decoder.cc \
+ third_party/webrtc/modules/video_coding/main/source/generic_encoder.cc \
+ third_party/webrtc/modules/video_coding/main/source/inter_frame_delay.cc \
+ third_party/webrtc/modules/video_coding/main/source/jitter_buffer.cc \
+ third_party/webrtc/modules/video_coding/main/source/jitter_estimator.cc \
+ third_party/webrtc/modules/video_coding/main/source/media_opt_util.cc \
+ third_party/webrtc/modules/video_coding/main/source/media_optimization.cc \
+ third_party/webrtc/modules/video_coding/main/source/packet.cc \
+ third_party/webrtc/modules/video_coding/main/source/qm_select.cc \
+ third_party/webrtc/modules/video_coding/main/source/receiver.cc \
+ third_party/webrtc/modules/video_coding/main/source/rtt_filter.cc \
+ third_party/webrtc/modules/video_coding/main/source/session_info.cc \
+ third_party/webrtc/modules/video_coding/main/source/timestamp_extrapolator.cc \
+ third_party/webrtc/modules/video_coding/main/source/timestamp_map.cc \
+ third_party/webrtc/modules/video_coding/main/source/timing.cc \
+ third_party/webrtc/modules/video_coding/main/source/video_coding_impl.cc \
+ third_party/webrtc/modules/video_coding/main/source/video_sender.cc \
+ third_party/webrtc/modules/video_coding/main/source/video_receiver.cc
# Flags passed to both C and C++ files.
@@ -70,6 +62,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -78,11 +71,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +82,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +113,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +131,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,6 +161,7 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
@@ -187,11 +170,6 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +181,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +212,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +231,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +261,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +275,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +297,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_modules_webrtc_video_coding_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: webrtc_video_coding
+webrtc_video_coding: third_party_webrtc_modules_webrtc_video_coding_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/supplement.gypi b/supplement.gypi
index 7898d4c..2fad574 100644
--- a/supplement.gypi
+++ b/supplement.gypi
@@ -1,5 +1,6 @@
{
'variables': {
'build_with_chromium': 0,
+ 'use_sanitizer_options': 0,
}
}
diff --git a/system_wrappers/interface/scoped_ptr.h b/system_wrappers/interface/scoped_ptr.h
index fb20363..8998f81 100644
--- a/system_wrappers/interface/scoped_ptr.h
+++ b/system_wrappers/interface/scoped_ptr.h
@@ -563,82 +563,4 @@
return p1 != p2.get();
}
-namespace webrtc {
-
-// DEPRECATED: Use scoped_ptr<T[]> instead.
-// TODO(ajm): Remove scoped_array.
-//
-// scoped_array extends scoped_ptr to arrays. Deletion of the array pointed to
-// is guaranteed, either on destruction of the scoped_array or via an explicit
-// reset(). Use shared_array or std::vector if your needs are more complex.
-
-template<typename T>
-class scoped_array {
- private:
-
- T* ptr;
-
- scoped_array(scoped_array const &);
- scoped_array & operator=(scoped_array const &);
-
- public:
-
- typedef T element_type;
-
- explicit scoped_array(T* p = NULL) : ptr(p) {}
-
- ~scoped_array() {
- typedef char type_must_be_complete[sizeof(T)];
- delete[] ptr;
- }
-
- void reset(T* p = NULL) {
- typedef char type_must_be_complete[sizeof(T)];
-
- if (ptr != p) {
- T* arr = ptr;
- ptr = p;
- // Delete last, in case arr destructor indirectly results in ~scoped_array
- delete [] arr;
- }
- }
-
- T& operator[](ptrdiff_t i) const {
- assert(ptr != NULL);
- assert(i >= 0);
- return ptr[i];
- }
-
- T* get() const {
- return ptr;
- }
-
- void swap(scoped_array & b) {
- T* tmp = b.ptr;
- b.ptr = ptr;
- ptr = tmp;
- }
-
- T* release() {
- T* tmp = ptr;
- ptr = NULL;
- return tmp;
- }
-
- T** accept() {
- if (ptr) {
- delete [] ptr;
- ptr = NULL;
- }
- return &ptr;
- }
-};
-
-template<class T> inline
-void swap(scoped_array<T>& a, scoped_array<T>& b) {
- a.swap(b);
-}
-
-} // namespace webrtc
-
#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_PTR_H_
diff --git a/modules/NetEq.target.darwin-x86.mk b/system_wrappers/source/cpu_features_android.target.darwin-arm64.mk
similarity index 62%
copy from modules/NetEq.target.darwin-x86.mk
copy to system_wrappers/source/cpu_features_android.target.darwin-arm64.mk
index 3b05cd9..eb07d4e 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/system_wrappers/source/cpu_features_android.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_system_wrappers_source_cpu_features_android_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,7 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/system_wrappers/source/cpu_features_android.c
# Flags passed to both C and C++ files.
@@ -70,19 +38,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +54,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +85,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +103,6 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,19 +130,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +146,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +177,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +196,6 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +223,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +237,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -318,7 +248,8 @@
LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-LOCAL_STATIC_LIBRARIES :=
+LOCAL_STATIC_LIBRARIES := \
+ cpufeatures
# Enable grouping to fix circular references
LOCAL_GROUP_STATIC_LIBRARIES := true
@@ -329,10 +260,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_system_wrappers_source_cpu_features_android_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: cpu_features_android
+cpu_features_android: third_party_webrtc_system_wrappers_source_cpu_features_android_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/system_wrappers/source/cpu_features_android.target.linux-arm64.mk
similarity index 62%
copy from modules/NetEq.target.darwin-x86.mk
copy to system_wrappers/source/cpu_features_android.target.linux-arm64.mk
index 3b05cd9..eb07d4e 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/system_wrappers/source/cpu_features_android.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_system_wrappers_source_cpu_features_android_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -23,39 +23,7 @@
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/system_wrappers/source/cpu_features_android.c
# Flags passed to both C and C++ files.
@@ -70,19 +38,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +54,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +85,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +103,6 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,19 +130,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +146,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +177,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +196,6 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +223,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +237,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -318,7 +248,8 @@
LOCAL_LDFLAGS := $(LOCAL_LDFLAGS_$(GYP_CONFIGURATION))
-LOCAL_STATIC_LIBRARIES :=
+LOCAL_STATIC_LIBRARIES := \
+ cpufeatures
# Enable grouping to fix circular references
LOCAL_GROUP_STATIC_LIBRARIES := true
@@ -329,10 +260,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_system_wrappers_source_cpu_features_android_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: cpu_features_android
+cpu_features_android: third_party_webrtc_system_wrappers_source_cpu_features_android_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/system_wrappers/source/system_wrappers.target.darwin-arm64.mk
similarity index 63%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to system_wrappers/source/system_wrappers.target.darwin-arm64.mk
index c84869c..a8f5ad8 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/system_wrappers/source/system_wrappers.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_system_wrappers_source_system_wrappers_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,43 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/system_wrappers/source/aligned_malloc.cc \
+ third_party/webrtc/system_wrappers/source/atomic32_posix.cc \
+ third_party/webrtc/system_wrappers/source/clock.cc \
+ third_party/webrtc/system_wrappers/source/condition_variable.cc \
+ third_party/webrtc/system_wrappers/source/condition_variable_posix.cc \
+ third_party/webrtc/system_wrappers/source/cpu_info.cc \
+ third_party/webrtc/system_wrappers/source/cpu_features.cc \
+ third_party/webrtc/system_wrappers/source/critical_section.cc \
+ third_party/webrtc/system_wrappers/source/critical_section_posix.cc \
+ third_party/webrtc/system_wrappers/source/data_log_c.cc \
+ third_party/webrtc/system_wrappers/source/data_log_no_op.cc \
+ third_party/webrtc/system_wrappers/source/event.cc \
+ third_party/webrtc/system_wrappers/source/event_posix.cc \
+ third_party/webrtc/system_wrappers/source/event_tracer.cc \
+ third_party/webrtc/system_wrappers/source/file_impl.cc \
+ third_party/webrtc/system_wrappers/source/logcat_trace_context.cc \
+ third_party/webrtc/system_wrappers/source/logging.cc \
+ third_party/webrtc/system_wrappers/source/rw_lock.cc \
+ third_party/webrtc/system_wrappers/source/rw_lock_generic.cc \
+ third_party/webrtc/system_wrappers/source/rw_lock_posix.cc \
+ third_party/webrtc/system_wrappers/source/sleep.cc \
+ third_party/webrtc/system_wrappers/source/sort.cc \
+ third_party/webrtc/system_wrappers/source/tick_util.cc \
+ third_party/webrtc/system_wrappers/source/thread.cc \
+ third_party/webrtc/system_wrappers/source/thread_posix.cc \
+ third_party/webrtc/system_wrappers/source/trace_impl.cc \
+ third_party/webrtc/system_wrappers/source/trace_posix.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,17 +70,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +112,8 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_THREAD_RR' \
+ '-DWEBRTC_CLOCK_TYPE_REALTIME' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +132,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/source/spreadsortlib \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +151,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -185,17 +166,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +208,8 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_THREAD_RR' \
+ '-DWEBRTC_CLOCK_TYPE_REALTIME' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +229,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/source/spreadsortlib \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +258,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +272,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +294,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_system_wrappers_source_system_wrappers_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: system_wrappers
+system_wrappers: third_party_webrtc_system_wrappers_source_system_wrappers_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86_64.mk b/system_wrappers/source/system_wrappers.target.linux-arm64.mk
similarity index 63%
copy from modules/NetEq.target.darwin-x86_64.mk
copy to system_wrappers/source/system_wrappers.target.linux-arm64.mk
index c84869c..a8f5ad8 100644
--- a/modules/NetEq.target.darwin-x86_64.mk
+++ b/system_wrappers/source/system_wrappers.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_system_wrappers_source_system_wrappers_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,49 +18,43 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/system_wrappers/source/aligned_malloc.cc \
+ third_party/webrtc/system_wrappers/source/atomic32_posix.cc \
+ third_party/webrtc/system_wrappers/source/clock.cc \
+ third_party/webrtc/system_wrappers/source/condition_variable.cc \
+ third_party/webrtc/system_wrappers/source/condition_variable_posix.cc \
+ third_party/webrtc/system_wrappers/source/cpu_info.cc \
+ third_party/webrtc/system_wrappers/source/cpu_features.cc \
+ third_party/webrtc/system_wrappers/source/critical_section.cc \
+ third_party/webrtc/system_wrappers/source/critical_section_posix.cc \
+ third_party/webrtc/system_wrappers/source/data_log_c.cc \
+ third_party/webrtc/system_wrappers/source/data_log_no_op.cc \
+ third_party/webrtc/system_wrappers/source/event.cc \
+ third_party/webrtc/system_wrappers/source/event_posix.cc \
+ third_party/webrtc/system_wrappers/source/event_tracer.cc \
+ third_party/webrtc/system_wrappers/source/file_impl.cc \
+ third_party/webrtc/system_wrappers/source/logcat_trace_context.cc \
+ third_party/webrtc/system_wrappers/source/logging.cc \
+ third_party/webrtc/system_wrappers/source/rw_lock.cc \
+ third_party/webrtc/system_wrappers/source/rw_lock_generic.cc \
+ third_party/webrtc/system_wrappers/source/rw_lock_posix.cc \
+ third_party/webrtc/system_wrappers/source/sleep.cc \
+ third_party/webrtc/system_wrappers/source/sort.cc \
+ third_party/webrtc/system_wrappers/source/tick_util.cc \
+ third_party/webrtc/system_wrappers/source/thread.cc \
+ third_party/webrtc/system_wrappers/source/thread_posix.cc \
+ third_party/webrtc/system_wrappers/source/trace_impl.cc \
+ third_party/webrtc/system_wrappers/source/trace_posix.cc
# Flags passed to both C and C++ files.
MY_CFLAGS_Debug := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -76,17 +70,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -126,8 +112,8 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_THREAD_RR' \
+ '-DWEBRTC_CLOCK_TYPE_REALTIME' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +132,8 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/source/spreadsortlib \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -169,7 +151,6 @@
# Flags passed to both C and C++ files.
MY_CFLAGS_Release := \
- -fstack-protector \
--param=ssp-buffer-size=4 \
-Werror \
-fno-exceptions \
@@ -185,17 +166,9 @@
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -m64 \
- -march=x86-64 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
- -fstack-protector \
-fno-short-enums \
-finline-limit=64 \
-Wa,--noexecstack \
@@ -235,8 +208,8 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
+ '-DWEBRTC_THREAD_RR' \
+ '-DWEBRTC_CLOCK_TYPE_REALTIME' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +229,8 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/source/spreadsortlib \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +258,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +272,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m64 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +294,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_system_wrappers_source_system_wrappers_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: system_wrappers
+system_wrappers: third_party_webrtc_system_wrappers_source_system_wrappers_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/test/direct_transport.h b/test/direct_transport.h
index 9dc9e1f..c40a8c3 100644
--- a/test/direct_transport.h
+++ b/test/direct_transport.h
@@ -49,7 +49,7 @@
scoped_ptr<CriticalSectionWrapper> lock_;
scoped_ptr<EventWrapper> packet_event_;
scoped_ptr<ThreadWrapper> thread_;
- Clock* clock_;
+ Clock* const clock_;
bool shutting_down_;
diff --git a/test/fake_decoder.cc b/test/fake_decoder.cc
index a9e6f50..1624ea0 100644
--- a/test/fake_decoder.cc
+++ b/test/fake_decoder.cc
@@ -36,6 +36,7 @@
const CodecSpecificInfo* codec_specific_info,
int64_t render_time_ms) {
frame_.set_timestamp(input._timeStamp);
+ frame_.set_ntp_time_ms(input.ntp_time_ms_);
frame_.set_render_time_ms(render_time_ms);
callback_->Decoded(frame_);
diff --git a/test/fake_encoder.h b/test/fake_encoder.h
index c0709c1..2a444a1 100644
--- a/test/fake_encoder.h
+++ b/test/fake_encoder.h
@@ -42,7 +42,7 @@
uint32_t framerate) OVERRIDE;
private:
- Clock* clock_;
+ Clock* const clock_;
VideoCodec config_;
EncodedImageCallback* callback_;
int target_bitrate_kbps_;
diff --git a/test/fake_network_pipe_unittest.cc b/test/fake_network_pipe_unittest.cc
index 5076bc0..6655fa1 100644
--- a/test/fake_network_pipe_unittest.cc
+++ b/test/fake_network_pipe_unittest.cc
@@ -47,7 +47,7 @@
}
void SendPackets(FakeNetworkPipe* pipe, int number_packets, int kPacketSize) {
- scoped_array<uint8_t> packet(new uint8_t[kPacketSize]);
+ scoped_ptr<uint8_t[]> packet(new uint8_t[kPacketSize]);
for (int i = 0; i < number_packets; ++i) {
pipe->SendPacket(packet.get(), kPacketSize);
}
diff --git a/test/frame_generator_capturer.cc b/test/frame_generator_capturer.cc
index 6570c6f..d1c17c7 100644
--- a/test/frame_generator_capturer.cc
+++ b/test/frame_generator_capturer.cc
@@ -67,7 +67,8 @@
tick_(EventWrapper::Create()),
lock_(CriticalSectionWrapper::CreateCriticalSection()),
frame_generator_(frame_generator),
- target_fps_(target_fps) {
+ target_fps_(target_fps),
+ first_frame_capture_time_(-1) {
assert(input != NULL);
assert(frame_generator != NULL);
assert(target_fps > 0);
@@ -113,6 +114,9 @@
if (sending_) {
I420VideoFrame* frame = frame_generator_->NextFrame();
frame->set_render_time_ms(clock_->CurrentNtpInMilliseconds());
+ if (first_frame_capture_time_ == -1) {
+ first_frame_capture_time_ = frame->render_time_ms();
+ }
input_->SwapFrame(frame);
}
}
diff --git a/test/frame_generator_capturer.h b/test/frame_generator_capturer.h
index 5be6bb2..ee3f0e0 100644
--- a/test/frame_generator_capturer.h
+++ b/test/frame_generator_capturer.h
@@ -43,6 +43,8 @@
virtual void Start() OVERRIDE;
virtual void Stop() OVERRIDE;
+ int64_t first_frame_capture_time() const { return first_frame_capture_time_; }
+
private:
FrameGeneratorCapturer(Clock* clock,
VideoSendStreamInput* input,
@@ -52,7 +54,7 @@
void InsertFrame();
static bool Run(void* obj);
- Clock* clock_;
+ Clock* const clock_;
bool sending_;
scoped_ptr<EventWrapper> tick_;
@@ -61,6 +63,8 @@
scoped_ptr<FrameGenerator> frame_generator_;
int target_fps_;
+
+ int64_t first_frame_capture_time_;
};
} // test
} // webrtc
diff --git a/test/rtp_rtcp_observer.h b/test/rtp_rtcp_observer.h
index 00422cc..e448653 100644
--- a/test/rtp_rtcp_observer.h
+++ b/test/rtp_rtcp_observer.h
@@ -53,15 +53,15 @@
protected:
RtpRtcpObserver(unsigned int event_timeout_ms,
const FakeNetworkPipe::Config& configuration)
- : lock_(CriticalSectionWrapper::CreateCriticalSection()),
+ : crit_(CriticalSectionWrapper::CreateCriticalSection()),
observation_complete_(EventWrapper::Create()),
parser_(RtpHeaderParser::Create()),
- send_transport_(lock_.get(),
+ send_transport_(crit_.get(),
this,
&RtpRtcpObserver::OnSendRtp,
&RtpRtcpObserver::OnSendRtcp,
configuration),
- receive_transport_(lock_.get(),
+ receive_transport_(crit_.get(),
this,
&RtpRtcpObserver::OnReceiveRtp,
&RtpRtcpObserver::OnReceiveRtcp,
@@ -69,15 +69,15 @@
timeout_ms_(event_timeout_ms) {}
explicit RtpRtcpObserver(unsigned int event_timeout_ms)
- : lock_(CriticalSectionWrapper::CreateCriticalSection()),
+ : crit_(CriticalSectionWrapper::CreateCriticalSection()),
observation_complete_(EventWrapper::Create()),
parser_(RtpHeaderParser::Create()),
- send_transport_(lock_.get(),
+ send_transport_(crit_.get(),
this,
&RtpRtcpObserver::OnSendRtp,
&RtpRtcpObserver::OnSendRtcp,
FakeNetworkPipe::Config()),
- receive_transport_(lock_.get(),
+ receive_transport_(crit_.get(),
this,
&RtpRtcpObserver::OnReceiveRtp,
&RtpRtcpObserver::OnReceiveRtcp,
@@ -89,23 +89,26 @@
DROP_PACKET,
};
- virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
+ virtual Action OnSendRtp(const uint8_t* packet, size_t length)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_) {
return SEND_PACKET;
}
- virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
+ virtual Action OnSendRtcp(const uint8_t* packet, size_t length)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_) {
return SEND_PACKET;
}
- virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) {
+ virtual Action OnReceiveRtp(const uint8_t* packet, size_t length)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_) {
return SEND_PACKET;
}
- virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) {
+ virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_) {
return SEND_PACKET;
}
-
private:
class PacketTransport : public test::DirectTransport {
public:
@@ -118,7 +121,7 @@
PacketTransportAction on_rtcp,
const FakeNetworkPipe::Config& configuration)
: test::DirectTransport(configuration),
- lock_(lock),
+ crit_(lock),
observer_(observer),
on_rtp_(on_rtp),
on_rtcp_(on_rtcp) {}
@@ -128,7 +131,7 @@
EXPECT_FALSE(RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)));
Action action;
{
- CriticalSectionScoped crit_(lock_);
+ CriticalSectionScoped lock(crit_);
action = (observer_->*on_rtp_)(packet, length);
}
switch (action) {
@@ -145,7 +148,7 @@
EXPECT_TRUE(RtpHeaderParser::IsRtcp(packet, static_cast<int>(length)));
Action action;
{
- CriticalSectionScoped crit_(lock_);
+ CriticalSectionScoped lock(crit_);
action = (observer_->*on_rtcp_)(packet, length);
}
switch (action) {
@@ -159,16 +162,16 @@
}
// Pointer to shared lock instance protecting on_rtp_/on_rtcp_ calls.
- CriticalSectionWrapper* lock_;
+ CriticalSectionWrapper* const crit_;
- RtpRtcpObserver* observer_;
- PacketTransportAction on_rtp_, on_rtcp_;
+ RtpRtcpObserver* const observer_;
+ const PacketTransportAction on_rtp_, on_rtcp_;
};
protected:
- scoped_ptr<CriticalSectionWrapper> lock_;
- scoped_ptr<EventWrapper> observation_complete_;
- scoped_ptr<RtpHeaderParser> parser_;
+ const scoped_ptr<CriticalSectionWrapper> crit_;
+ const scoped_ptr<EventWrapper> observation_complete_;
+ const scoped_ptr<RtpHeaderParser> parser_;
private:
PacketTransport send_transport_, receive_transport_;
diff --git a/test/testsupport/fileutils.cc b/test/testsupport/fileutils.cc
index 9d04ab0..a3e6620 100644
--- a/test/testsupport/fileutils.cc
+++ b/test/testsupport/fileutils.cc
@@ -10,8 +10,9 @@
#include "webrtc/test/testsupport/fileutils.h"
-#ifdef WIN32
#include <assert.h>
+
+#ifdef WIN32
#include <direct.h>
#include <tchar.h>
#include <windows.h>
diff --git a/test/testsupport/metrics/video_metrics.cc b/test/testsupport/metrics/video_metrics.cc
index f537e03..1e19806 100644
--- a/test/testsupport/metrics/video_metrics.cc
+++ b/test/testsupport/metrics/video_metrics.cc
@@ -111,8 +111,8 @@
const size_t frame_length = 3 * width * height >> 1;
I420VideoFrame ref_frame;
I420VideoFrame test_frame;
- scoped_array<uint8_t> ref_buffer(new uint8_t[frame_length]);
- scoped_array<uint8_t> test_buffer(new uint8_t[frame_length]);
+ scoped_ptr<uint8_t[]> ref_buffer(new uint8_t[frame_length]);
+ scoped_ptr<uint8_t[]> test_buffer(new uint8_t[frame_length]);
// Set decoded image parameters.
int half_width = (width + 1) / 2;
diff --git a/tools/frame_editing/frame_editing_lib.cc b/tools/frame_editing/frame_editing_lib.cc
index 6e252e8..93a548f 100644
--- a/tools/frame_editing/frame_editing_lib.cc
+++ b/tools/frame_editing/frame_editing_lib.cc
@@ -38,7 +38,7 @@
// Frame size of I420.
int frame_length = CalcBufferSize(kI420, width, height);
- webrtc::scoped_array<uint8_t> temp_buffer(new uint8_t[frame_length]);
+ webrtc::scoped_ptr<uint8_t[]> temp_buffer(new uint8_t[frame_length]);
FILE* out_fid = fopen(out_path.c_str(), "wb");
diff --git a/tools/frame_editing/frame_editing_unittest.cc b/tools/frame_editing/frame_editing_unittest.cc
index 8330218..bdcc2f2 100644
--- a/tools/frame_editing/frame_editing_unittest.cc
+++ b/tools/frame_editing/frame_editing_unittest.cc
@@ -53,8 +53,8 @@
}
// Compares the frames in both streams to the end of one of the streams.
void CompareToTheEnd(FILE* test_video_fid, FILE* ref_video_fid,
- scoped_array<int>* ref_buffer,
- scoped_array<int>* test_buffer) {
+ scoped_ptr<int[]>* ref_buffer,
+ scoped_ptr<int[]>* test_buffer) {
while (!feof(test_video_fid) && !feof(ref_video_fid)) {
num_bytes_read_ = fread(ref_buffer->get(), 1, kFrameSize, ref_video_fid);
if (!feof(ref_video_fid)) {
@@ -78,8 +78,8 @@
FILE* original_fid_;
FILE* edited_fid_;
int num_bytes_read_;
- scoped_array<int> original_buffer_;
- scoped_array<int> edited_buffer_;
+ scoped_ptr<int[]> original_buffer_;
+ scoped_ptr<int[]> edited_buffer_;
int num_frames_read_;
};
diff --git a/video/bitrate_estimator_tests.cc b/video/bitrate_estimator_tests.cc
index 73e66a2..0cb30e3 100644
--- a/video/bitrate_estimator_tests.cc
+++ b/video/bitrate_estimator_tests.cc
@@ -17,6 +17,7 @@
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_decoder.h"
@@ -54,7 +55,6 @@
// Create receiver call first so that we are guaranteed to have a trace
// callback when sender call is created.
Call::Config receiver_call_config(&receive_transport_);
- receiver_call_config.trace_callback = &receiver_trace_;
receiver_call_.reset(Call::Create(receiver_call_config));
Call::Config sender_call_config(&send_transport_);
@@ -112,14 +112,14 @@
}
void PushExpectedLogLine(const std::string& expected_log_line) {
- CriticalSectionScoped cs(crit_sect_.get());
+ CriticalSectionScoped lock(crit_sect_.get());
expected_log_lines_.push_back(expected_log_line);
}
virtual void Print(TraceLevel level,
const char* message,
int length) OVERRIDE {
- CriticalSectionScoped cs(crit_sect_.get());
+ CriticalSectionScoped lock(crit_sect_.get());
if (!(level & kTraceStateInfo)) {
return;
}
@@ -148,9 +148,9 @@
private:
typedef std::list<std::string> Strings;
- scoped_ptr<CriticalSectionWrapper> crit_sect_;
- Strings received_log_lines_;
- Strings expected_log_lines_;
+ const scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ Strings received_log_lines_ GUARDED_BY(crit_sect_);
+ Strings expected_log_lines_ GUARDED_BY(crit_sect_);
scoped_ptr<EventWrapper> done_;
};
@@ -175,7 +175,7 @@
test_->send_config_.encoder_settings.streams[0].height,
30,
Clock::GetRealTimeClock()));
- send_stream_->StartSending();
+ send_stream_->Start();
frame_generator_capturer_->Start();
ExternalVideoDecoder decoder;
@@ -186,7 +186,7 @@
test_->receive_config_.external_decoders.push_back(decoder);
receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
test_->receive_config_);
- receive_stream_->StartReceiving();
+ receive_stream_->Start();
is_sending_receiving_ = true;
}
@@ -202,8 +202,8 @@
void StopSending() {
if (is_sending_receiving_) {
frame_generator_capturer_->Stop();
- send_stream_->StopSending();
- receive_stream_->StopReceiving();
+ send_stream_->Stop();
+ receive_stream_->Stop();
is_sending_receiving_ = false;
}
}
diff --git a/video/call.cc b/video/call.cc
index bcdcbf7..a327654 100644
--- a/video/call.cc
+++ b/video/call.cc
@@ -21,6 +21,7 @@
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include "webrtc/video/video_receive_stream.h"
#include "webrtc/video/video_send_stream.h"
@@ -45,18 +46,18 @@
virtual ~CpuOveruseObserverProxy() {}
virtual void OveruseDetected() OVERRIDE {
- CriticalSectionScoped cs(crit_.get());
+ CriticalSectionScoped lock(crit_.get());
overuse_callback_->OnOveruse();
}
virtual void NormalUsage() OVERRIDE {
- CriticalSectionScoped cs(crit_.get());
+ CriticalSectionScoped lock(crit_.get());
overuse_callback_->OnNormalUse();
}
private:
- scoped_ptr<CriticalSectionWrapper> crit_;
- OveruseCallback* overuse_callback_;
+ const scoped_ptr<CriticalSectionWrapper> crit_;
+ OveruseCallback* overuse_callback_ GUARDED_BY(crit_);
};
class Call : public webrtc::Call, public PacketReceiver {
@@ -95,10 +96,11 @@
Call::Config config_;
- std::map<uint32_t, VideoReceiveStream*> receive_ssrcs_;
+ std::map<uint32_t, VideoReceiveStream*> receive_ssrcs_
+ GUARDED_BY(receive_lock_);
scoped_ptr<RWLockWrapper> receive_lock_;
- std::map<uint32_t, VideoSendStream*> send_ssrcs_;
+ std::map<uint32_t, VideoSendStream*> send_ssrcs_ GUARDED_BY(send_lock_);
scoped_ptr<RWLockWrapper> send_lock_;
scoped_ptr<RtpHeaderParser> rtp_header_parser_;
@@ -115,83 +117,7 @@
};
} // namespace internal
-class TraceDispatcher : public TraceCallback {
- public:
- TraceDispatcher()
- : lock_(CriticalSectionWrapper::CreateCriticalSection()),
- filter_(kTraceNone) {
- Trace::CreateTrace();
- VideoEngine::SetTraceCallback(this);
- VideoEngine::SetTraceFilter(kTraceNone);
- }
-
- ~TraceDispatcher() {
- Trace::ReturnTrace();
- VideoEngine::SetTraceCallback(NULL);
- }
-
- virtual void Print(TraceLevel level,
- const char* message,
- int length) OVERRIDE {
- CriticalSectionScoped crit(lock_.get());
- for (std::map<Call*, Call::Config*>::iterator it = callbacks_.begin();
- it != callbacks_.end();
- ++it) {
- if ((level & it->second->trace_filter) != kTraceNone)
- it->second->trace_callback->Print(level, message, length);
- }
- }
-
- void RegisterCallback(Call* call, Call::Config* config) {
- if (config->trace_callback == NULL)
- return;
-
- CriticalSectionScoped crit(lock_.get());
- callbacks_[call] = config;
-
- filter_ |= config->trace_filter;
- VideoEngine::SetTraceFilter(filter_);
- }
-
- void DeregisterCallback(Call* call) {
- CriticalSectionScoped crit(lock_.get());
- callbacks_.erase(call);
-
- filter_ = kTraceNone;
- for (std::map<Call*, Call::Config*>::iterator it = callbacks_.begin();
- it != callbacks_.end();
- ++it) {
- filter_ |= it->second->trace_filter;
- }
-
- VideoEngine::SetTraceFilter(filter_);
- }
-
- private:
- scoped_ptr<CriticalSectionWrapper> lock_;
- unsigned int filter_;
- std::map<Call*, Call::Config*> callbacks_;
-};
-
-namespace internal {
-TraceDispatcher* global_trace_dispatcher = NULL;
-} // internal
-
-void CreateTraceDispatcher() {
- if (internal::global_trace_dispatcher == NULL) {
- TraceDispatcher* dispatcher = new TraceDispatcher();
- // TODO(pbos): Atomic compare and exchange.
- if (internal::global_trace_dispatcher == NULL) {
- internal::global_trace_dispatcher = dispatcher;
- } else {
- delete dispatcher;
- }
- }
-}
-
Call* Call::Create(const Call::Config& config) {
- CreateTraceDispatcher();
-
VideoEngine* video_engine = config.webrtc_config != NULL
? VideoEngine::Create(*config.webrtc_config)
: VideoEngine::Create();
@@ -217,8 +143,6 @@
new CpuOveruseObserverProxy(config.overuse_callback));
}
- global_trace_dispatcher->RegisterCallback(this, &config_);
-
rtp_rtcp_ = ViERTP_RTCP::GetInterface(video_engine_);
assert(rtp_rtcp_ != NULL);
@@ -235,7 +159,6 @@
}
Call::~Call() {
- global_trace_dispatcher->DeregisterCallback(this);
base_->DeleteChannel(base_channel_id_);
base_->Release();
codec_->Release();
diff --git a/video/call_perf_tests.cc b/video/call_perf_tests.cc
index f7768fb..6b8ff96 100644
--- a/video/call_perf_tests.cc
+++ b/video/call_perf_tests.cc
@@ -22,6 +22,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_audio_device.h"
@@ -67,19 +68,24 @@
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));
- send_stream_->StartSending();
+ send_stream_->Start();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, observer->Wait());
observer->StopSending();
frame_generator_capturer->Stop();
- send_stream_->StopSending();
+ send_stream_->Stop();
call->DestroyVideoSendStream(send_stream_);
}
void TestMinTransmitBitrate(bool pad_to_min_bitrate);
+ void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
+ int threshold_ms,
+ int start_time_ms,
+ int run_time_ms);
+
VideoSendStream* send_stream_;
test::FakeEncoder fake_encoder_;
};
@@ -88,7 +94,7 @@
public:
explicit SyncRtcpObserver(const FakeNetworkPipe::Config& config)
: test::RtpRtcpObserver(kLongTimeoutMs, config),
- critical_section_(CriticalSectionWrapper::CreateCriticalSection()) {}
+ crit_(CriticalSectionWrapper::CreateCriticalSection()) {}
virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
RTCPUtility::RTCPParserV2 parser(packet, length, true);
@@ -110,7 +116,7 @@
}
int64_t RtpTimestampToNtp(uint32_t timestamp) const {
- CriticalSectionScoped cs(critical_section_.get());
+ CriticalSectionScoped lock(crit_.get());
int64_t timestamp_in_ms = -1;
if (ntp_rtp_pairs_.size() == 2) {
// TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the
@@ -124,7 +130,7 @@
private:
void StoreNtpRtpPair(synchronization::RtcpMeasurement ntp_rtp_pair) {
- CriticalSectionScoped cs(critical_section_.get());
+ CriticalSectionScoped lock(crit_.get());
for (synchronization::RtcpList::iterator it = ntp_rtp_pairs_.begin();
it != ntp_rtp_pairs_.end();
++it) {
@@ -142,8 +148,8 @@
ntp_rtp_pairs_.push_front(ntp_rtp_pair);
}
- scoped_ptr<CriticalSectionWrapper> critical_section_;
- synchronization::RtcpList ntp_rtp_pairs_;
+ const scoped_ptr<CriticalSectionWrapper> crit_;
+ synchronization::RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_);
};
class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
@@ -208,7 +214,7 @@
}
private:
- Clock* clock_;
+ Clock* const clock_;
int voe_channel_;
VoEVideoSync* voe_sync_;
SyncRtcpObserver* audio_observer_;
@@ -310,8 +316,8 @@
send_config.encoder_settings.streams[0].height,
30,
Clock::GetRealTimeClock()));
- receive_stream->StartReceiving();
- send_stream->StartSending();
+ receive_stream->Start();
+ send_stream->Start();
capturer->Start();
fake_audio_device.Start();
@@ -328,8 +334,8 @@
fake_audio_device.Stop();
capturer->Stop();
- send_stream->StopSending();
- receive_stream->StopReceiving();
+ send_stream->Stop();
+ receive_stream->Stop();
observer.StopSending();
audio_observer.StopSending();
@@ -343,6 +349,195 @@
VoiceEngine::Delete(voice_engine);
}
+class CaptureNtpTimeObserver : public test::RtpRtcpObserver,
+ public VideoRenderer {
+ public:
+ CaptureNtpTimeObserver(Clock* clock,
+ const FakeNetworkPipe::Config& config,
+ int threshold_ms,
+ int start_time_ms,
+ int run_time_ms)
+ : RtpRtcpObserver(kLongTimeoutMs, config),
+ clock_(clock),
+ threshold_ms_(threshold_ms),
+ start_time_ms_(start_time_ms),
+ run_time_ms_(run_time_ms),
+ creation_time_ms_(clock_->TimeInMilliseconds()),
+ capturer_(NULL),
+ rtp_start_timestamp_set_(false),
+ rtp_start_timestamp_(0) {}
+
+ virtual void RenderFrame(const I420VideoFrame& video_frame,
+ int time_to_render_ms) OVERRIDE {
+ if (video_frame.ntp_time_ms() <= 0) {
+ // Haven't got enough RTCP SR in order to calculate the capture ntp time.
+ return;
+ }
+
+ int64_t now_ms = clock_->TimeInMilliseconds();
+ int64_t time_since_creation = now_ms - creation_time_ms_;
+ if (time_since_creation < start_time_ms_) {
+ // Wait for |start_time_ms_| before start measuring.
+ return;
+ }
+
+ if (time_since_creation > run_time_ms_) {
+ observation_complete_->Set();
+ }
+
+ FrameCaptureTimeList::iterator iter =
+ capture_time_list_.find(video_frame.timestamp());
+ EXPECT_TRUE(iter != capture_time_list_.end());
+
+ // The real capture time has been wrapped to uint32_t before converted
+ // to rtp timestamp in the sender side. So here we convert the estimated
+ // capture time to a uint32_t 90k timestamp also for comparing.
+ uint32_t estimated_capture_timestamp =
+ 90 * static_cast<uint32_t>(video_frame.ntp_time_ms());
+ uint32_t real_capture_timestamp = iter->second;
+ int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp;
+ time_offset_ms = time_offset_ms / 90;
+ std::stringstream ss;
+ ss << time_offset_ms;
+
+ webrtc::test::PrintResult("capture_ntp_time",
+ "",
+ "real - estimated",
+ ss.str(),
+ "ms",
+ true);
+ EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
+ }
+
+ virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
+ RTPHeader header;
+ EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header));
+
+ if (!rtp_start_timestamp_set_) {
+ // Calculate the rtp timestamp offset in order to calculate the real
+ // capture time.
+ uint32_t first_capture_timestamp =
+ 90 * static_cast<uint32_t>(capturer_->first_frame_capture_time());
+ rtp_start_timestamp_ = header.timestamp - first_capture_timestamp;
+ rtp_start_timestamp_set_ = true;
+ }
+
+ uint32_t capture_timestamp = header.timestamp - rtp_start_timestamp_;
+ capture_time_list_.insert(capture_time_list_.end(),
+ std::make_pair(header.timestamp,
+ capture_timestamp));
+ return SEND_PACKET;
+ }
+
+ void SetCapturer(test::FrameGeneratorCapturer* capturer) {
+ capturer_ = capturer;
+ }
+
+ private:
+ Clock* clock_;
+ int threshold_ms_;
+ int start_time_ms_;
+ int run_time_ms_;
+ int64_t creation_time_ms_;
+ test::FrameGeneratorCapturer* capturer_;
+ bool rtp_start_timestamp_set_;
+ uint32_t rtp_start_timestamp_;
+ typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
+ FrameCaptureTimeList capture_time_list_;
+};
+
+void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
+ int threshold_ms,
+ int start_time_ms,
+ int run_time_ms) {
+ CaptureNtpTimeObserver observer(Clock::GetRealTimeClock(),
+ net_config,
+ threshold_ms,
+ start_time_ms,
+ run_time_ms);
+
+ // Sender/receiver call.
+ Call::Config receiver_config(observer.ReceiveTransport());
+ scoped_ptr<Call> receiver_call(Call::Create(receiver_config));
+ scoped_ptr<Call> sender_call(
+ Call::Create(Call::Config(observer.SendTransport())));
+ observer.SetReceivers(receiver_call->Receiver(), sender_call->Receiver());
+
+ // Configure send stream.
+ VideoSendStream::Config send_config = GetSendTestConfig(sender_call.get());
+ VideoSendStream* send_stream =
+ sender_call->CreateVideoSendStream(send_config);
+ scoped_ptr<test::FrameGeneratorCapturer> capturer(
+ test::FrameGeneratorCapturer::Create(
+ send_stream->Input(),
+ send_config.encoder_settings.streams[0].width,
+ send_config.encoder_settings.streams[0].height,
+ 30,
+ Clock::GetRealTimeClock()));
+ observer.SetCapturer(capturer.get());
+
+ // Configure receive stream.
+ VideoReceiveStream::Config receive_config =
+ receiver_call->GetDefaultReceiveConfig();
+ assert(receive_config.codecs.empty());
+ VideoCodec codec =
+ test::CreateDecoderVideoCodec(send_config.encoder_settings);
+ receive_config.codecs.push_back(codec);
+ assert(receive_config.external_decoders.empty());
+ ExternalVideoDecoder decoder;
+ test::FakeDecoder fake_decoder;
+ decoder.decoder = &fake_decoder;
+ decoder.payload_type = send_config.encoder_settings.payload_type;
+ receive_config.external_decoders.push_back(decoder);
+ receive_config.rtp.remote_ssrc = send_config.rtp.ssrcs[0];
+ receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
+ receive_config.renderer = &observer;
+ // Enable the receiver side rtt calculation.
+ receive_config.rtp.rtcp_xr.receiver_reference_time_report = true;
+ VideoReceiveStream* receive_stream =
+ receiver_call->CreateVideoReceiveStream(receive_config);
+
+ // Start the test
+ receive_stream->Start();
+ send_stream->Start();
+ capturer->Start();
+
+ EXPECT_EQ(kEventSignaled, observer.Wait())
+ << "Timed out while waiting for estimated capture ntp time to be "
+ << "within bounds.";
+
+ capturer->Stop();
+ send_stream->Stop();
+ receive_stream->Stop();
+ observer.StopSending();
+
+ sender_call->DestroyVideoSendStream(send_stream);
+ receiver_call->DestroyVideoReceiveStream(receive_stream);
+}
+
+TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
+ FakeNetworkPipe::Config net_config;
+ net_config.queue_delay_ms = 100;
+ // TODO(wu): lower the threshold as the calculation/estimatation becomes more
+ // accurate.
+ const int kThresholdMs = 30;
+ const int kStartTimeMs = 10000;
+ const int kRunTimeMs = 20000;
+ TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
+}
+
+// Flaky, webrtc:3271.
+TEST_F(CallPerfTest, DISABLED_CaptureNtpTimeWithNetworkJitter) {
+ FakeNetworkPipe::Config net_config;
+ net_config.delay_standard_deviation_ms = 10;
+ // TODO(wu): lower the threshold as the calculation/estimatation becomes more
+ // accurate.
+ const int kThresholdMs = 30;
+ const int kStartTimeMs = 10000;
+ const int kRunTimeMs = 20000;
+ TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
+}
+
TEST_F(CallPerfTest, RegisterCpuOveruseObserver) {
// Verifies that either a normal or overuse callback is triggered.
class OveruseCallbackObserver : public test::RtpRtcpObserver,
@@ -477,15 +672,15 @@
30,
Clock::GetRealTimeClock()));
observer.SetSendStream(send_stream);
- receive_stream->StartReceiving();
- send_stream->StartSending();
+ receive_stream->Start();
+ send_stream->Start();
capturer->Start();
EXPECT_EQ(kEventSignaled, observer.Wait())
<< "Timeout while waiting for send-bitrate stats.";
- send_stream->StopSending();
- receive_stream->StopReceiving();
+ send_stream->Stop();
+ receive_stream->Stop();
observer.StopSending();
capturer->Stop();
sender_call->DestroyVideoSendStream(send_stream);
diff --git a/video/call_tests.cc b/video/call_tests.cc
index 44d442d..617b944 100644
--- a/video/call_tests.cc
+++ b/video/call_tests.cc
@@ -106,8 +106,8 @@
}
void StartSending() {
- receive_stream_->StartReceiving();
- send_stream_->StartSending();
+ receive_stream_->Start();
+ send_stream_->Start();
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Start();
}
@@ -116,9 +116,9 @@
if (frame_generator_capturer_.get() != NULL)
frame_generator_capturer_->Stop();
if (send_stream_ != NULL)
- send_stream_->StopSending();
+ send_stream_->Stop();
if (receive_stream_ != NULL)
- receive_stream_->StopReceiving();
+ receive_stream_->Stop();
}
void DestroyStreams() {
@@ -222,63 +222,6 @@
int nacks_left_;
};
-// Test disabled, ongoing work disabled traces causing UsesTraceCallback to
-// fail. Tracked by webrtc:3157.
-TEST_F(CallTest, DISABLED_UsesTraceCallback) {
- const unsigned int kSenderTraceFilter = kTraceDebug;
- const unsigned int kReceiverTraceFilter = kTraceDefault & (~kTraceDebug);
- class TraceObserver : public TraceCallback {
- public:
- explicit TraceObserver(unsigned int filter)
- : filter_(filter), messages_left_(50), done_(EventWrapper::Create()) {}
-
- virtual void Print(TraceLevel level,
- const char* message,
- int length) OVERRIDE {
- EXPECT_EQ(0u, level & (~filter_));
- if (--messages_left_ == 0)
- done_->Set();
- }
-
- EventTypeWrapper Wait() { return done_->Wait(kDefaultTimeoutMs); }
-
- private:
- unsigned int filter_;
- unsigned int messages_left_;
- scoped_ptr<EventWrapper> done_;
- } sender_trace(kSenderTraceFilter), receiver_trace(kReceiverTraceFilter);
-
- test::DirectTransport send_transport, receive_transport;
- Call::Config sender_call_config(&send_transport);
- sender_call_config.trace_callback = &sender_trace;
- sender_call_config.trace_filter = kSenderTraceFilter;
- Call::Config receiver_call_config(&receive_transport);
- receiver_call_config.trace_callback = &receiver_trace;
- receiver_call_config.trace_filter = kReceiverTraceFilter;
- CreateCalls(sender_call_config, receiver_call_config);
- send_transport.SetReceiver(receiver_call_->Receiver());
- receive_transport.SetReceiver(sender_call_->Receiver());
-
- CreateTestConfigs();
-
- CreateStreams();
- CreateFrameGenerator();
- StartSending();
-
- // Wait() waits for a couple of trace callbacks to occur.
- EXPECT_EQ(kEventSignaled, sender_trace.Wait());
- EXPECT_EQ(kEventSignaled, receiver_trace.Wait());
-
- StopSending();
- send_transport.StopSending();
- receive_transport.StopSending();
- DestroyStreams();
-
- // The TraceCallback instance MUST outlive Calls, destroy Calls explicitly.
- sender_call_.reset();
- receiver_call_.reset();
-}
-
TEST_F(CallTest, ReceiverCanBeStartedTwice) {
test::NullTransport transport;
CreateCalls(Call::Config(&transport), Call::Config(&transport));
@@ -286,8 +229,8 @@
CreateTestConfigs();
CreateStreams();
- receive_stream_->StartReceiving();
- receive_stream_->StartReceiving();
+ receive_stream_->Start();
+ receive_stream_->Start();
DestroyStreams();
}
@@ -299,8 +242,8 @@
CreateTestConfigs();
CreateStreams();
- receive_stream_->StopReceiving();
- receive_stream_->StopReceiving();
+ receive_stream_->Stop();
+ receive_stream_->Stop();
DestroyStreams();
}
@@ -492,7 +435,8 @@
DestroyStreams();
}
-TEST_F(CallTest, CanReceiveFec) {
+// TODO(pbos): Flaky, webrtc:3269
+TEST_F(CallTest, DISABLED_CanReceiveFec) {
class FecRenderObserver : public test::RtpRtcpObserver, public VideoRenderer {
public:
FecRenderObserver()
@@ -502,7 +446,8 @@
protected_frame_timestamp_(0) {}
private:
- virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
+ virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE
+ EXCLUSIVE_LOCKS_REQUIRED(crit_) {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header));
@@ -543,7 +488,7 @@
virtual void RenderFrame(const I420VideoFrame& video_frame,
int time_to_render_ms) OVERRIDE {
- CriticalSectionScoped crit_(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
// Rendering frame with timestamp associated with dropped packet -> FEC
// protection worked.
if (state_ == kProtectedPacketDropped &&
@@ -559,8 +504,8 @@
kProtectedPacketDropped,
} state_;
- uint32_t protected_sequence_number_;
- uint32_t protected_frame_timestamp_;
+ uint32_t protected_sequence_number_ GUARDED_BY(crit_);
+ uint32_t protected_frame_timestamp_ GUARDED_BY(crit_);
} observer;
CreateCalls(Call::Config(observer.SendTransport()),
@@ -637,7 +582,7 @@
}
virtual void FrameCallback(I420VideoFrame* frame) OVERRIDE {
- CriticalSectionScoped crit_(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
if (frame->timestamp() == retransmitted_timestamp_) {
EXPECT_TRUE(frame_retransmitted_);
observation_complete_->Set();
@@ -841,7 +786,7 @@
virtual void RenderFrame(const I420VideoFrame& video_frame,
int time_to_render_ms) OVERRIDE {
- CriticalSectionScoped crit_(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
if (received_pli_ && video_frame.timestamp() > highest_dropped_timestamp_) {
observation_complete_->Set();
}
@@ -1116,7 +1061,7 @@
stream->min_bitrate_bps = stream->target_bitrate_bps =
stream->max_bitrate_bps = 100000;
send_streams[i] = sender_call->CreateVideoSendStream(send_config);
- send_streams[i]->StartSending();
+ send_streams[i]->Start();
VideoReceiveStream::Config receive_config =
receiver_call->GetDefaultReceiveConfig();
@@ -1128,7 +1073,7 @@
receive_config.codecs.push_back(codec);
receive_streams[i] =
receiver_call->CreateVideoReceiveStream(receive_config);
- receive_streams[i]->StartReceiving();
+ receive_streams[i]->Start();
frame_generators[i] = test::FrameGeneratorCapturer::Create(
send_streams[i]->Input(), width, height, 30, Clock::GetRealTimeClock());
diff --git a/video/full_stack.cc b/video/full_stack.cc
index 0d3c8ee..51c0a8a 100644
--- a/video/full_stack.cc
+++ b/video/full_stack.cc
@@ -24,6 +24,7 @@
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_encoder.h"
@@ -81,18 +82,18 @@
transport_(transport),
receiver_(NULL),
test_label_(test_label),
+ frames_left_(duration_frames),
dropped_frames_(0),
- rtp_timestamp_delta_(0),
- first_send_frame_(NULL),
last_render_time_(0),
+ rtp_timestamp_delta_(0),
+ crit_(CriticalSectionWrapper::CreateCriticalSection()),
+ first_send_frame_(NULL),
avg_psnr_threshold_(avg_psnr_threshold),
avg_ssim_threshold_(avg_ssim_threshold),
- frames_left_(duration_frames),
- crit_(CriticalSectionWrapper::CreateCriticalSection()),
comparison_lock_(CriticalSectionWrapper::CreateCriticalSection()),
comparison_thread_(ThreadWrapper::CreateThread(&FrameComparisonThread,
this)),
- trigger_(EventWrapper::Create()) {
+ done_(EventWrapper::Create()) {
unsigned int id;
EXPECT_TRUE(comparison_thread_->Start(id));
}
@@ -117,7 +118,7 @@
RTPHeader header;
parser->Parse(packet, static_cast<int>(length), &header);
{
- CriticalSectionScoped cs(crit_.get());
+ CriticalSectionScoped lock(crit_.get());
recv_times_[header.timestamp - rtp_timestamp_delta_] =
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
}
@@ -132,7 +133,7 @@
virtual void SwapFrame(I420VideoFrame* video_frame) OVERRIDE {
I420VideoFrame* copy = NULL;
{
- CriticalSectionScoped cs(crit_.get());
+ CriticalSectionScoped lock(crit_.get());
if (frame_pool_.size() > 0) {
copy = frame_pool_.front();
frame_pool_.pop_front();
@@ -145,7 +146,7 @@
copy->set_timestamp(copy->render_time_ms() * 90);
{
- CriticalSectionScoped cs(crit_.get());
+ CriticalSectionScoped lock(crit_.get());
if (first_send_frame_ == NULL && rtp_timestamp_delta_ == 0)
first_send_frame_ = copy;
@@ -161,7 +162,7 @@
parser->Parse(packet, static_cast<int>(length), &header);
{
- CriticalSectionScoped cs(crit_.get());
+ CriticalSectionScoped lock(crit_.get());
if (rtp_timestamp_delta_ == 0) {
rtp_timestamp_delta_ =
header.timestamp - first_send_frame_->timestamp();
@@ -184,29 +185,27 @@
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
uint32_t send_timestamp = video_frame.timestamp() - rtp_timestamp_delta_;
- {
- CriticalSectionScoped cs(crit_.get());
- while (frames_.front()->timestamp() < send_timestamp) {
- AddFrameComparison(
- frames_.front(), &last_rendered_frame_, true, render_time_ms);
- frame_pool_.push_back(frames_.front());
- frames_.pop_front();
- }
-
- I420VideoFrame* reference_frame = frames_.front();
+ CriticalSectionScoped lock(crit_.get());
+ while (frames_.front()->timestamp() < send_timestamp) {
+ AddFrameComparison(
+ frames_.front(), &last_rendered_frame_, true, render_time_ms);
+ frame_pool_.push_back(frames_.front());
frames_.pop_front();
- assert(reference_frame != NULL);
- EXPECT_EQ(reference_frame->timestamp(), send_timestamp);
- assert(reference_frame->timestamp() == send_timestamp);
-
- AddFrameComparison(reference_frame, &video_frame, false, render_time_ms);
- frame_pool_.push_back(reference_frame);
}
+ I420VideoFrame* reference_frame = frames_.front();
+ frames_.pop_front();
+ assert(reference_frame != NULL);
+ EXPECT_EQ(reference_frame->timestamp(), send_timestamp);
+ assert(reference_frame->timestamp() == send_timestamp);
+
+ AddFrameComparison(reference_frame, &video_frame, false, render_time_ms);
+ frame_pool_.push_back(reference_frame);
+
last_rendered_frame_.CopyFrame(video_frame);
}
- void Wait() { trigger_->Wait(120 * 1000); }
+ void Wait() { done_->Wait(120 * 1000); }
VideoSendStreamInput* input_;
Transport* transport_;
@@ -250,7 +249,8 @@
void AddFrameComparison(const I420VideoFrame* reference,
const I420VideoFrame* render,
bool dropped,
- int64_t render_time_ms) {
+ int64_t render_time_ms)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_) {
int64_t send_time_ms = send_times_[reference->timestamp()];
send_times_.erase(reference->timestamp());
int64_t recv_time_ms = recv_times_[reference->timestamp()];
@@ -313,7 +313,7 @@
PrintResult("time_between_rendered_frames", rendered_delta_, " ms");
EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_);
EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_);
- trigger_->Set();
+ done_->Set();
return false;
}
@@ -353,31 +353,32 @@
unit);
}
- const char* test_label_;
+ const char* const test_label_;
test::Statistics sender_time_;
test::Statistics receiver_time_;
test::Statistics psnr_;
test::Statistics ssim_;
test::Statistics end_to_end_;
test::Statistics rendered_delta_;
-
- int dropped_frames_;
- std::deque<I420VideoFrame*> frames_;
- std::deque<I420VideoFrame*> frame_pool_;
- I420VideoFrame last_rendered_frame_;
- std::map<uint32_t, int64_t> send_times_;
- std::map<uint32_t, int64_t> recv_times_;
- uint32_t rtp_timestamp_delta_;
- I420VideoFrame* first_send_frame_;
- int64_t last_render_time_;
- double avg_psnr_threshold_;
- double avg_ssim_threshold_;
int frames_left_;
- scoped_ptr<CriticalSectionWrapper> crit_;
- scoped_ptr<CriticalSectionWrapper> comparison_lock_;
- scoped_ptr<ThreadWrapper> comparison_thread_;
- std::deque<FrameComparison> comparisons_;
- scoped_ptr<EventWrapper> trigger_;
+ int dropped_frames_;
+ int64_t last_render_time_;
+ uint32_t rtp_timestamp_delta_;
+
+ const scoped_ptr<CriticalSectionWrapper> crit_;
+ std::deque<I420VideoFrame*> frames_ GUARDED_BY(crit_);
+ std::deque<I420VideoFrame*> frame_pool_ GUARDED_BY(crit_);
+ I420VideoFrame last_rendered_frame_ GUARDED_BY(crit_);
+ std::map<uint32_t, int64_t> send_times_ GUARDED_BY(crit_);
+ std::map<uint32_t, int64_t> recv_times_ GUARDED_BY(crit_);
+ I420VideoFrame* first_send_frame_ GUARDED_BY(crit_);
+ double avg_psnr_threshold_ GUARDED_BY(crit_);
+ double avg_ssim_threshold_ GUARDED_BY(crit_);
+
+ const scoped_ptr<CriticalSectionWrapper> comparison_lock_;
+ const scoped_ptr<ThreadWrapper> comparison_thread_;
+ std::deque<FrameComparison> comparisons_ GUARDED_BY(comparison_lock_);
+ const scoped_ptr<EventWrapper> done_;
};
TEST_P(FullStackTest, NoPacketLoss) {
@@ -437,16 +438,15 @@
VideoReceiveStream* receive_stream =
call->CreateVideoReceiveStream(receive_config);
- receive_stream->StartReceiving();
- send_stream->StartSending();
-
+ receive_stream->Start();
+ send_stream->Start();
file_capturer->Start();
analyzer.Wait();
file_capturer->Stop();
- send_stream->StopSending();
- receive_stream->StopReceiving();
+ send_stream->Stop();
+ receive_stream->Stop();
call->DestroyVideoReceiveStream(receive_stream);
call->DestroyVideoSendStream(send_stream);
diff --git a/video/loopback.cc b/video/loopback.cc
index 0131aa9..c99183a 100644
--- a/video/loopback.cc
+++ b/video/loopback.cc
@@ -91,15 +91,15 @@
VideoReceiveStream* receive_stream =
call->CreateVideoReceiveStream(receive_config);
- receive_stream->StartReceiving();
- send_stream->StartSending();
+ receive_stream->Start();
+ send_stream->Start();
camera->Start();
test::PressEnterToContinue();
camera->Stop();
- send_stream->StopSending();
- receive_stream->StopReceiving();
+ send_stream->Stop();
+ receive_stream->Stop();
call->DestroyVideoReceiveStream(receive_stream);
call->DestroyVideoSendStream(send_stream);
diff --git a/video/rampup_tests.cc b/video/rampup_tests.cc
index ca6fb10..32a2998 100644
--- a/video/rampup_tests.cc
+++ b/video/rampup_tests.cc
@@ -48,14 +48,14 @@
StreamObserver(const SsrcMap& rtx_media_ssrcs,
newapi::Transport* feedback_transport,
Clock* clock)
- : critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
+ : clock_(clock),
test_done_(EventWrapper::Create()),
rtp_parser_(RtpHeaderParser::Create()),
feedback_transport_(feedback_transport),
receive_stats_(ReceiveStatistics::Create(clock)),
payload_registry_(
new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))),
- clock_(clock),
+ crit_(CriticalSectionWrapper::CreateCriticalSection()),
expected_bitrate_bps_(0),
rtx_media_ssrcs_(rtx_media_ssrcs),
total_sent_(0),
@@ -85,12 +85,13 @@
}
void set_expected_bitrate_bps(unsigned int expected_bitrate_bps) {
+ CriticalSectionScoped lock(crit_.get());
expected_bitrate_bps_ = expected_bitrate_bps;
}
virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
unsigned int bitrate) OVERRIDE {
- CriticalSectionScoped lock(critical_section_.get());
+ CriticalSectionScoped lock(crit_.get());
assert(expected_bitrate_bps_ > 0);
if (bitrate >= expected_bitrate_bps_) {
// Just trigger if there was any rtx padding packet.
@@ -104,7 +105,7 @@
}
virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE {
- CriticalSectionScoped lock(critical_section_.get());
+ CriticalSectionScoped lock(crit_.get());
RTPHeader header;
EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
receive_stats_->IncomingPacket(header, length, false);
@@ -156,7 +157,7 @@
value, units, false);
}
- void TriggerTestDone() {
+ void TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_) {
ReportResult("total-sent", total_sent_, "bytes");
ReportResult("padding-sent", padding_sent_, "bytes");
ReportResult("rtx-media-sent", rtx_media_sent_, "bytes");
@@ -166,23 +167,24 @@
test_done_->Set();
}
- scoped_ptr<CriticalSectionWrapper> critical_section_;
- scoped_ptr<EventWrapper> test_done_;
- scoped_ptr<RtpHeaderParser> rtp_parser_;
+ Clock* const clock_;
+ const scoped_ptr<EventWrapper> test_done_;
+ const scoped_ptr<RtpHeaderParser> rtp_parser_;
scoped_ptr<RtpRtcp> rtp_rtcp_;
internal::TransportAdapter feedback_transport_;
- scoped_ptr<ReceiveStatistics> receive_stats_;
- scoped_ptr<RTPPayloadRegistry> payload_registry_;
+ const scoped_ptr<ReceiveStatistics> receive_stats_;
+ const scoped_ptr<RTPPayloadRegistry> payload_registry_;
scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
- Clock* clock_;
- unsigned int expected_bitrate_bps_;
- SsrcMap rtx_media_ssrcs_;
- size_t total_sent_;
- size_t padding_sent_;
- size_t rtx_media_sent_;
- int total_packets_sent_;
- int padding_packets_sent_;
- int rtx_media_packets_sent_;
+
+ const scoped_ptr<CriticalSectionWrapper> crit_;
+ unsigned int expected_bitrate_bps_ GUARDED_BY(crit_);
+ SsrcMap rtx_media_ssrcs_ GUARDED_BY(crit_);
+ size_t total_sent_ GUARDED_BY(crit_);
+ size_t padding_sent_ GUARDED_BY(crit_);
+ size_t rtx_media_sent_ GUARDED_BY(crit_);
+ int total_packets_sent_ GUARDED_BY(crit_);
+ int padding_packets_sent_ GUARDED_BY(crit_);
+ int rtx_media_packets_sent_ GUARDED_BY(crit_);
};
class LowRateStreamObserver : public test::DirectTransport,
@@ -193,21 +195,21 @@
Clock* clock,
size_t number_of_streams,
bool rtx_used)
- : critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
+ : clock_(clock),
+ number_of_streams_(number_of_streams),
+ rtx_used_(rtx_used),
test_done_(EventWrapper::Create()),
rtp_parser_(RtpHeaderParser::Create()),
feedback_transport_(feedback_transport),
receive_stats_(ReceiveStatistics::Create(clock)),
- clock_(clock),
+ crit_(CriticalSectionWrapper::CreateCriticalSection()),
+ send_stream_(NULL),
test_state_(kFirstRampup),
state_start_ms_(clock_->TimeInMilliseconds()),
interval_start_ms_(state_start_ms_),
last_remb_bps_(0),
sent_bytes_(0),
total_overuse_bytes_(0),
- number_of_streams_(number_of_streams),
- rtx_used_(rtx_used),
- send_stream_(NULL),
suspended_in_stats_(false) {
RtpRtcp::Configuration config;
config.receive_statistics = receive_stats_.get();
@@ -231,12 +233,13 @@
}
virtual void SetSendStream(const VideoSendStream* send_stream) {
+ CriticalSectionScoped lock(crit_.get());
send_stream_ = send_stream;
}
virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs,
unsigned int bitrate) {
- CriticalSectionScoped lock(critical_section_.get());
+ CriticalSectionScoped lock(crit_.get());
rtp_rtcp_->SetREMBData(
bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]);
rtp_rtcp_->Process();
@@ -244,6 +247,7 @@
}
virtual bool SendRtp(const uint8_t* data, size_t length) OVERRIDE {
+ CriticalSectionScoped lock(crit_.get());
sent_bytes_ += length;
int64_t now_ms = clock_->TimeInMilliseconds();
if (now_ms > interval_start_ms_ + 1000) { // Let at least 1 second pass.
@@ -265,7 +269,7 @@
}
virtual bool DeliverPacket(const uint8_t* packet, size_t length) OVERRIDE {
- CriticalSectionScoped lock(critical_section_.get());
+ CriticalSectionScoped lock(crit_.get());
RTPHeader header;
EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header));
receive_stats_->IncomingPacket(header, length, false);
@@ -300,8 +304,8 @@
// This method defines the state machine for the ramp up-down-up test.
void EvolveTestState(unsigned int bitrate_bps) {
int64_t now = clock_->TimeInMilliseconds();
+ CriticalSectionScoped lock(crit_.get());
assert(send_stream_ != NULL);
- CriticalSectionScoped lock(critical_section_.get());
switch (test_state_) {
case kFirstRampup: {
EXPECT_FALSE(suspended_in_stats_);
@@ -374,25 +378,26 @@
static const unsigned int kExpectedLowBitrateBps = 20000;
enum TestStates { kFirstRampup, kLowRate, kSecondRampup };
- scoped_ptr<CriticalSectionWrapper> critical_section_;
- scoped_ptr<EventWrapper> test_done_;
- scoped_ptr<RtpHeaderParser> rtp_parser_;
- scoped_ptr<RtpRtcp> rtp_rtcp_;
- internal::TransportAdapter feedback_transport_;
- scoped_ptr<ReceiveStatistics> receive_stats_;
- scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
- Clock* clock_;
- FakeNetworkPipe::Config forward_transport_config_;
- TestStates test_state_;
- int64_t state_start_ms_;
- int64_t interval_start_ms_;
- unsigned int last_remb_bps_;
- size_t sent_bytes_;
- size_t total_overuse_bytes_;
+ Clock* const clock_;
const size_t number_of_streams_;
const bool rtx_used_;
- const VideoSendStream* send_stream_;
- bool suspended_in_stats_ GUARDED_BY(critical_section_);
+ const scoped_ptr<EventWrapper> test_done_;
+ const scoped_ptr<RtpHeaderParser> rtp_parser_;
+ scoped_ptr<RtpRtcp> rtp_rtcp_;
+ internal::TransportAdapter feedback_transport_;
+ const scoped_ptr<ReceiveStatistics> receive_stats_;
+ scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_;
+
+ scoped_ptr<CriticalSectionWrapper> crit_;
+ const VideoSendStream* send_stream_ GUARDED_BY(crit_);
+ FakeNetworkPipe::Config forward_transport_config_ GUARDED_BY(crit_);
+ TestStates test_state_ GUARDED_BY(crit_);
+ int64_t state_start_ms_ GUARDED_BY(crit_);
+ int64_t interval_start_ms_ GUARDED_BY(crit_);
+ unsigned int last_remb_bps_ GUARDED_BY(crit_);
+ size_t sent_bytes_ GUARDED_BY(crit_);
+ size_t total_overuse_bytes_ GUARDED_BY(crit_);
+ bool suspended_in_stats_ GUARDED_BY(crit_);
};
}
@@ -469,13 +474,13 @@
send_config.encoder_settings.streams.back().max_framerate,
Clock::GetRealTimeClock()));
- send_stream->StartSending();
+ send_stream->Start();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, stream_observer.Wait());
frame_generator_capturer->Stop();
- send_stream->StopSending();
+ send_stream->Stop();
call->DestroyVideoSendStream(send_stream);
}
@@ -528,7 +533,7 @@
30,
Clock::GetRealTimeClock()));
- send_stream->StartSending();
+ send_stream->Start();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, stream_observer.Wait());
@@ -536,7 +541,7 @@
stream_observer.StopSending();
receiver_transport.StopSending();
frame_generator_capturer->Stop();
- send_stream->StopSending();
+ send_stream->Stop();
call->DestroyVideoSendStream(send_stream);
}
diff --git a/video/receive_statistics_proxy.cc b/video/receive_statistics_proxy.cc
index f42e4d3..6004281 100644
--- a/video/receive_statistics_proxy.cc
+++ b/video/receive_statistics_proxy.cc
@@ -22,13 +22,13 @@
ViECodec* codec,
int channel)
: channel_(channel),
- lock_(CriticalSectionWrapper::CreateCriticalSection()),
clock_(clock),
+ codec_(codec),
+ rtp_rtcp_(rtp_rtcp),
+ crit_(CriticalSectionWrapper::CreateCriticalSection()),
// 1000ms window, scale 1000 for ms to s.
decode_fps_estimator_(1000, 1000),
- renders_fps_estimator_(1000, 1000),
- codec_(codec),
- rtp_rtcp_(rtp_rtcp) {
+ renders_fps_estimator_(1000, 1000) {
stats_.ssrc = ssrc;
}
@@ -37,7 +37,7 @@
VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
VideoReceiveStream::Stats stats;
{
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
stats = stats_;
}
stats.c_name = GetCName();
@@ -59,7 +59,7 @@
void ReceiveStatisticsProxy::IncomingRate(const int video_channel,
const unsigned int framerate,
const unsigned int bitrate) {
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
stats_.network_frame_rate = framerate;
stats_.bitrate_bps = bitrate;
}
@@ -67,7 +67,7 @@
void ReceiveStatisticsProxy::StatisticsUpdated(
const webrtc::RtcpStatistics& statistics,
uint32_t ssrc) {
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
stats_.rtcp_stats = statistics;
}
@@ -75,7 +75,7 @@
void ReceiveStatisticsProxy::DataCountersUpdated(
const webrtc::StreamDataCounters& counters,
uint32_t ssrc) {
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
stats_.rtp_stats = counters;
}
@@ -83,7 +83,7 @@
void ReceiveStatisticsProxy::OnDecodedFrame() {
uint64_t now = clock_->TimeInMilliseconds();
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
decode_fps_estimator_.Update(1, now);
stats_.decode_frame_rate = decode_fps_estimator_.Rate(now);
}
@@ -91,7 +91,7 @@
void ReceiveStatisticsProxy::OnRenderedFrame() {
uint64_t now = clock_->TimeInMilliseconds();
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
renders_fps_estimator_.Update(1, now);
stats_.render_frame_rate = renders_fps_estimator_.Rate(now);
}
diff --git a/video/receive_statistics_proxy.h b/video/receive_statistics_proxy.h
index bedebb3..35e5cc3 100644
--- a/video/receive_statistics_proxy.h
+++ b/video/receive_statistics_proxy.h
@@ -16,6 +16,7 @@
#include "webrtc/common_types.h"
#include "webrtc/frame_callback.h"
#include "webrtc/modules/remote_bitrate_estimator/rate_statistics.h"
+#include "webrtc/system_wrappers/interface/thread_annotations.h"
#include "webrtc/video_engine/include/vie_codec.h"
#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
#include "webrtc/video_receive_stream.h"
@@ -73,13 +74,14 @@
std::string GetCName() const;
const int channel_;
- scoped_ptr<CriticalSectionWrapper> lock_;
- Clock* clock_;
- VideoReceiveStream::Stats stats_;
- RateStatistics decode_fps_estimator_;
- RateStatistics renders_fps_estimator_;
- ViECodec* codec_;
- ViERTP_RTCP* rtp_rtcp_;
+ Clock* const clock_;
+ ViECodec* const codec_;
+ ViERTP_RTCP* const rtp_rtcp_;
+
+ scoped_ptr<CriticalSectionWrapper> crit_;
+ VideoReceiveStream::Stats stats_ GUARDED_BY(crit_);
+ RateStatistics decode_fps_estimator_ GUARDED_BY(crit_);
+ RateStatistics renders_fps_estimator_ GUARDED_BY(crit_);
};
} // namespace internal
diff --git a/video/send_statistics_proxy.cc b/video/send_statistics_proxy.cc
index 1cd4e26..c9bd05c 100644
--- a/video/send_statistics_proxy.cc
+++ b/video/send_statistics_proxy.cc
@@ -20,33 +20,34 @@
const VideoSendStream::Config& config,
SendStatisticsProxy::StatsProvider* stats_provider)
: config_(config),
- lock_(CriticalSectionWrapper::CreateCriticalSection()),
- stats_provider_(stats_provider) {}
+ stats_provider_(stats_provider),
+ crit_(CriticalSectionWrapper::CreateCriticalSection()) {
+}
SendStatisticsProxy::~SendStatisticsProxy() {}
void SendStatisticsProxy::OutgoingRate(const int video_channel,
const unsigned int framerate,
const unsigned int bitrate) {
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
stats_.encode_frame_rate = framerate;
}
void SendStatisticsProxy::SuspendChange(int video_channel, bool is_suspended) {
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
stats_.suspended = is_suspended;
}
void SendStatisticsProxy::CapturedFrameRate(const int capture_id,
const unsigned char frame_rate) {
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
stats_.input_frame_rate = frame_rate;
}
VideoSendStream::Stats SendStatisticsProxy::GetStats() const {
VideoSendStream::Stats stats;
{
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
stats = stats_;
}
stats_provider_->GetSendSideDelay(&stats);
@@ -68,7 +69,7 @@
void SendStatisticsProxy::StatisticsUpdated(const RtcpStatistics& statistics,
uint32_t ssrc) {
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
@@ -79,7 +80,7 @@
void SendStatisticsProxy::DataCountersUpdated(
const StreamDataCounters& counters,
uint32_t ssrc) {
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
@@ -89,7 +90,7 @@
void SendStatisticsProxy::Notify(const BitrateStatistics& bitrate,
uint32_t ssrc) {
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
@@ -100,7 +101,7 @@
void SendStatisticsProxy::FrameCountUpdated(FrameType frame_type,
uint32_t frame_count,
const unsigned int ssrc) {
- CriticalSectionScoped cs(lock_.get());
+ CriticalSectionScoped lock(crit_.get());
StreamStats* stats = GetStatsEntry(ssrc);
if (stats == NULL)
return;
diff --git a/video/send_statistics_proxy.h b/video/send_statistics_proxy.h
index 2f45ff7..a1ff14c 100644
--- a/video/send_statistics_proxy.h
+++ b/video/send_statistics_proxy.h
@@ -81,12 +81,12 @@
const CaptureAlarm alarm) OVERRIDE {}
private:
- StreamStats* GetStatsEntry(uint32_t ssrc) EXCLUSIVE_LOCKS_REQUIRED(lock_);
+ StreamStats* GetStatsEntry(uint32_t ssrc) EXCLUSIVE_LOCKS_REQUIRED(crit_);
const VideoSendStream::Config config_;
- scoped_ptr<CriticalSectionWrapper> lock_;
- VideoSendStream::Stats stats_ GUARDED_BY(lock_);
StatsProvider* const stats_provider_;
+ scoped_ptr<CriticalSectionWrapper> crit_;
+ VideoSendStream::Stats stats_ GUARDED_BY(crit_);
};
} // namespace webrtc
diff --git a/video/video_receive_stream.cc b/video/video_receive_stream.cc
index d3eedc3..012ca68 100644
--- a/video/video_receive_stream.cc
+++ b/video/video_receive_stream.cc
@@ -212,7 +212,7 @@
rtp_rtcp_->Release();
}
-void VideoReceiveStream::StartReceiving() {
+void VideoReceiveStream::Start() {
transport_adapter_.Enable();
if (render_->StartRender(channel_) != 0)
abort();
@@ -220,7 +220,7 @@
abort();
}
-void VideoReceiveStream::StopReceiving() {
+void VideoReceiveStream::Stop() {
if (render_->StopRender(channel_) != 0)
abort();
if (video_engine_base_->StopReceive(channel_) != 0)
diff --git a/video/video_receive_stream.h b/video/video_receive_stream.h
index 14653de..4ff086a 100644
--- a/video/video_receive_stream.h
+++ b/video/video_receive_stream.h
@@ -49,8 +49,8 @@
int base_channel);
virtual ~VideoReceiveStream();
- virtual void StartReceiving() OVERRIDE;
- virtual void StopReceiving() OVERRIDE;
+ virtual void Start() OVERRIDE;
+ virtual void Stop() OVERRIDE;
virtual Stats GetStats() const OVERRIDE;
virtual void GetCurrentReceiveCodec(VideoCodec* receive_codec) OVERRIDE;
@@ -70,7 +70,7 @@
TransportAdapter transport_adapter_;
EncodedFrameCallbackAdapter encoded_frame_proxy_;
VideoReceiveStream::Config config_;
- Clock* clock_;
+ Clock* const clock_;
ViEBase* video_engine_base_;
ViECodec* codec_;
diff --git a/video/video_send_stream.cc b/video/video_send_stream.cc
index a7eebef..e6e683a 100644
--- a/video/video_send_stream.cc
+++ b/video/video_send_stream.cc
@@ -37,7 +37,8 @@
codec_lock_(CriticalSectionWrapper::CreateCriticalSection()),
config_(config),
external_codec_(NULL),
- channel_(-1) {
+ channel_(-1),
+ stats_proxy_(new SendStatisticsProxy(config, this)) {
video_engine_base_ = ViEBase::GetInterface(video_engine);
video_engine_base_->CreateChannel(channel_, base_channel);
assert(channel_ != -1);
@@ -142,8 +143,6 @@
codec_->SuspendBelowMinBitrate(channel_);
}
- stats_proxy_.reset(new SendStatisticsProxy(config, this));
-
rtp_rtcp_->RegisterSendChannelRtcpStatisticsCallback(channel_,
stats_proxy_.get());
rtp_rtcp_->RegisterSendChannelRtpStatisticsCallback(channel_,
@@ -210,13 +209,13 @@
VideoSendStreamInput* VideoSendStream::Input() { return this; }
-void VideoSendStream::StartSending() {
+void VideoSendStream::Start() {
transport_adapter_.Enable();
video_engine_base_->StartSend(channel_);
video_engine_base_->StartReceive(channel_);
}
-void VideoSendStream::StopSending() {
+void VideoSendStream::Stop() {
video_engine_base_->StopSend(channel_);
video_engine_base_->StopReceive(channel_);
transport_adapter_.Disable();
diff --git a/video/video_send_stream.h b/video/video_send_stream.h
index 15cf642..b8f5661 100644
--- a/video/video_send_stream.h
+++ b/video/video_send_stream.h
@@ -46,9 +46,8 @@
virtual ~VideoSendStream();
- virtual void StartSending() OVERRIDE;
-
- virtual void StopSending() OVERRIDE;
+ virtual void Start() OVERRIDE;
+ virtual void Stop() OVERRIDE;
virtual bool ReconfigureVideoEncoder(const std::vector<VideoStream>& streams,
void* encoder_settings) OVERRIDE;
@@ -88,7 +87,7 @@
int channel_;
int capture_id_;
- scoped_ptr<SendStatisticsProxy> stats_proxy_;
+ const scoped_ptr<SendStatisticsProxy> stats_proxy_;
};
} // namespace internal
} // namespace webrtc
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 656bfab..13ac01c 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -51,14 +51,14 @@
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));
- send_stream_->StartSending();
+ send_stream_->Start();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, observer->Wait());
observer->StopSending();
frame_generator_capturer->Stop();
- send_stream_->StopSending();
+ send_stream_->Stop();
call->DestroyVideoSendStream(send_stream_);
}
@@ -179,7 +179,7 @@
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));
- send_stream_->StartSending();
+ send_stream_->Start();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, observer.Wait())
@@ -195,7 +195,7 @@
observer.StopSending();
frame_generator_capturer->Stop();
- send_stream_->StopSending();
+ send_stream_->Stop();
call->DestroyVideoSendStream(send_stream_);
}
@@ -206,8 +206,8 @@
VideoSendStream::Config config = GetSendTestConfig(call.get(), 1);
VideoSendStream* stream = call->CreateVideoSendStream(config);
- stream->StartSending();
- stream->StartSending();
+ stream->Start();
+ stream->Start();
call->DestroyVideoSendStream(stream);
}
@@ -218,8 +218,8 @@
VideoSendStream::Config config = GetSendTestConfig(call.get(), 1);
VideoSendStream* stream = call->CreateVideoSendStream(config);
- stream->StopSending();
- stream->StopSending();
+ stream->Stop();
+ stream->Stop();
call->DestroyVideoSendStream(stream);
}
@@ -415,7 +415,7 @@
VideoSendStream::Config send_config = GetSendTestConfig(call.get(), 1);
VideoSendStream* video_send_stream = call->CreateVideoSendStream(send_config);
- video_send_stream->StartSending();
+ video_send_stream->Start();
I420VideoFrame frame;
frame.CreateEmptyFrame(
@@ -820,14 +820,14 @@
: RtpRtcpObserver(30 * 1000), // Timeout after 30 seconds.
transport_adapter_(&transport_),
clock_(Clock::GetRealTimeClock()),
+ send_stream_ptr_(send_stream_ptr),
+ crit_(CriticalSectionWrapper::CreateCriticalSection()),
test_state_(kBeforeSuspend),
rtp_count_(0),
last_sequence_number_(0),
suspended_frame_count_(0),
low_remb_bps_(0),
- high_remb_bps_(0),
- crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
- send_stream_ptr_(send_stream_ptr) {
+ high_remb_bps_(0) {
transport_adapter_.Enable();
}
@@ -838,13 +838,13 @@
virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
// Receive statistics reporting having lost 0% of the packets.
// This is needed for the send-side bitrate controller to work properly.
- CriticalSectionScoped lock(crit_sect_.get());
+ CriticalSectionScoped lock(crit_.get());
SendRtcpFeedback(0); // REMB is only sent if value is > 0.
return SEND_PACKET;
}
virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
- CriticalSectionScoped lock(crit_sect_.get());
+ CriticalSectionScoped lock(crit_.get());
++rtp_count_;
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, static_cast<int>(length), &header));
@@ -880,7 +880,7 @@
// This method implements the I420FrameCallback.
void FrameCallback(I420VideoFrame* video_frame) OVERRIDE {
- CriticalSectionScoped lock(crit_sect_.get());
+ CriticalSectionScoped lock(crit_.get());
if (test_state_ == kDuringSuspend &&
++suspended_frame_count_ > kSuspendTimeFrames) {
assert(*send_stream_ptr_);
@@ -891,9 +891,15 @@
}
}
- void set_low_remb_bps(int value) { low_remb_bps_ = value; }
+ void set_low_remb_bps(int value) {
+ CriticalSectionScoped lock(crit_.get());
+ low_remb_bps_ = value;
+ }
- void set_high_remb_bps(int value) { high_remb_bps_ = value; }
+ void set_high_remb_bps(int value) {
+ CriticalSectionScoped lock(crit_.get());
+ high_remb_bps_ = value;
+ }
void Stop() { transport_.StopSending(); }
@@ -905,7 +911,8 @@
kWaitingForStats
};
- virtual void SendRtcpFeedback(int remb_value) {
+ virtual void SendRtcpFeedback(int remb_value)
+ EXCLUSIVE_LOCKS_REQUIRED(crit_) {
FakeReceiveStatistics receive_stats(
kSendSsrc, last_sequence_number_, rtp_count_, 0);
RTCPSender rtcp_sender(0, false, clock_, &receive_stats);
@@ -923,15 +930,16 @@
internal::TransportAdapter transport_adapter_;
test::DirectTransport transport_;
- Clock* clock_;
- TestState test_state_;
- int rtp_count_;
- int last_sequence_number_;
- int suspended_frame_count_;
- int low_remb_bps_;
- int high_remb_bps_;
- scoped_ptr<CriticalSectionWrapper> crit_sect_;
- VideoSendStream** send_stream_ptr_;
+ Clock* const clock_;
+ VideoSendStream** const send_stream_ptr_;
+
+ const scoped_ptr<CriticalSectionWrapper> crit_;
+ TestState test_state_ GUARDED_BY(crit_);
+ int rtp_count_ GUARDED_BY(crit_);
+ int last_sequence_number_ GUARDED_BY(crit_);
+ int suspended_frame_count_ GUARDED_BY(crit_);
+ int low_remb_bps_ GUARDED_BY(crit_);
+ int high_remb_bps_ GUARDED_BY(crit_);
} observer(&send_stream_);
// Note that |send_stream_| is created in RunSendTest(), called below. This
// is why a pointer to |send_stream_| must be provided here.
@@ -961,26 +969,27 @@
PacketObserver()
: RtpRtcpObserver(30 * 1000), // Timeout after 30 seconds.
clock_(Clock::GetRealTimeClock()),
- last_packet_time_ms_(-1),
transport_adapter_(ReceiveTransport()),
- capturer_(NULL),
- crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {
+ crit_(CriticalSectionWrapper::CreateCriticalSection()),
+ last_packet_time_ms_(-1),
+ capturer_(NULL) {
transport_adapter_.Enable();
}
void SetCapturer(test::FrameGeneratorCapturer* capturer) {
+ CriticalSectionScoped lock(crit_.get());
capturer_ = capturer;
}
virtual Action OnSendRtp(const uint8_t* packet, size_t length) OVERRIDE {
- CriticalSectionScoped lock(crit_sect_.get());
+ CriticalSectionScoped lock(crit_.get());
last_packet_time_ms_ = clock_->TimeInMilliseconds();
capturer_->Stop();
return SEND_PACKET;
}
virtual Action OnSendRtcp(const uint8_t* packet, size_t length) OVERRIDE {
- CriticalSectionScoped lock(crit_sect_.get());
+ CriticalSectionScoped lock(crit_.get());
const int kVideoMutedThresholdMs = 10000;
if (last_packet_time_ms_ > 0 &&
clock_->TimeInMilliseconds() - last_packet_time_ms_ >
@@ -1002,11 +1011,11 @@
}
private:
- Clock* clock_;
- int64_t last_packet_time_ms_;
+ Clock* const clock_;
internal::TransportAdapter transport_adapter_;
- test::FrameGeneratorCapturer* capturer_;
- scoped_ptr<CriticalSectionWrapper> crit_sect_;
+ const scoped_ptr<CriticalSectionWrapper> crit_;
+ int64_t last_packet_time_ms_ GUARDED_BY(crit_);
+ test::FrameGeneratorCapturer* capturer_ GUARDED_BY(crit_);
} observer;
Call::Config call_config(observer.SendTransport());
@@ -1020,7 +1029,7 @@
test::FrameGeneratorCapturer::Create(
send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));
observer.SetCapturer(frame_generator_capturer.get());
- send_stream_->StartSending();
+ send_stream_->Start();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, observer.Wait())
@@ -1028,7 +1037,7 @@
observer.StopSending();
frame_generator_capturer->Stop();
- send_stream_->StopSending();
+ send_stream_->Stop();
call->DestroyVideoSendStream(send_stream_);
}
@@ -1107,7 +1116,7 @@
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));
- send_stream_->StartSending();
+ send_stream_->Start();
frame_generator_capturer->Start();
EXPECT_TRUE(observer.WaitForFilledStats())
@@ -1115,7 +1124,7 @@
observer.StopSending();
frame_generator_capturer->Stop();
- send_stream_->StopSending();
+ send_stream_->Stop();
call->DestroyVideoSendStream(send_stream_);
}
@@ -1200,7 +1209,7 @@
scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
test::FrameGeneratorCapturer::Create(
send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));
- send_stream_->StartSending();
+ send_stream_->Start();
frame_generator_capturer->Start();
EXPECT_EQ(kEventSignaled, observer.Wait())
@@ -1208,7 +1217,7 @@
observer.StopSending();
frame_generator_capturer->Stop();
- send_stream_->StopSending();
+ send_stream_->Stop();
call->DestroyVideoSendStream(send_stream_);
}
diff --git a/video_engine/test/auto_test/automated/vie_extended_integration_test.cc b/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
index 01d92d1..e43fc75 100644
--- a/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
+++ b/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
@@ -53,7 +53,7 @@
}
TEST_F(DISABLED_ON_MAC(ViEExtendedIntegrationTest),
- RunsRtpRtcpTestWithoutErrors) {
+ DISABLED_RunsRtpRtcpTestWithoutErrors) {
tests_->ViERtpRtcpExtendedTest();
}
diff --git a/video_engine/test/auto_test/automated/vie_standard_integration_test.cc b/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
index 51208ba..584f9e9 100644
--- a/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
+++ b/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
@@ -51,13 +51,8 @@
tests_->ViERenderStandardTest();
}
-// Fails on Mac, see https://code.google.com/p/webrtc/issues/detail?id=1790
-#if defined(WEBRTC_MAC)
-#define MAYBE_RunsRtpRtcpTestWithoutErrors DISABLED_RunsRtpRtcpTestWithoutErrors
-#else
-#define MAYBE_RunsRtpRtcpTestWithoutErrors RunsRtpRtcpTestWithoutErrors
-#endif
-TEST_F(ViEStandardIntegrationTest, MAYBE_RunsRtpRtcpTestWithoutErrors) {
+// Flaky, see webrtc:1790.
+TEST_F(ViEStandardIntegrationTest, DISABLED_RunsRtpRtcpTestWithoutErrors) {
tests_->ViERtpRtcpStandardTest();
}
diff --git a/video_engine/test/auto_test/source/vie_autotest_codec.cc b/video_engine/test/auto_test/source/vie_autotest_codec.cc
index d806b47..93738c2 100644
--- a/video_engine/test/auto_test/source/vie_autotest_codec.cc
+++ b/video_engine/test/auto_test/source/vie_autotest_codec.cc
@@ -510,6 +510,13 @@
break;
}
}
+ const unsigned int kMinBitrate = 123;
+ video_codec.minBitrate = kMinBitrate;
+ video_codec.startBitrate = 50;
+ EXPECT_EQ(0, codec->SetSendCodec(video_channel, video_codec));
+ EXPECT_EQ(0, codec->GetSendCodec(video_channel, video_codec));
+ EXPECT_EQ(kMinBitrate, video_codec.startBitrate);
+
memset(&video_codec, 0, sizeof(video_codec));
EXPECT_EQ(0, codec->GetSendCodec(video_channel, video_codec));
EXPECT_EQ(webrtc::kVideoCodecVP8, video_codec.codecType);
diff --git a/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc b/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
index 7864957..16c73f4 100644
--- a/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
+++ b/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
@@ -30,7 +30,7 @@
memcpy(this->buffer.get(), buffer, buffer_size);
}
- webrtc::scoped_array<unsigned char> buffer;
+ webrtc::scoped_ptr<unsigned char[]> buffer;
int buffer_size;
uint32_t timestamp;
int64_t render_time;
diff --git a/modules/NetEq.target.darwin-x86.mk b/video_engine/video_engine_core.target.darwin-arm64.mk
similarity index 62%
copy from modules/NetEq.target.darwin-x86.mk
copy to video_engine/video_engine_core.target.darwin-arm64.mk
index 3b05cd9..fd2c96a 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/video_engine/video_engine_core.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_video_engine_video_engine_core_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,42 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/video_engine/call_stats.cc \
+ third_party/webrtc/video_engine/encoder_state_feedback.cc \
+ third_party/webrtc/video_engine/overuse_frame_detector.cc \
+ third_party/webrtc/video_engine/stream_synchronization.cc \
+ third_party/webrtc/video_engine/vie_base_impl.cc \
+ third_party/webrtc/video_engine/vie_capture_impl.cc \
+ third_party/webrtc/video_engine/vie_codec_impl.cc \
+ third_party/webrtc/video_engine/vie_external_codec_impl.cc \
+ third_party/webrtc/video_engine/vie_image_process_impl.cc \
+ third_party/webrtc/video_engine/vie_impl.cc \
+ third_party/webrtc/video_engine/vie_network_impl.cc \
+ third_party/webrtc/video_engine/vie_ref_count.cc \
+ third_party/webrtc/video_engine/vie_render_impl.cc \
+ third_party/webrtc/video_engine/vie_rtp_rtcp_impl.cc \
+ third_party/webrtc/video_engine/vie_shared_data.cc \
+ third_party/webrtc/video_engine/vie_capturer.cc \
+ third_party/webrtc/video_engine/vie_channel.cc \
+ third_party/webrtc/video_engine/vie_channel_group.cc \
+ third_party/webrtc/video_engine/vie_channel_manager.cc \
+ third_party/webrtc/video_engine/vie_encoder.cc \
+ third_party/webrtc/video_engine/vie_file_image.cc \
+ third_party/webrtc/video_engine/vie_frame_provider_base.cc \
+ third_party/webrtc/video_engine/vie_input_manager.cc \
+ third_party/webrtc/video_engine/vie_manager_base.cc \
+ third_party/webrtc/video_engine/vie_receiver.cc \
+ third_party/webrtc/video_engine/vie_remb.cc \
+ third_party/webrtc/video_engine/vie_renderer.cc \
+ third_party/webrtc/video_engine/vie_render_manager.cc \
+ third_party/webrtc/video_engine/vie_sender.cc \
+ third_party/webrtc/video_engine/vie_sync_module.cc
# Flags passed to both C and C++ files.
@@ -70,19 +68,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +84,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +115,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +133,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,19 +163,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +179,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +210,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +229,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +259,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +273,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +295,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_video_engine_video_engine_core_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: video_engine_core
+video_engine_core: third_party_webrtc_video_engine_video_engine_core_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/video_engine/video_engine_core.target.linux-arm64.mk
similarity index 62%
copy from modules/NetEq.target.darwin-x86.mk
copy to video_engine/video_engine_core.target.linux-arm64.mk
index 3b05cd9..fd2c96a 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/video_engine/video_engine_core.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_video_engine_video_engine_core_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,42 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/video_engine/call_stats.cc \
+ third_party/webrtc/video_engine/encoder_state_feedback.cc \
+ third_party/webrtc/video_engine/overuse_frame_detector.cc \
+ third_party/webrtc/video_engine/stream_synchronization.cc \
+ third_party/webrtc/video_engine/vie_base_impl.cc \
+ third_party/webrtc/video_engine/vie_capture_impl.cc \
+ third_party/webrtc/video_engine/vie_codec_impl.cc \
+ third_party/webrtc/video_engine/vie_external_codec_impl.cc \
+ third_party/webrtc/video_engine/vie_image_process_impl.cc \
+ third_party/webrtc/video_engine/vie_impl.cc \
+ third_party/webrtc/video_engine/vie_network_impl.cc \
+ third_party/webrtc/video_engine/vie_ref_count.cc \
+ third_party/webrtc/video_engine/vie_render_impl.cc \
+ third_party/webrtc/video_engine/vie_rtp_rtcp_impl.cc \
+ third_party/webrtc/video_engine/vie_shared_data.cc \
+ third_party/webrtc/video_engine/vie_capturer.cc \
+ third_party/webrtc/video_engine/vie_channel.cc \
+ third_party/webrtc/video_engine/vie_channel_group.cc \
+ third_party/webrtc/video_engine/vie_channel_manager.cc \
+ third_party/webrtc/video_engine/vie_encoder.cc \
+ third_party/webrtc/video_engine/vie_file_image.cc \
+ third_party/webrtc/video_engine/vie_frame_provider_base.cc \
+ third_party/webrtc/video_engine/vie_input_manager.cc \
+ third_party/webrtc/video_engine/vie_manager_base.cc \
+ third_party/webrtc/video_engine/vie_receiver.cc \
+ third_party/webrtc/video_engine/vie_remb.cc \
+ third_party/webrtc/video_engine/vie_renderer.cc \
+ third_party/webrtc/video_engine/vie_render_manager.cc \
+ third_party/webrtc/video_engine/vie_sender.cc \
+ third_party/webrtc/video_engine/vie_sync_module.cc
# Flags passed to both C and C++ files.
@@ -70,19 +68,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +84,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +115,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +133,9 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,19 +163,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +179,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +210,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +229,9 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
- $(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/interface \
+ $(LOCAL_PATH)/third_party/webrtc/common_video/libyuv/include \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +259,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +273,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +295,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_video_engine_video_engine_core_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: video_engine_core
+video_engine_core: third_party_webrtc_video_engine_video_engine_core_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/video_engine/vie_capture_impl.cc b/video_engine/vie_capture_impl.cc
index c4f4344..9581fb7 100644
--- a/video_engine/vie_capture_impl.cc
+++ b/video_engine/vie_capture_impl.cc
@@ -213,7 +213,7 @@
int ViECaptureImpl::StartCapture(const int capture_id,
const CaptureCapability& capture_capability) {
- LOG(LS_ERROR) << "StartCapture " << capture_id;
+ LOG(LS_INFO) << "StartCapture " << capture_id;
ViEInputManagerScoped is(*(shared_data_->input_manager()));
ViECapturer* vie_capture = is.Capture(capture_id);
diff --git a/video_engine/vie_capturer.cc b/video_engine/vie_capturer.cc
index 94993ca..65159a1 100644
--- a/video_engine/vie_capturer.cc
+++ b/video_engine/vie_capturer.cc
@@ -535,7 +535,7 @@
unsigned int length = CalcBufferSize(kI420,
video_frame->width(),
video_frame->height());
- scoped_array<uint8_t> video_buffer(new uint8_t[length]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[length]);
ExtractBuffer(*video_frame, length, video_buffer.get());
effect_filter_->Transform(length,
video_buffer.get(),
diff --git a/video_engine/vie_channel.cc b/video_engine/vie_channel.cc
index 94210a3..d5e0878 100644
--- a/video_engine/vie_channel.cc
+++ b/video_engine/vie_channel.cc
@@ -1401,7 +1401,7 @@
unsigned int length = CalcBufferSize(kI420,
video_frame.width(),
video_frame.height());
- scoped_array<uint8_t> video_buffer(new uint8_t[length]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[length]);
ExtractBuffer(video_frame, length, video_buffer.get());
effect_filter_->Transform(length,
video_buffer.get(),
diff --git a/video_engine/vie_codec_impl.cc b/video_engine/vie_codec_impl.cc
index d2aaab9..3ba56de 100644
--- a/video_engine/vie_codec_impl.cc
+++ b/video_engine/vie_codec_impl.cc
@@ -29,19 +29,19 @@
static void LogCodec(const VideoCodec& codec) {
LOG(LS_INFO) << "CodecType " << codec.codecType
- << ", pl_type " << codec.plType
+ << ", pl_type " << static_cast<int>(codec.plType)
<< ", resolution " << codec.width
<< " x " << codec.height
<< ", start br " << codec.startBitrate
<< ", min br " << codec.minBitrate
<< ", max br " << codec.maxBitrate
- << ", max fpr " << codec.maxFramerate
+ << ", max fps " << static_cast<int>(codec.maxFramerate)
<< ", max qp " << codec.qpMax
<< ", number of streams "
- << codec.numberOfSimulcastStreams;
+ << static_cast<int>(codec.numberOfSimulcastStreams);
if (codec.codecType == kVideoCodecVP8) {
LOG(LS_INFO) << "VP8 specific settings";
- LOG(LS_INFO) << "pivtureLossIndicationOn "
+ LOG(LS_INFO) << "pictureLossIndicationOn "
<< codec.codecSpecific.VP8.pictureLossIndicationOn
<< ", feedbackModeOn "
<< codec.codecSpecific.VP8.feedbackModeOn
@@ -49,15 +49,17 @@
<< codec.codecSpecific.VP8.complexity
<< ", resilience "
<< codec.codecSpecific.VP8.resilience
- << ", numberOfTeporalLayers "
- << codec.codecSpecific.VP8.numberOfTemporalLayers
+ << ", numberOfTemporalLayers "
+ << static_cast<int>(
+ codec.codecSpecific.VP8.numberOfTemporalLayers)
<< ", keyFrameinterval "
<< codec.codecSpecific.VP8.keyFrameInterval;
for (int idx = 0; idx < codec.numberOfSimulcastStreams; ++idx) {
LOG(LS_INFO) << "Stream " << codec.simulcastStream[idx].width
<< " x " << codec.simulcastStream[idx].height;
LOG(LS_INFO) << "Temporal layers "
- << codec.simulcastStream[idx].numberOfTemporalLayers
+ << static_cast<int>(
+ codec.simulcastStream[idx].numberOfTemporalLayers)
<< ", min br "
<< codec.simulcastStream[idx].minBitrate
<< ", target br "
@@ -66,7 +68,6 @@
<< codec.simulcastStream[idx].maxBitrate
<< ", qp max "
<< codec.simulcastStream[idx].qpMax;
-
}
}
}
@@ -168,6 +169,9 @@
LOG(LS_INFO) << "New max bitrate set " << video_codec_internal.maxBitrate;
}
+ if (video_codec_internal.startBitrate < video_codec_internal.minBitrate) {
+ video_codec_internal.startBitrate = video_codec_internal.minBitrate;
+ }
if (video_codec_internal.startBitrate > video_codec_internal.maxBitrate) {
video_codec_internal.startBitrate = video_codec_internal.maxBitrate;
}
diff --git a/video_engine/vie_encoder.cc b/video_engine/vie_encoder.cc
index e2b70eb..957407d 100644
--- a/video_engine/vie_encoder.cc
+++ b/video_engine/vie_encoder.cc
@@ -503,7 +503,7 @@
unsigned int length = CalcBufferSize(kI420,
video_frame->width(),
video_frame->height());
- scoped_array<uint8_t> video_buffer(new uint8_t[length]);
+ scoped_ptr<uint8_t[]> video_buffer(new uint8_t[length]);
ExtractBuffer(*video_frame, length, video_buffer.get());
effect_filter_->Transform(length,
video_buffer.get(),
diff --git a/video_engine/vie_receiver.cc b/video_engine/vie_receiver.cc
index 75f53ab..76228b3 100644
--- a/video_engine/vie_receiver.cc
+++ b/video_engine/vie_receiver.cc
@@ -21,7 +21,9 @@
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/utility/interface/rtp_dump.h"
#include "webrtc/modules/video_coding/main/interface/video_coding.h"
+#include "webrtc/modules/video_coding/main/source/timestamp_extrapolator.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/trace.h"
@@ -45,6 +47,8 @@
rtp_rtcp_(NULL),
vcm_(module_vcm),
remote_bitrate_estimator_(remote_bitrate_estimator),
+ clock_(Clock::GetRealTimeClock()),
+ ts_extrapolator_(new VCMTimestampExtrapolator(clock_)),
rtp_dump_(NULL),
receiving_(false),
restored_packet_in_use_(false),
@@ -171,14 +175,37 @@
int32_t ViEReceiver::OnReceivedPayloadData(
const uint8_t* payload_data, const uint16_t payload_size,
const WebRtcRTPHeader* rtp_header) {
- // TODO(wu): Calculate ntp_time_ms
- if (vcm_->IncomingPacket(payload_data, payload_size, *rtp_header) != 0) {
+ WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
+ CalculateCaptureNtpTime(&rtp_header_with_ntp);
+ if (vcm_->IncomingPacket(payload_data,
+ payload_size,
+ rtp_header_with_ntp) != 0) {
// Check this...
return -1;
}
return 0;
}
+void ViEReceiver::CalculateCaptureNtpTime(WebRtcRTPHeader* rtp_header) {
+ if (rtcp_list_.size() < 2) {
+ // We need two RTCP SR reports to calculate NTP.
+ return;
+ }
+
+ int64_t sender_capture_ntp_ms = 0;
+ if (!synchronization::RtpToNtpMs(rtp_header->header.timestamp,
+ rtcp_list_,
+ &sender_capture_ntp_ms)) {
+ return;
+ }
+ uint32_t timestamp = sender_capture_ntp_ms * 90;
+ int64_t receiver_capture_ms =
+ ts_extrapolator_->ExtrapolateLocalTime(timestamp);
+ int64_t ntp_offset =
+ clock_->CurrentNtpInMilliseconds() - clock_->TimeInMilliseconds();
+ rtp_header->ntp_time_ms = receiver_capture_ms + ntp_offset;
+}
+
bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
int rtp_packet_length) {
RTPHeader header;
@@ -329,7 +356,56 @@
}
}
assert(rtp_rtcp_); // Should be set by owner at construction time.
- return rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
+ int ret = rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
+ if (ret != 0) {
+ return ret;
+ }
+
+ if (!GetRtcpTimestamp()) {
+ LOG(LS_WARNING) << "Failed to retrieve timestamp information from RTCP SR.";
+ }
+
+ return 0;
+}
+
+bool ViEReceiver::GetRtcpTimestamp() {
+ uint16_t rtt = 0;
+ rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL);
+ if (rtt == 0) {
+ // Waiting for valid rtt.
+ return true;
+ }
+
+ // Update RTCP list
+ uint32_t ntp_secs = 0;
+ uint32_t ntp_frac = 0;
+ uint32_t rtp_timestamp = 0;
+ if (0 != rtp_rtcp_->RemoteNTP(&ntp_secs,
+ &ntp_frac,
+ NULL,
+ NULL,
+ &rtp_timestamp)) {
+ return false;
+ }
+
+ bool new_rtcp_sr = false;
+ if (!synchronization::UpdateRtcpList(
+ ntp_secs, ntp_frac, rtp_timestamp, &rtcp_list_, &new_rtcp_sr)) {
+ return false;
+ }
+
+ if (!new_rtcp_sr) {
+ // No new RTCP SR since last time this function was called.
+ return true;
+ }
+
+ // Update extrapolator with the new arrival time.
+ // The extrapolator assumes the TimeInMilliseconds time.
+ int64_t receiver_arrival_time = clock_->TimeInMilliseconds();
+ int64_t sender_send_time_ms = Clock::NtpToMs(ntp_secs, ntp_frac);
+ int64_t sender_arrival_time_90k = (sender_send_time_ms + rtt / 2) * 90;
+ ts_extrapolator_->Update(receiver_arrival_time, sender_arrival_time_90k);
+ return true;
}
void ViEReceiver::StartReceive() {
diff --git a/video_engine/vie_receiver.h b/video_engine/vie_receiver.h
index b987c99..512a7b6 100644
--- a/video_engine/vie_receiver.h
+++ b/video_engine/vie_receiver.h
@@ -14,6 +14,7 @@
#include <list>
#include "webrtc/engine_configurations.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/rtp_to_ntp.h"
#include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
@@ -21,6 +22,9 @@
#include "webrtc/video_engine/include/vie_network.h"
#include "webrtc/video_engine/vie_defines.h"
+// TODO(wu): Move rtp_to_ntp.h and timestamp_extrapolator.h to somewhere that
+// can be shared between audio and video.
+
namespace webrtc {
class CriticalSectionWrapper;
@@ -32,6 +36,7 @@
class RTPPayloadRegistry;
class RtpReceiver;
class RtpRtcp;
+class VCMTimestampExtrapolator;
class VideoCodingModule;
struct ReceiveBandwidthEstimatorStats;
@@ -105,6 +110,9 @@
bool IsPacketInOrder(const RTPHeader& header) const;
bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
+ bool GetRtcpTimestamp();
+ void CalculateCaptureNtpTime(WebRtcRTPHeader* rtp_header);
+
scoped_ptr<CriticalSectionWrapper> receive_cs_;
const int32_t channel_id_;
scoped_ptr<RtpHeaderParser> rtp_header_parser_;
@@ -117,6 +125,10 @@
VideoCodingModule* vcm_;
RemoteBitrateEstimator* remote_bitrate_estimator_;
+ Clock* clock_;
+ scoped_ptr<VCMTimestampExtrapolator> ts_extrapolator_;
+ synchronization::RtcpList rtcp_list_;
+
RtpDump* rtp_dump_;
bool receiving_;
uint8_t restored_packet_[kViEMaxMtu];
diff --git a/video_engine/vie_sync_module.cc b/video_engine/vie_sync_module.cc
index 89da022..b7c74a7 100644
--- a/video_engine/vie_sync_module.cc
+++ b/video_engine/vie_sync_module.cc
@@ -30,31 +30,24 @@
return -1;
if (!receiver.LastReceivedTimeMs(&stream->latest_receive_time_ms))
return -1;
- synchronization::RtcpMeasurement measurement;
- if (0 != rtp_rtcp.RemoteNTP(&measurement.ntp_secs,
- &measurement.ntp_frac,
+
+ uint32_t ntp_secs = 0;
+ uint32_t ntp_frac = 0;
+ uint32_t rtp_timestamp = 0;
+ if (0 != rtp_rtcp.RemoteNTP(&ntp_secs,
+ &ntp_frac,
NULL,
NULL,
- &measurement.rtp_timestamp)) {
+ &rtp_timestamp)) {
return -1;
}
- if (measurement.ntp_secs == 0 && measurement.ntp_frac == 0) {
+
+ bool new_rtcp_sr = false;
+ if (!synchronization::UpdateRtcpList(
+ ntp_secs, ntp_frac, rtp_timestamp, &stream->rtcp, &new_rtcp_sr)) {
return -1;
}
- for (synchronization::RtcpList::iterator it = stream->rtcp.begin();
- it != stream->rtcp.end(); ++it) {
- if (measurement.ntp_secs == (*it).ntp_secs &&
- measurement.ntp_frac == (*it).ntp_frac) {
- // This RTCP has already been added to the list.
- return 0;
- }
- }
- // We need two RTCP SR reports to map between RTP and NTP. More than two will
- // not improve the mapping.
- if (stream->rtcp.size() == 2) {
- stream->rtcp.pop_back();
- }
- stream->rtcp.push_front(measurement);
+
return 0;
}
diff --git a/video_receive_stream.h b/video_receive_stream.h
index febb221..2584e20 100644
--- a/video_receive_stream.h
+++ b/video_receive_stream.h
@@ -171,8 +171,8 @@
int target_delay_ms;
};
- virtual void StartReceiving() = 0;
- virtual void StopReceiving() = 0;
+ virtual void Start() = 0;
+ virtual void Stop() = 0;
virtual Stats GetStats() const = 0;
// TODO(mflodman) Replace this with callback.
diff --git a/video_send_stream.h b/video_send_stream.h
index 4aa0419..aa027b0 100644
--- a/video_send_stream.h
+++ b/video_send_stream.h
@@ -159,8 +159,8 @@
// VideoSendStream is valid.
virtual VideoSendStreamInput* Input() = 0;
- virtual void StartSending() = 0;
- virtual void StopSending() = 0;
+ virtual void Start() = 0;
+ virtual void Stop() = 0;
// Set which streams to send. Must have at least as many SSRCs as configured
// in the config. Encoder settings are passed on to the encoder instance along
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index fb73c13..b3c53f8 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -3237,6 +3237,17 @@
return SetSendRtpHeaderExtension(enable, kRtpExtensionAudioLevel, id);
}
+int Channel::SetReceiveAudioLevelIndicationStatus(bool enable,
+ unsigned char id) {
+ rtp_header_parser_->DeregisterRtpHeaderExtension(
+ kRtpExtensionAudioLevel);
+ if (enable && !rtp_header_parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionAudioLevel, id)) {
+ return -1;
+ }
+ return 0;
+}
+
int Channel::SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id) {
return SetSendRtpHeaderExtension(enable, kRtpExtensionAbsoluteSendTime, id);
}
@@ -4268,7 +4279,7 @@
int32_t
Channel::MixOrReplaceAudioWithFile(int mixingFrequency)
{
- scoped_array<int16_t> fileBuffer(new int16_t[640]);
+ scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
int fileSamples(0);
{
@@ -4338,7 +4349,7 @@
{
assert(mixingFrequency <= 32000);
- scoped_array<int16_t> fileBuffer(new int16_t[640]);
+ scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
int fileSamples(0);
{
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 8f47b48..8166f1c 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -319,6 +319,7 @@
int GetRemoteSSRC(unsigned int& ssrc);
int GetRemoteCSRCs(unsigned int arrCSRC[15]);
int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
+ int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
int SetRTCPStatus(bool enable);
diff --git a/voice_engine/include/mock/fake_voe_external_media.h b/voice_engine/include/mock/fake_voe_external_media.h
index f45e1ba..b327f3c 100644
--- a/voice_engine/include/mock/fake_voe_external_media.h
+++ b/voice_engine/include/mock/fake_voe_external_media.h
@@ -53,7 +53,7 @@
int samples_per_channel, int sample_rate_hz,
int num_channels) {
const int length = samples_per_channel * num_channels;
- scoped_array<int16_t> data;
+ scoped_ptr<int16_t[]> data;
if (!audio) {
data.reset(new int16_t[length]);
memset(data.get(), 0, length * sizeof(data[0]));
diff --git a/voice_engine/include/voe_rtp_rtcp.h b/voice_engine/include/voe_rtp_rtcp.h
index 0dd649e..448cfc6 100644
--- a/voice_engine/include/voe_rtp_rtcp.h
+++ b/voice_engine/include/voe_rtp_rtcp.h
@@ -157,6 +157,15 @@
bool enable,
unsigned char id = 1) = 0;
+ // Sets the status of receiving rtp-audio-level-indication on a specific
+ // |channel|.
+ virtual int SetReceiveAudioLevelIndicationStatus(int channel,
+ bool enable,
+ unsigned char id = 1) {
+ // TODO(wu): Remove default implementation once talk is updated.
+ return 0;
+ }
+
// Sets the status of sending absolute sender time on a specific |channel|.
virtual int SetSendAbsoluteSenderTimeStatus(int channel,
bool enable,
diff --git a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc b/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
index ebb874e..408ebf9 100644
--- a/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
+++ b/voice_engine/test/auto_test/fixtures/before_initialization_fixture.cc
@@ -10,19 +10,10 @@
#include "webrtc/voice_engine/test/auto_test/fixtures/before_initialization_fixture.h"
-#include "gflags/gflags.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/system_wrappers/interface/sleep.h"
-DECLARE_bool(use_acm_version_2);
-
-BeforeInitializationFixture::BeforeInitializationFixture() {
- // TODO(minyue): Remove when the old ACM is removed (latest 2014-04-01).
- config_.Set<webrtc::AudioCodingModuleFactory>(FLAGS_use_acm_version_2 ?
- new webrtc::NewAudioCodingModuleFactory() :
- new webrtc::AudioCodingModuleFactory());
- voice_engine_ = webrtc::VoiceEngine::Create(config_);
-
+BeforeInitializationFixture::BeforeInitializationFixture()
+ : voice_engine_(webrtc::VoiceEngine::Create()) {
EXPECT_TRUE(voice_engine_ != NULL);
voe_base_ = webrtc::VoEBase::GetInterface(voice_engine_);
diff --git a/voice_engine/test/auto_test/voe_standard_test.cc b/voice_engine/test/auto_test/voe_standard_test.cc
index 5361445..944f3b1 100644
--- a/voice_engine/test/auto_test/voe_standard_test.cc
+++ b/voice_engine/test/auto_test/voe_standard_test.cc
@@ -10,11 +10,11 @@
#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
+#include <assert.h>
#include <stdio.h>
#include <string.h>
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/voice_engine/include/voe_neteq_stats.h"
#include "webrtc/voice_engine/test/auto_test/automated_mode.h"
@@ -29,8 +29,6 @@
DEFINE_bool(automated, false,
"If true, we'll run the automated tests we have in noninteractive "
"mode.");
-DEFINE_bool(use_acm_version_2, false,
- "If true, we'll run the tests with Audio Coding Module version 2.");
using namespace webrtc;
@@ -127,12 +125,7 @@
return false;
}
- // TODO(minyue): Remove when the old ACM is removed (latest 2014-04-01).
- config_.Set<AudioCodingModuleFactory>(FLAGS_use_acm_version_2 ?
- new NewAudioCodingModuleFactory() :
- new AudioCodingModuleFactory());
- voice_engine_ = VoiceEngine::Create(config_);
-
+ voice_engine_ = VoiceEngine::Create();
if (!voice_engine_) {
TEST_LOG("Failed to create VoiceEngine\n");
return false;
diff --git a/voice_engine/test/auto_test/voe_standard_test.h b/voice_engine/test/auto_test/voe_standard_test.h
index 566014d..3bf8936 100644
--- a/voice_engine/test/auto_test/voe_standard_test.h
+++ b/voice_engine/test/auto_test/voe_standard_test.h
@@ -15,7 +15,6 @@
#include <string>
#include "gflags/gflags.h"
-#include "webrtc/common.h"
#include "webrtc/voice_engine/include/voe_audio_processing.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_dtmf.h"
@@ -201,8 +200,6 @@
VoEAudioProcessing* voe_apm_;
ResourceManager resource_manager_;
-
- Config config_;
};
} // namespace voetest
diff --git a/voice_engine/test/cmd_test/voe_cmd_test.cc b/voice_engine/test/cmd_test/voe_cmd_test.cc
index 1d13a82..4f6f132 100644
--- a/voice_engine/test/cmd_test/voe_cmd_test.cc
+++ b/voice_engine/test/cmd_test/voe_cmd_test.cc
@@ -19,10 +19,7 @@
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/common.h"
-#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/test/channel_transport/include/channel_transport.h"
@@ -42,8 +39,6 @@
#include "webrtc/voice_engine/include/voe_video_sync.h"
#include "webrtc/voice_engine/include/voe_volume_control.h"
-DEFINE_bool(use_acm_version_1, false,
- "If true, we'll run the tests with Audio Coding Module version 2.");
DEFINE_bool(use_log_file, false,
"Output logs to a file; by default they will be printed to stderr.");
@@ -129,13 +124,7 @@
printf("Test started \n");
- // TODO(minyue): Remove when the old ACM is removed.
- Config config;
- config.Set<AudioCodingModuleFactory>(FLAGS_use_acm_version_1 ?
- new AudioCodingModuleFactory() :
- new NewAudioCodingModuleFactory());
- m_voe = VoiceEngine::Create(config);
-
+ m_voe = VoiceEngine::Create();
base1 = VoEBase::GetInterface(m_voe);
codec = VoECodec::GetInterface(m_voe);
apm = VoEAudioProcessing::GetInterface(m_voe);
diff --git a/voice_engine/transmit_mixer.cc b/voice_engine/transmit_mixer.cc
index d2467cf..cce2f94 100644
--- a/voice_engine/transmit_mixer.cc
+++ b/voice_engine/transmit_mixer.cc
@@ -1217,7 +1217,7 @@
int32_t TransmitMixer::MixOrReplaceAudioWithFile(
int mixingFrequency)
{
- scoped_array<int16_t> fileBuffer(new int16_t[640]);
+ scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
int fileSamples(0);
{
diff --git a/voice_engine/voe_rtp_rtcp_impl.cc b/voice_engine/voe_rtp_rtcp_impl.cc
index e20695a..e25a933 100644
--- a/voice_engine/voe_rtp_rtcp_impl.cc
+++ b/voice_engine/voe_rtp_rtcp_impl.cc
@@ -246,6 +246,36 @@
return channelPtr->SetSendAudioLevelIndicationStatus(enable, id);
}
+int VoERTP_RTCPImpl::SetReceiveAudioLevelIndicationStatus(int channel,
+ bool enable,
+ unsigned char id) {
+ WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
+ "SetReceiveAudioLevelIndicationStatus(channel=%d, enable=%d, id=%u)",
+ channel, enable, id);
+ if (!_shared->statistics().Initialized()) {
+ _shared->SetLastError(VE_NOT_INITED, kTraceError);
+ return -1;
+ }
+ if (enable &&
+ (id < kVoiceEngineMinRtpExtensionId ||
+ id > kVoiceEngineMaxRtpExtensionId)) {
+ // [RFC5285] The 4-bit id is the local identifier of this element in
+ // the range 1-14 inclusive.
+ _shared->SetLastError(VE_INVALID_ARGUMENT, kTraceError,
+ "SetReceiveAbsoluteSenderTimeStatus() invalid id parameter");
+ return -1;
+ }
+ // Set state and id for the specified channel.
+ voe::ChannelOwner ch = _shared->channel_manager().GetChannel(channel);
+ voe::Channel* channel_ptr = ch.channel();
+ if (channel_ptr == NULL) {
+ _shared->SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
+ "SetReceiveAudioLevelIndicationStatus() failed to locate channel");
+ return -1;
+ }
+ return channel_ptr->SetReceiveAudioLevelIndicationStatus(enable, id);
+}
+
int VoERTP_RTCPImpl::SetSendAbsoluteSenderTimeStatus(int channel,
bool enable,
unsigned char id) {
diff --git a/voice_engine/voe_rtp_rtcp_impl.h b/voice_engine/voe_rtp_rtcp_impl.h
index f902b66..b89c0dc 100644
--- a/voice_engine/voe_rtp_rtcp_impl.h
+++ b/voice_engine/voe_rtp_rtcp_impl.h
@@ -66,6 +66,9 @@
virtual int SetSendAudioLevelIndicationStatus(int channel,
bool enable,
unsigned char id);
+ virtual int SetReceiveAudioLevelIndicationStatus(int channel,
+ bool enable,
+ unsigned char id);
// RTP Header Extension for Absolute Sender Time
virtual int SetSendAbsoluteSenderTimeStatus(int channel,
diff --git a/modules/NetEq.target.darwin-x86.mk b/voice_engine/voice_engine.target.darwin-arm64.mk
similarity index 67%
copy from modules/NetEq.target.darwin-x86.mk
copy to voice_engine/voice_engine.target.darwin-arm64.mk
index 3b05cd9..5ae8e56 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/voice_engine/voice_engine.target.darwin-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_voice_engine_voice_engine_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,36 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/voice_engine/channel.cc \
+ third_party/webrtc/voice_engine/channel_manager.cc \
+ third_party/webrtc/voice_engine/dtmf_inband.cc \
+ third_party/webrtc/voice_engine/dtmf_inband_queue.cc \
+ third_party/webrtc/voice_engine/level_indicator.cc \
+ third_party/webrtc/voice_engine/monitor_module.cc \
+ third_party/webrtc/voice_engine/output_mixer.cc \
+ third_party/webrtc/voice_engine/shared_data.cc \
+ third_party/webrtc/voice_engine/statistics.cc \
+ third_party/webrtc/voice_engine/transmit_mixer.cc \
+ third_party/webrtc/voice_engine/utility.cc \
+ third_party/webrtc/voice_engine/voe_audio_processing_impl.cc \
+ third_party/webrtc/voice_engine/voe_base_impl.cc \
+ third_party/webrtc/voice_engine/voe_codec_impl.cc \
+ third_party/webrtc/voice_engine/voe_dtmf_impl.cc \
+ third_party/webrtc/voice_engine/voe_external_media_impl.cc \
+ third_party/webrtc/voice_engine/voe_file_impl.cc \
+ third_party/webrtc/voice_engine/voe_hardware_impl.cc \
+ third_party/webrtc/voice_engine/voe_neteq_stats_impl.cc \
+ third_party/webrtc/voice_engine/voe_network_impl.cc \
+ third_party/webrtc/voice_engine/voe_rtp_rtcp_impl.cc \
+ third_party/webrtc/voice_engine/voe_video_sync_impl.cc \
+ third_party/webrtc/voice_engine/voe_volume_control_impl.cc \
+ third_party/webrtc/voice_engine/voice_engine_impl.cc
# Flags passed to both C and C++ files.
@@ -70,19 +62,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +78,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +109,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +127,16 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_conference_mixer/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,19 +164,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +180,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +211,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +230,16 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_conference_mixer/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +267,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +281,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +303,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_voice_engine_voice_engine_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: voice_engine
+voice_engine: third_party_webrtc_voice_engine_voice_engine_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/modules/NetEq.target.darwin-x86.mk b/voice_engine/voice_engine.target.linux-arm64.mk
similarity index 67%
copy from modules/NetEq.target.darwin-x86.mk
copy to voice_engine/voice_engine.target.linux-arm64.mk
index 3b05cd9..5ae8e56 100644
--- a/modules/NetEq.target.darwin-x86.mk
+++ b/voice_engine/voice_engine.target.linux-arm64.mk
@@ -3,7 +3,7 @@
include $(CLEAR_VARS)
LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := third_party_webrtc_modules_NetEq_gyp
+LOCAL_MODULE := third_party_webrtc_voice_engine_voice_engine_gyp
LOCAL_MODULE_SUFFIX := .a
LOCAL_MODULE_TAGS := optional
LOCAL_MODULE_TARGET_ARCH := $(TARGET_$(GYP_VAR_PREFIX)ARCH)
@@ -18,44 +18,36 @@
# Make sure our deps and generated files are built first.
LOCAL_ADDITIONAL_DEPENDENCIES := $(GYP_TARGET_DEPENDENCIES) $(GYP_GENERATED_OUTPUTS)
+LOCAL_CPP_EXTENSION := .cc
LOCAL_GENERATED_SOURCES :=
GYP_COPIED_SOURCE_ORIGIN_DIRS :=
LOCAL_SRC_FILES := \
- third_party/webrtc/modules/audio_coding/neteq/accelerate.c \
- third_party/webrtc/modules/audio_coding/neteq/automode.c \
- third_party/webrtc/modules/audio_coding/neteq/bgn_update.c \
- third_party/webrtc/modules/audio_coding/neteq/bufstats_decision.c \
- third_party/webrtc/modules/audio_coding/neteq/cng_internal.c \
- third_party/webrtc/modules/audio_coding/neteq/codec_db.c \
- third_party/webrtc/modules/audio_coding/neteq/correlator.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp.c \
- third_party/webrtc/modules/audio_coding/neteq/dsp_helpfunctions.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/dtmf_tonegen.c \
- third_party/webrtc/modules/audio_coding/neteq/expand.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_address_init.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_dsp_common.c \
- third_party/webrtc/modules/audio_coding/neteq/mcu_reset.c \
- third_party/webrtc/modules/audio_coding/neteq/merge.c \
- third_party/webrtc/modules/audio_coding/neteq/min_distortion.c \
- third_party/webrtc/modules/audio_coding/neteq/mix_voice_unvoice.c \
- third_party/webrtc/modules/audio_coding/neteq/mute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/normal.c \
- third_party/webrtc/modules/audio_coding/neteq/packet_buffer.c \
- third_party/webrtc/modules/audio_coding/neteq/peak_detection.c \
- third_party/webrtc/modules/audio_coding/neteq/preemptive_expand.c \
- third_party/webrtc/modules/audio_coding/neteq/random_vector.c \
- third_party/webrtc/modules/audio_coding/neteq/recin.c \
- third_party/webrtc/modules/audio_coding/neteq/recout.c \
- third_party/webrtc/modules/audio_coding/neteq/rtcp.c \
- third_party/webrtc/modules/audio_coding/neteq/rtp.c \
- third_party/webrtc/modules/audio_coding/neteq/set_fs.c \
- third_party/webrtc/modules/audio_coding/neteq/signal_mcu.c \
- third_party/webrtc/modules/audio_coding/neteq/split_and_insert.c \
- third_party/webrtc/modules/audio_coding/neteq/unmute_signal.c \
- third_party/webrtc/modules/audio_coding/neteq/webrtc_neteq.c
+ third_party/webrtc/voice_engine/channel.cc \
+ third_party/webrtc/voice_engine/channel_manager.cc \
+ third_party/webrtc/voice_engine/dtmf_inband.cc \
+ third_party/webrtc/voice_engine/dtmf_inband_queue.cc \
+ third_party/webrtc/voice_engine/level_indicator.cc \
+ third_party/webrtc/voice_engine/monitor_module.cc \
+ third_party/webrtc/voice_engine/output_mixer.cc \
+ third_party/webrtc/voice_engine/shared_data.cc \
+ third_party/webrtc/voice_engine/statistics.cc \
+ third_party/webrtc/voice_engine/transmit_mixer.cc \
+ third_party/webrtc/voice_engine/utility.cc \
+ third_party/webrtc/voice_engine/voe_audio_processing_impl.cc \
+ third_party/webrtc/voice_engine/voe_base_impl.cc \
+ third_party/webrtc/voice_engine/voe_codec_impl.cc \
+ third_party/webrtc/voice_engine/voe_dtmf_impl.cc \
+ third_party/webrtc/voice_engine/voe_external_media_impl.cc \
+ third_party/webrtc/voice_engine/voe_file_impl.cc \
+ third_party/webrtc/voice_engine/voe_hardware_impl.cc \
+ third_party/webrtc/voice_engine/voe_neteq_stats_impl.cc \
+ third_party/webrtc/voice_engine/voe_network_impl.cc \
+ third_party/webrtc/voice_engine/voe_rtp_rtcp_impl.cc \
+ third_party/webrtc/voice_engine/voe_video_sync_impl.cc \
+ third_party/webrtc/voice_engine/voe_volume_control_impl.cc \
+ third_party/webrtc/voice_engine/voice_engine_impl.cc
# Flags passed to both C and C++ files.
@@ -70,19 +62,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -94,7 +78,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-g \
-fomit-frame-pointer \
@@ -126,8 +109,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -146,12 +127,16 @@
LOCAL_C_INCLUDES_Debug := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_conference_mixer/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -179,19 +164,11 @@
-fvisibility=hidden \
-pipe \
-fPIC \
+ -Wno-unused-local-typedefs \
-fno-builtin-cos \
-fno-builtin-sin \
-fno-builtin-cosf \
-fno-builtin-sinf \
- -fno-builtin-cos \
- -fno-builtin-sin \
- -fno-builtin-cosf \
- -fno-builtin-sinf \
- -msse2 \
- -mfpmath=sse \
- -mmmx \
- -m32 \
- -fuse-ld=gold \
-ffunction-sections \
-funwind-tables \
-g \
@@ -203,7 +180,6 @@
-Wno-ignored-qualifiers \
-Wno-type-limits \
-Wno-unused-but-set-variable \
- -fno-stack-protector \
-Os \
-fno-ident \
-fdata-sections \
@@ -235,8 +211,6 @@
'-DWEBRTC_LINUX' \
'-DWEBRTC_ANDROID' \
'-DWEBRTC_ANDROID_OPENSLES' \
- '-DNETEQ_VOICEENGINE_CODECS' \
- '-DSCRATCH' \
'-DUSE_OPENSSL=1' \
'-DUSE_OPENSSL_CERTS=1' \
'-D__STDC_CONSTANT_MACROS' \
@@ -256,12 +230,16 @@
LOCAL_C_INCLUDES_Release := \
$(LOCAL_PATH)/third_party \
$(LOCAL_PATH) \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/neteq/interface \
- $(LOCAL_PATH)/third_party/webrtc \
- $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/codecs/cng/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/resampler/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/signal_processing/include \
$(LOCAL_PATH)/third_party/webrtc/common_audio/vad/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_coding/main/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/interface \
+ $(LOCAL_PATH)/third_party/webrtc \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_conference_mixer/interface \
+ $(LOCAL_PATH)/third_party/webrtc/modules/audio_device/include \
+ $(LOCAL_PATH)/third_party/webrtc/modules/media_file/interface \
+ $(LOCAL_PATH)/third_party/webrtc/system_wrappers/interface \
$(PWD)/frameworks/wilhelm/include \
$(PWD)/bionic \
$(PWD)/external/stlport/stlport
@@ -289,8 +267,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -305,8 +281,6 @@
-Wl,--fatal-warnings \
-Wl,-z,noexecstack \
-fPIC \
- -m32 \
- -fuse-ld=gold \
-nostdlib \
-Wl,--no-undefined \
-Wl,--exclude-libs=ALL \
@@ -329,10 +303,10 @@
# Add target alias to "gyp_all_modules" target.
.PHONY: gyp_all_modules
-gyp_all_modules: third_party_webrtc_modules_NetEq_gyp
+gyp_all_modules: third_party_webrtc_voice_engine_voice_engine_gyp
# Alias gyp target name.
-.PHONY: NetEq
-NetEq: third_party_webrtc_modules_NetEq_gyp
+.PHONY: voice_engine
+voice_engine: third_party_webrtc_voice_engine_voice_engine_gyp
include $(BUILD_STATIC_LIBRARY)
diff --git a/webrtc_tests.gypi b/webrtc_tests.gypi
index 57fd907..1ff3a9c 100644
--- a/webrtc_tests.gypi
+++ b/webrtc_tests.gypi
@@ -63,7 +63,7 @@
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
- 'modules/modules.gyp:neteq_unittest_tools', # Needed by neteq_performance_unittest.
+ 'modules/modules.gyp:neteq_test_support', # Needed by neteq_performance_unittest.
'modules/modules.gyp:rtp_rtcp',
'test/webrtc_test_common.gyp:webrtc_test_common',
'webrtc',
