diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 4fdb1f61ad..4646eb138e 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -131,7 +131,7 @@ if (rtc_include_tests) { "../api/audio_codecs:audio_codecs_api", "../api/audio_codecs/opus:audio_decoder_opus", "../api/audio_codecs/opus:audio_encoder_opus", - "../api/task_queue:global_task_queue_factory", + "../api/task_queue:default_task_queue_factory", "../api/units:time_delta", "../call:mock_bitrate_allocator", "../call:mock_call_interfaces", @@ -142,12 +142,8 @@ if (rtc_include_tests) { "../common_audio", "../logging:mocks", "../logging:rtc_event_log_api", + "../modules/audio_device:audio_device_impl", # For TestAudioDeviceModule "../modules/audio_device:mock_audio_device", - "../rtc_base:rtc_base_tests_utils", - "../test:field_trial", - - # For TestAudioDeviceModule - "../modules/audio_device:audio_device_impl", "../modules/audio_mixer:audio_mixer_impl", "../modules/audio_processing:audio_processing_statistics", "../modules/audio_processing:mocks", @@ -157,11 +153,13 @@ if (rtc_include_tests) { "../modules/utility", "../rtc_base:checks", "../rtc_base:rtc_base_approved", - "../rtc_base:rtc_task_queue", + "../rtc_base:rtc_base_tests_utils", "../rtc_base:safe_compare", + "../rtc_base:task_queue_for_test", "../rtc_base:timeutils", "../system_wrappers", "../test:audio_codec_mocks", + "../test:field_trial", "../test:rtp_test_utils", "../test:test_common", "../test:test_support", @@ -254,6 +252,7 @@ if (rtc_include_tests) { "../call:simulated_network", "../common_audio", "../rtc_base:rtc_base_approved", + "../rtc_base:task_queue_for_test", "../system_wrappers", "../test:field_trial", "../test:fileutils", diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc index 1d946c5ed8..0db569bb53 100644 --- a/audio/audio_send_stream_unittest.cc +++ b/audio/audio_send_stream_unittest.cc @@ -13,7 +13,7 @@ #include #include "absl/memory/memory.h" -#include "api/task_queue/global_task_queue_factory.h" +#include "api/task_queue/default_task_queue_factory.h" #include "api/test/mock_frame_encryptor.h" #include "audio/audio_send_stream.h" #include "audio/audio_state.h" @@ -28,7 +28,7 @@ #include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h" #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" -#include "rtc_base/task_queue.h" +#include "rtc_base/task_queue_for_test.h" #include "system_wrappers/include/clock.h" #include "test/field_trial.h" #include "test/gtest.h" @@ -128,10 +128,13 @@ rtc::scoped_refptr SetupEncoderFactoryMock() { struct ConfigHelper { ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call) : clock_(1000000), + task_queue_factory_(CreateDefaultTaskQueueFactory()), stream_config_(/*send_transport=*/nullptr, /*media_transport=*/nullptr), audio_processing_(new rtc::RefCountedObject()), bitrate_allocator_(&clock_, &limit_observer_), - worker_queue_("ConfigHelper_worker_queue"), + worker_queue_(task_queue_factory_->CreateTaskQueue( + "ConfigHelper_worker_queue", + TaskQueueFactory::Priority::NORMAL)), audio_encoder_(nullptr) { using testing::Invoke; @@ -167,7 +170,7 @@ struct ConfigHelper { return std::unique_ptr( new internal::AudioSendStream( Clock::GetRealTimeClock(), stream_config_, audio_state_, - &GlobalTaskQueueFactory(), &rtp_transport_, &bitrate_allocator_, + task_queue_factory_.get(), &rtp_transport_, &bitrate_allocator_, &event_log_, &rtcp_rtt_stats_, absl::nullopt, std::unique_ptr(channel_send_))); } @@ -289,6 +292,7 @@ struct ConfigHelper { private: SimulatedClock clock_; + std::unique_ptr task_queue_factory_; rtc::scoped_refptr audio_state_; AudioSendStream::Config stream_config_; testing::StrictMock* channel_send_ = nullptr; @@ -303,7 +307,7 @@ struct ConfigHelper { BitrateAllocator bitrate_allocator_; // |worker_queue| is defined last to ensure all pending tasks are cancelled // and deleted before any other members. - rtc::TaskQueue worker_queue_; + TaskQueueForTest worker_queue_; std::unique_ptr audio_encoder_; }; } // namespace diff --git a/audio/test/audio_bwe_integration_test.cc b/audio/test/audio_bwe_integration_test.cc index 8a8c30675a..950d54070a 100644 --- a/audio/test/audio_bwe_integration_test.cc +++ b/audio/test/audio_bwe_integration_test.cc @@ -16,6 +16,7 @@ #include "call/fake_network_pipe.h" #include "call/simulated_network.h" #include "common_audio/wav_file.h" +#include "rtc_base/task_queue_for_test.h" #include "system_wrappers/include/sleep.h" #include "test/field_trial.h" #include "test/gtest.h" @@ -150,7 +151,7 @@ class NoBandwidthDropAfterDtx : public AudioBweTest { private: Call* sender_call_; - rtc::TaskQueue stats_poller_; + TaskQueueForTest stats_poller_; }; using AudioBweIntegrationTest = CallTest; diff --git a/audio/test/media_transport_test.cc b/audio/test/media_transport_test.cc index e07b79e8b6..5ab13c5b4b 100644 --- a/audio/test/media_transport_test.cc +++ b/audio/test/media_transport_test.cc @@ -13,7 +13,7 @@ #include "api/audio_codecs/audio_encoder_factory_template.h" #include "api/audio_codecs/opus/audio_decoder_opus.h" #include "api/audio_codecs/opus/audio_encoder_opus.h" -#include "api/task_queue/global_task_queue_factory.h" +#include "api/task_queue/default_task_queue_factory.h" #include "api/test/loopback_media_transport.h" #include "api/test/mock_audio_mixer.h" #include "audio/audio_receive_stream.h" @@ -25,7 +25,6 @@ #include "modules/audio_mixer/audio_mixer_impl.h" #include "modules/audio_processing/include/mock_audio_processing.h" #include "modules/utility/include/process_thread.h" -#include "rtc_base/task_queue.h" #include "rtc_base/time_utils.h" #include "test/gtest.h" #include "test/mock_transport.h" @@ -123,13 +122,15 @@ TEST(AudioWithMediaTransport, DeliversAudio) { send_config.encoder_factory = CreateAudioEncoderFactory(); std::unique_ptr send_process_thread = ProcessThread::Create("audio send thread"); + std::unique_ptr task_queue_factory = + CreateDefaultTaskQueueFactory(); RtpTransportControllerSend rtp_transport( Clock::GetRealTimeClock(), null_event_log.get(), nullptr, BitrateConstraints(), ProcessThread::Create("Pacer"), - &GlobalTaskQueueFactory()); + task_queue_factory.get()); webrtc::internal::AudioSendStream send_stream( Clock::GetRealTimeClock(), send_config, audio_state, - &GlobalTaskQueueFactory(), send_process_thread.get(), &rtp_transport, + task_queue_factory.get(), send_process_thread.get(), &rtp_transport, &bitrate_allocator, null_event_log.get(), /*rtcp_rtt_stats=*/nullptr, absl::optional());