e5f4c6b8d2
This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5. The original CL didn't attach the definition of the macro NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have to be related to //rtc_base anymore but to //rtc_base:threading). Original change's description: > Refactor rtc_base build targets. > > The "//rtc_base:rtc_base" build target has historically been one of the > biggest targets in the WebRTC build. Big targets are the main source of > circular dependencies and non-API types leakage. > > This CL is a step forward into splitting "//rtc_base:rtc_base" into > smaller targets (as originally started in 2018). > > The only non-automated changes are (like re-wiring the build system): > * The creation of //rtc_base/async_resolver.{h,cc} which allows to > break a circular dependency (is has been extracted from > //rtc_base/net_helpers.{h,cc}). > * The creation of //rtc_base/internal/default_socket_server.{h,cc} to > break another circular dependency. > > Bug: webrtc:9987 > Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32941} Bug: webrtc:9987 Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33001}
309 lines
9.5 KiB
Plaintext
309 lines
9.5 KiB
Plaintext
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE file in the root of the source
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# tree. An additional intellectual property rights grant can be found
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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import("../webrtc.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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import("//build/config/android/rules.gni")
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}
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rtc_library("audio") {
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sources = [
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"audio_level.cc",
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"audio_level.h",
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"audio_receive_stream.cc",
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"audio_receive_stream.h",
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"audio_send_stream.cc",
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"audio_send_stream.h",
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"audio_state.cc",
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"audio_state.h",
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"audio_transport_impl.cc",
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"audio_transport_impl.h",
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"channel_receive.cc",
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"channel_receive.h",
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"channel_receive_frame_transformer_delegate.cc",
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"channel_receive_frame_transformer_delegate.h",
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"channel_send.cc",
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"channel_send.h",
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"channel_send_frame_transformer_delegate.cc",
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"channel_send_frame_transformer_delegate.h",
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"conversion.h",
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"null_audio_poller.cc",
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"null_audio_poller.h",
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"remix_resample.cc",
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"remix_resample.h",
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]
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deps = [
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"../api:array_view",
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"../api:call_api",
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"../api:frame_transformer_interface",
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"../api:function_view",
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"../api:rtp_headers",
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"../api:rtp_parameters",
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"../api:scoped_refptr",
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"../api:transport_api",
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"../api/audio:aec3_factory",
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"../api/audio:audio_frame_api",
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"../api/audio:audio_frame_processor",
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"../api/audio:audio_mixer_api",
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"../api/audio_codecs:audio_codecs_api",
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"../api/crypto:frame_decryptor_interface",
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"../api/crypto:frame_encryptor_interface",
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"../api/crypto:options",
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"../api/neteq:neteq_api",
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"../api/rtc_event_log",
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"../api/task_queue",
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"../api/transport/rtp:rtp_source",
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"../call:audio_sender_interface",
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"../call:bitrate_allocator",
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"../call:call_interfaces",
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"../call:rtp_interfaces",
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"../common_audio",
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"../common_audio:common_audio_c",
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"../logging:rtc_event_audio",
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"../logging:rtc_stream_config",
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"../modules/async_audio_processing",
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"../modules/audio_coding",
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"../modules/audio_coding:audio_coding_module_typedefs",
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"../modules/audio_coding:audio_encoder_cng",
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"../modules/audio_coding:audio_network_adaptor_config",
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"../modules/audio_coding:red",
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"../modules/audio_device",
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"../modules/audio_processing",
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"../modules/audio_processing:api",
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"../modules/audio_processing:audio_frame_proxies",
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"../modules/audio_processing:rms_level",
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"../modules/pacing",
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"../modules/rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/utility",
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"../rtc_base",
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"../rtc_base:audio_format_to_string",
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"../rtc_base:checks",
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"../rtc_base:rate_limiter",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_task_queue",
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"../rtc_base:safe_minmax",
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"../rtc_base:threading",
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"../rtc_base/experiments:field_trial_parser",
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"../rtc_base/synchronization:mutex",
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"../rtc_base/synchronization:sequence_checker",
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"../rtc_base/system:no_unique_address",
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"../rtc_base/task_utils:to_queued_task",
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"../system_wrappers",
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"../system_wrappers:field_trial",
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"../system_wrappers:metrics",
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"utility:audio_frame_operations",
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]
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absl_deps = [
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"//third_party/abseil-cpp/absl/memory",
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"//third_party/abseil-cpp/absl/types:optional",
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]
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}
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if (rtc_include_tests) {
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rtc_library("audio_end_to_end_test") {
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testonly = true
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sources = [
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"test/audio_end_to_end_test.cc",
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"test/audio_end_to_end_test.h",
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]
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deps = [
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":audio",
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"../api:simulated_network_api",
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"../api/task_queue",
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"../call:fake_network",
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"../call:simulated_network",
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"../system_wrappers",
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"../test:test_common",
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"../test:test_support",
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]
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}
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rtc_library("audio_tests") {
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testonly = true
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sources = [
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"audio_receive_stream_unittest.cc",
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"audio_send_stream_tests.cc",
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"audio_send_stream_unittest.cc",
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"audio_state_unittest.cc",
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"channel_receive_frame_transformer_delegate_unittest.cc",
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"channel_send_frame_transformer_delegate_unittest.cc",
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"mock_voe_channel_proxy.h",
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"remix_resample_unittest.cc",
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"test/audio_stats_test.cc",
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]
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deps = [
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":audio",
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":audio_end_to_end_test",
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"../api:libjingle_peerconnection_api",
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"../api:mock_audio_mixer",
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"../api:mock_frame_decryptor",
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"../api:mock_frame_encryptor",
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"../api/audio:audio_frame_api",
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"../api/audio_codecs:audio_codecs_api",
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"../api/audio_codecs/opus:audio_decoder_opus",
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"../api/audio_codecs/opus:audio_encoder_opus",
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"../api/rtc_event_log",
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"../api/task_queue:default_task_queue_factory",
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"../api/units:time_delta",
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"../call:mock_bitrate_allocator",
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"../call:mock_call_interfaces",
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"../call:mock_rtp_interfaces",
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"../call:rtp_interfaces",
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"../call:rtp_receiver",
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"../call:rtp_sender",
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"../common_audio",
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"../logging:mocks",
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"../modules/audio_device:audio_device_impl", # For TestAudioDeviceModule
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"../modules/audio_device:mock_audio_device",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/audio_mixer:audio_mixer_test_utils",
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"../modules/audio_processing:audio_processing_statistics",
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"../modules/audio_processing:mocks",
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"../modules/pacing",
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"../modules/rtp_rtcp:mock_rtp_rtcp",
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"../modules/rtp_rtcp:rtp_rtcp_format",
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"../modules/utility",
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"../rtc_base:checks",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:rtc_base_tests_utils",
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"../rtc_base:safe_compare",
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"../rtc_base:task_queue_for_test",
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"../rtc_base:timeutils",
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"../system_wrappers",
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"../test:audio_codec_mocks",
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"../test:field_trial",
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"../test:mock_frame_transformer",
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"../test:mock_transformable_frame",
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"../test:mock_transport",
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"../test:rtp_test_utils",
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"../test:test_common",
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"../test:test_support",
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"utility:utility_tests",
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"//testing/gtest",
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]
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}
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if (rtc_enable_protobuf) {
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rtc_test("low_bandwidth_audio_test") {
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testonly = true
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sources = [
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"test/low_bandwidth_audio_test.cc",
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"test/low_bandwidth_audio_test_flags.cc",
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"test/pc_low_bandwidth_audio_test.cc",
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]
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deps = [
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":audio_end_to_end_test",
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"../api:create_network_emulation_manager",
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"../api:create_peerconnection_quality_test_fixture",
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"../api:network_emulation_manager_api",
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"../api:peer_connection_quality_test_fixture_api",
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"../api:simulated_network_api",
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"../api:time_controller",
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"../call:simulated_network",
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"../common_audio",
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"../system_wrappers",
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"../test:fileutils",
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"../test:perf_test",
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"../test:test_common",
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"../test:test_main",
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"../test:test_support",
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"../test/pc/e2e:network_quality_metrics_reporter",
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"//testing/gtest",
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"//third_party/abseil-cpp/absl/flags:flag",
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]
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if (is_android) {
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deps += [ "//testing/android/native_test:native_test_native_code" ]
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}
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data = [
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"../resources/voice_engine/audio_tiny16.wav",
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"../resources/voice_engine/audio_tiny48.wav",
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]
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}
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group("low_bandwidth_audio_perf_test") {
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testonly = true
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deps = [
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":low_bandwidth_audio_test",
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"//third_party/catapult/tracing/tracing/proto:histogram_proto",
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"//third_party/protobuf:py_proto_runtime",
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]
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data = [
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"test/low_bandwidth_audio_test.py",
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"../resources/voice_engine/audio_tiny16.wav",
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"../resources/voice_engine/audio_tiny48.wav",
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"${root_out_dir}/pyproto/tracing/tracing/proto/histogram_pb2.py",
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]
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# TODO(http://crbug.com/1029452): Create a cleaner target with just the
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# tracing python code. We don't need Polymer for instance.
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data_deps = [ "//third_party/catapult/tracing:convert_chart_json" ]
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if (is_win) {
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data += [ "${root_out_dir}/low_bandwidth_audio_test.exe" ]
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} else {
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data += [ "${root_out_dir}/low_bandwidth_audio_test" ]
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}
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if (is_linux || is_chromeos || is_android) {
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data += [
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"../tools_webrtc/audio_quality/linux/PolqaOem64",
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"../tools_webrtc/audio_quality/linux/pesq",
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]
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}
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if (is_win) {
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data += [
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"../tools_webrtc/audio_quality/win/PolqaOem64.dll",
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"../tools_webrtc/audio_quality/win/PolqaOem64.exe",
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"../tools_webrtc/audio_quality/win/pesq.exe",
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"../tools_webrtc/audio_quality/win/vcomp120.dll",
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]
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}
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if (is_mac) {
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data += [ "../tools_webrtc/audio_quality/mac/pesq" ]
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}
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write_runtime_deps = "${root_out_dir}/${target_name}.runtime_deps"
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}
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}
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rtc_library("audio_perf_tests") {
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testonly = true
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sources = [
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"test/audio_bwe_integration_test.cc",
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"test/audio_bwe_integration_test.h",
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]
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deps = [
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"../api:simulated_network_api",
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"../api/task_queue",
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"../call:fake_network",
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"../call:simulated_network",
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"../common_audio",
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"../rtc_base:rtc_base_approved",
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"../rtc_base:task_queue_for_test",
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"../system_wrappers",
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"../test:field_trial",
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"../test:fileutils",
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"../test:test_common",
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"../test:test_main",
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"../test:test_support",
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"//testing/gtest",
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]
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data = [ "//resources/voice_engine/audio_dtx16.wav" ]
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}
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}
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