webrtc_m130/audio/BUILD.gn

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# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ ) Reason for revert: Starting to work on a fix (it seems that there are third_party dependencies that depends on the path to the webrtc.gni file) Original issue's description: > Revert of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2651543003/ ) > > Reason for revert: > This was causing the following failure: https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder/builds/838/steps/generate_build_files/logs/stdio > > Original issue's description: > > Moving webrtc.gni up one level from build/ > > > > BUG=webrtc:7030 > > > > Review-Url: https://codereview.webrtc.org/2651543003 > > Cr-Commit-Position: refs/heads/master@{#16241} > > Committed: https://chromium.googlesource.com/external/webrtc/+/35a32700fc9b5d932ddbd528c12f59c3274e4774 > > TBR=kjellander@webrtc.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:7030 > > Review-Url: https://codereview.webrtc.org/2657563002 > Cr-Commit-Position: refs/heads/master@{#16244} > Committed: https://chromium.googlesource.com/external/webrtc/+/69dc7dbe247ead087f3bae0eb7e23f27f0de1ec3 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7030 Review-Url: https://codereview.webrtc.org/2654773002 Cr-Commit-Position: refs/heads/master@{#16247}
2017-01-24 06:58:22 -08:00
import("../webrtc.gni")
if (is_android) {
import("//build/config/android/config.gni")
import("//build/config/android/rules.gni")
}
rtc_static_library("audio") {
sources = [
"audio_receive_stream.cc",
"audio_receive_stream.h",
"audio_send_stream.cc",
"audio_send_stream.h",
"audio_state.cc",
"audio_state.h",
"audio_transport_proxy.cc",
"audio_transport_proxy.h",
"conversion.h",
"null_audio_poller.cc",
"null_audio_poller.h",
"scoped_voe_interface.h",
"time_interval.cc",
"time_interval.h",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
"..:webrtc_common",
"../api:audio_mixer_api",
"../api:call_api",
"../api:optional",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../call:bitrate_allocator",
"../call:call_interfaces",
"../call:rtp_interfaces",
Replace AudioConferenceMixer with AudioMixer. This CL re-routes audio through AudioMixer instead of AudioConferenceMixer. This is done without any modifications to VoiceEngine. Previously, output audio was polled by an AudioDevice through an AudioTransport pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the request for data on to OutputMixer and further to AudioConferenceMixer. This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice to have another AudioTransport pointer, which points to an AudioTransportProxy. The AudioTransportProxy is responsible for feeding mixed data to the AudioProcessing component for echo cancellation, and to resample the audio data after AudioProcessing and before it is sent to the AudioDevice. The set up of the audio path was previously done during VoiceEngine initialization. Now it is changed in the AudioState constructor. This list shows where audio-path-related VoiceEngine functionality has been moved: OutputMixer --> AudioTransportProxy VoiceEngineImpl --> AudioState, AudioTransportProxy SharedData --> AudioState Channel --> AudioReceiveStream, ChannelProxy, Channel AudioState owns the new mixer and connects it to AudioTransport and AudioDevice on initialization. The audio input source is AudioReceiveStream, which registers itself with the mixer (which it gets from AudioState) on Start and Stop. # Since the AudioTransport interface contains non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2436033002 Cr-Commit-Position: refs/heads/master@{#15193}
2016-11-22 06:42:53 -08:00
"../common_audio",
"../modules/audio_coding:cng",
"../modules/audio_device",
"../modules/audio_processing",
"../modules/bitrate_controller:bitrate_controller",
"../modules/congestion_controller:congestion_controller",
"../modules/pacing:pacing",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp:rtp_rtcp",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../system_wrappers",
"../voice_engine",
]
}
if (rtc_include_tests) {
rtc_source_set("audio_end_to_end_test") {
testonly = true
sources = [
"test/audio_end_to_end_test.cc",
"test/audio_end_to_end_test.h",
]
deps = [
":audio",
"../system_wrappers:system_wrappers",
"../test:fake_audio_device",
"../test:test_common",
"../test:test_support",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_source_set("audio_tests") {
testonly = true
# TODO(kjellander): Remove (bugs.webrtc.org/6828)
# This needs remote_bitrate_estimator to be moved to webrtc/api first.
check_includes = false
sources = [
"audio_receive_stream_unittest.cc",
"audio_send_stream_tests.cc",
"audio_send_stream_unittest.cc",
"audio_state_unittest.cc",
"time_interval_unittest.cc",
]
deps = [
":audio",
":audio_end_to_end_test",
Replace AudioConferenceMixer with AudioMixer. This CL re-routes audio through AudioMixer instead of AudioConferenceMixer. This is done without any modifications to VoiceEngine. Previously, output audio was polled by an AudioDevice through an AudioTransport pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the request for data on to OutputMixer and further to AudioConferenceMixer. This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice to have another AudioTransport pointer, which points to an AudioTransportProxy. The AudioTransportProxy is responsible for feeding mixed data to the AudioProcessing component for echo cancellation, and to resample the audio data after AudioProcessing and before it is sent to the AudioDevice. The set up of the audio path was previously done during VoiceEngine initialization. Now it is changed in the AudioState constructor. This list shows where audio-path-related VoiceEngine functionality has been moved: OutputMixer --> AudioTransportProxy VoiceEngineImpl --> AudioState, AudioTransportProxy SharedData --> AudioState Channel --> AudioReceiveStream, ChannelProxy, Channel AudioState owns the new mixer and connects it to AudioTransport and AudioDevice on initialization. The audio input source is AudioReceiveStream, which registers itself with the mixer (which it gets from AudioState) on Start and Stop. # Since the AudioTransport interface contains non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2436033002 Cr-Commit-Position: refs/heads/master@{#15193}
2016-11-22 06:42:53 -08:00
"../api:mock_audio_mixer",
"../call:rtp_receiver",
"../modules/audio_device:mock_audio_device",
Replace AudioConferenceMixer with AudioMixer. This CL re-routes audio through AudioMixer instead of AudioConferenceMixer. This is done without any modifications to VoiceEngine. Previously, output audio was polled by an AudioDevice through an AudioTransport pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the request for data on to OutputMixer and further to AudioConferenceMixer. This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice to have another AudioTransport pointer, which points to an AudioTransportProxy. The AudioTransportProxy is responsible for feeding mixed data to the AudioProcessing component for echo cancellation, and to resample the audio data after AudioProcessing and before it is sent to the AudioDevice. The set up of the audio path was previously done during VoiceEngine initialization. Now it is changed in the AudioState constructor. This list shows where audio-path-related VoiceEngine functionality has been moved: OutputMixer --> AudioTransportProxy VoiceEngineImpl --> AudioState, AudioTransportProxy SharedData --> AudioState Channel --> AudioReceiveStream, ChannelProxy, Channel AudioState owns the new mixer and connects it to AudioTransport and AudioDevice on initialization. The audio input source is AudioReceiveStream, which registers itself with the mixer (which it gets from AudioState) on Start and Stop. # Since the AudioTransport interface contains non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2436033002 Cr-Commit-Position: refs/heads/master@{#15193}
2016-11-22 06:42:53 -08:00
"../modules/audio_mixer:audio_mixer_impl",
"../modules/congestion_controller:congestion_controller",
"../modules/congestion_controller:mock_congestion_controller",
"../modules/pacing:pacing",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../test:test_common",
"../test:test_support",
"utility:utility_tests",
"//testing/gmock",
"//testing/gtest",
]
if (!rtc_use_memcheck) {
# This test is timing dependent, which rules out running on memcheck bots.
sources += [ "test/audio_stats_test.cc" ]
}
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
if (rtc_enable_protobuf) {
rtc_test("low_bandwidth_audio_test") {
testonly = true
sources = [
"test/low_bandwidth_audio_test.cc",
]
deps = [
":audio_end_to_end_test",
"../common_audio",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../test:fake_audio_device",
"../test:test_common",
"../test:test_main",
"//testing/gmock",
"//testing/gtest",
]
if (is_android) {
deps += [ "//testing/android/native_test:native_test_native_code" ]
}
data = [
"../resources/voice_engine/audio_dtx16.wav",
"../resources/voice_engine/audio_tiny16.wav",
"../resources/voice_engine/audio_tiny48.wav",
"test/low_bandwidth_audio_test.py",
]
if (is_linux) {
data += [
"../tools_webrtc/audio_quality/linux/PolqaOem64",
"../tools_webrtc/audio_quality/linux/pesq",
]
}
if (is_win) {
data += [
"../tools_webrtc/audio_quality/win/PolqaOem64.dll",
"../tools_webrtc/audio_quality/win/PolqaOem64.exe",
"../tools_webrtc/audio_quality/win/pesq.exe",
"../tools_webrtc/audio_quality/win/vcomp120.dll",
]
}
if (is_mac) {
data += [ "../tools_webrtc/audio_quality/mac/pesq" ]
}
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}
rtc_source_set("audio_perf_tests") {
testonly = true
sources = [
"test/audio_bwe_integration_test.cc",
"test/audio_bwe_integration_test.h",
]
deps = [
"../common_audio",
"../rtc_base:rtc_base_approved",
"../system_wrappers",
"../test:fake_audio_device",
"../test:field_trial",
"../test:single_threaded_task_queue",
"../test:test_common",
"../test:test_main",
"//testing/gmock",
"//testing/gtest",
]
data = [
"//resources/voice_engine/audio_dtx16.wav",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}