ad89528051
This reverts commit 42d8c93ec351b68554825b58a3dc6525a7dc84da. Reason for revert: Got Aliby for FEC test flakes Original change's description: > Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" > > This reverts commit 304e9d2df347630d71fd4423f5971f30dac73e41. > > Reason for revert: Breaks downstream projects. > Seems to make VideoSendStreamTest.SupportsFlexfecSimulcastVp8 flaky. > > Original change's description: > > Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current > > > > Bug: webrtc:10191 > > Change-Id: I506cc50a90c73a6a4f6a3de36de0999cca72f5ba > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126230 > > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#27035} > > TBR=danilchap@webrtc.org,kwiberg@webrtc.org > > Change-Id: If98324f88e4b3d18bf2fe33597dfb9711867c243 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10191 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126484 > Reviewed-by: Yves Gerey <yvesg@webrtc.org> > Commit-Queue: Yves Gerey <yvesg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27041} TBR=danilchap@webrtc.org,kwiberg@webrtc.org,yvesg@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10191 Change-Id: Id87a17ae415142b8e0b11ba03ae7bad84a473fb0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126720 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Yves Gerey <yvesg@webrtc.org> Commit-Queue: Yves Gerey <yvesg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27056}
272 lines
7.9 KiB
Plaintext
272 lines
7.9 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_static_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_send.cc",
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"channel_send.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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"transport_feedback_packet_loss_tracker.cc",
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"transport_feedback_packet_loss_tracker.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:libjingle_peerconnection_api",
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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_mixer_api",
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"../api/audio_codecs:audio_codecs_api",
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"../api/task_queue",
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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_event_log_api",
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"../logging:rtc_stream_config",
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"../modules/audio_coding",
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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_device",
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"../modules/audio_processing",
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"../modules/audio_processing:api",
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"../modules/bitrate_controller:bitrate_controller",
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"../modules/pacing:pacing",
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"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
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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: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",
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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/experiments:audio_allocation_settings",
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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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"//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_source_set("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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"../call:fake_network",
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"../call:simulated_network",
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"../system_wrappers:system_wrappers",
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"../test:test_common",
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"../test:test_support",
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"//third_party/abseil-cpp/absl/memory",
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]
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}
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rtc_source_set("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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"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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"test/media_transport_test.cc",
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"transport_feedback_packet_loss_tracker_unittest.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:loopback_media_transport",
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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/task_queue:global_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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"../logging:rtc_event_log_api",
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"../modules/audio_device:mock_audio_device",
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"../rtc_base:rtc_base_tests_utils",
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"../test:field_trial",
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# For TestAudioDeviceModule
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"../modules/audio_device:audio_device_impl",
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"../modules/audio_mixer:audio_mixer_impl",
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"../modules/audio_processing:audio_processing_statistics",
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"../modules/audio_processing:mocks",
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"../modules/bitrate_controller:mocks",
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"../modules/pacing: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_task_queue",
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"../rtc_base:safe_compare",
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"../rtc_base:timeutils",
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"../system_wrappers:system_wrappers",
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"../test:audio_codec_mocks",
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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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"//third_party/abseil-cpp/absl/memory",
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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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]
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deps = [
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":audio_end_to_end_test",
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"../api:simulated_network_api",
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"../common_audio",
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"../rtc_base:rtc_base_approved",
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"../system_wrappers",
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"../test:fileutils",
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"../test:test_common",
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"../test:test_main",
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"//testing/gtest",
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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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]
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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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]
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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_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_source_set("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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"../system_wrappers",
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"../test:field_trial",
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"../test:fileutils",
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"../test:single_threaded_task_queue",
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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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"//third_party/abseil-cpp/absl/memory",
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]
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data = [
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"//resources/voice_engine/audio_dtx16.wav",
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]
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}
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}
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