Add support of specifying audio sample rate for PC test framework Bug: webrtc:10138 Change-Id: I6f868ede4b762884d7b2e9e7dac51bc60e9925d0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144940 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28513}
diff --git a/api/test/peerconnection_quality_test_fixture.h b/api/test/peerconnection_quality_test_fixture.h index 515dcfc..aa2481a 100644 --- a/api/test/peerconnection_quality_test_fixture.h +++ b/api/test/peerconnection_quality_test_fixture.h
@@ -196,8 +196,11 @@ absl::optional<std::string> input_dump_file_name; // If specified the output stream will be copied to specified file. absl::optional<std::string> output_dump_file_name; + // Audio options to use. cricket::AudioOptions audio_options; + // Sampling frequency of input audio data (from file or generated). + int sampling_frequency_in_hz = 48000; }; // This class is used to fully configure one peer inside the call.
diff --git a/test/pc/e2e/peer_connection_e2e_smoke_test.cc b/test/pc/e2e/peer_connection_e2e_smoke_test.cc index d5d4dcd..8c6b2cf 100644 --- a/test/pc/e2e/peer_connection_e2e_smoke_test.cc +++ b/test/pc/e2e/peer_connection_e2e_smoke_test.cc
@@ -148,6 +148,7 @@ audio.mode = AudioConfig::Mode::kFile; audio.input_file_name = test::ResourcePath("pc_quality_smoke_test_alice_source", "wav"); + audio.sampling_frequency_in_hz = 48000; alice->SetAudioConfig(std::move(audio)); }, [](PeerConfigurer* bob) {
diff --git a/test/pc/e2e/peer_connection_quality_test.cc b/test/pc/e2e/peer_connection_quality_test.cc index dbd8d5b..13e45ef 100644 --- a/test/pc/e2e/peer_connection_quality_test.cc +++ b/test/pc/e2e/peer_connection_quality_test.cc
@@ -257,13 +257,10 @@ // Audio streams are intercepted in AudioDeviceModule, so if it is required to // catch output of Alice's stream, Alice's output_dump_file_name should be // passed to Bob's TestPeer setup as audio output file name. - absl::optional<std::string> alice_audio_output_dump_file_name = - bob_params->audio_config ? bob_params->audio_config->output_dump_file_name - : absl::nullopt; - absl::optional<std::string> bob_audio_output_dump_file_name = - alice_params->audio_config - ? alice_params->audio_config->output_dump_file_name - : absl::nullopt; + absl::optional<TestPeer::RemotePeerAudioConfig> alice_remote_audio_config = + TestPeer::CreateRemoteAudioConfig(bob_params->audio_config); + absl::optional<TestPeer::RemotePeerAudioConfig> bob_remote_audio_config = + TestPeer::CreateRemoteAudioConfig(alice_params->audio_config); // Copy Alice and Bob video configs to correctly pass them into lambdas. std::vector<VideoConfig> alice_video_configs = alice_params->video_configs; std::vector<VideoConfig> bob_video_configs = bob_params->video_configs; @@ -277,8 +274,8 @@ }, [this]() { StartVideo(alice_video_sources_); }), video_quality_analyzer_injection_helper_.get(), signaling_thread.get(), - alice_audio_output_dump_file_name, - run_params.video_encoder_bitrate_multiplier, task_queue_.get()); + alice_remote_audio_config, run_params.video_encoder_bitrate_multiplier, + task_queue_.get()); bob_ = TestPeer::CreateTestPeer( std::move(bob_components), std::move(bob_params), absl::make_unique<FixturePeerConnectionObserver>( @@ -288,8 +285,8 @@ }, [this]() { StartVideo(bob_video_sources_); }), video_quality_analyzer_injection_helper_.get(), signaling_thread.get(), - bob_audio_output_dump_file_name, - run_params.video_encoder_bitrate_multiplier, task_queue_.get()); + bob_remote_audio_config, run_params.video_encoder_bitrate_multiplier, + task_queue_.get()); int num_cores = CpuInfo::DetectNumberOfCores(); RTC_DCHECK_GE(num_cores, 1);
diff --git a/test/pc/e2e/test_peer.cc b/test/pc/e2e/test_peer.cc index cf36bb7..4a3b650 100644 --- a/test/pc/e2e/test_peer.cc +++ b/test/pc/e2e/test_peer.cc
@@ -32,8 +32,13 @@ namespace webrtc_pc_e2e { namespace { +using RemotePeerAudioConfig = + ::webrtc::webrtc_pc_e2e::TestPeer::RemotePeerAudioConfig; +using AudioConfig = + ::webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::AudioConfig; + constexpr int16_t kGeneratedAudioMaxAmplitude = 32000; -constexpr int kSamplingFrequencyInHz = 48000; +constexpr int kDefaultSamplingFrequencyInHz = 48000; // Sets mandatory entities in injectable components like |pcf_dependencies| // and |pc_dependencies| if they are omitted. Also setup required @@ -59,7 +64,6 @@ } struct TestPeerComponents { - using AudioConfig = PeerConnectionE2EQualityTestFixture::AudioConfig; rtc::scoped_refptr<PeerConnectionFactoryInterface> peer_connection_factory; rtc::scoped_refptr<PeerConnectionInterface> peer_connection; @@ -70,7 +74,7 @@ MockPeerConnectionObserver* observer, VideoQualityAnalyzerInjectionHelper* video_analyzer_helper, rtc::Thread* signaling_thread, - absl::optional<std::string> audio_output_file_name, + absl::optional<RemotePeerAudioConfig> remote_audio_config, double bitrate_multiplier, rtc::TaskQueue* task_queue) { std::map<std::string, absl::optional<int>> stream_required_spatial_index; @@ -102,7 +106,7 @@ std::move(components->pcf_dependencies), params.audio_config, bitrate_multiplier, std::move(stream_required_spatial_index), video_analyzer_helper, components->network_thread, signaling_thread, - std::move(audio_output_file_name), task_queue); + std::move(remote_audio_config), task_queue); peer_connection_factory = CreateModularPeerConnectionFactory(std::move(pcf_deps)); @@ -118,7 +122,7 @@ AudioConfig audio_config) { if (audio_config.mode == AudioConfig::Mode::kGenerated) { return TestAudioDeviceModule::CreatePulsedNoiseCapturer( - kGeneratedAudioMaxAmplitude, kSamplingFrequencyInHz); + kGeneratedAudioMaxAmplitude, audio_config.sampling_frequency_in_hz); } if (audio_config.mode == AudioConfig::Mode::kFile) { RTC_DCHECK(audio_config.input_file_name); @@ -132,7 +136,7 @@ rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule( TaskQueueFactory* task_queue_factory, absl::optional<AudioConfig> audio_config, - absl::optional<std::string> audio_output_file_name) { + absl::optional<RemotePeerAudioConfig> remote_audio_config) { std::unique_ptr<TestAudioDeviceModule::Capturer> capturer; if (audio_config) { capturer = CreateAudioCapturer(audio_config.value()); @@ -141,7 +145,7 @@ // In such case use generated capturer. Despite of we provided audio here, // in test media setup audio stream won't be added into peer connection. capturer = TestAudioDeviceModule::CreatePulsedNoiseCapturer( - kGeneratedAudioMaxAmplitude, kSamplingFrequencyInHz); + kGeneratedAudioMaxAmplitude, kDefaultSamplingFrequencyInHz); } RTC_DCHECK(capturer); @@ -151,12 +155,13 @@ } std::unique_ptr<TestAudioDeviceModule::Renderer> renderer; - if (audio_output_file_name) { + if (remote_audio_config && remote_audio_config->output_file_name) { renderer = TestAudioDeviceModule::CreateBoundedWavFileWriter( - audio_output_file_name.value(), kSamplingFrequencyInHz); + remote_audio_config->output_file_name.value(), + remote_audio_config->sampling_frequency_in_hz); } else { - renderer = - TestAudioDeviceModule::CreateDiscardRenderer(kSamplingFrequencyInHz); + renderer = TestAudioDeviceModule::CreateDiscardRenderer( + kDefaultSamplingFrequencyInHz); } return TestAudioDeviceModule::Create(task_queue_factory, @@ -202,12 +207,12 @@ double bitrate_multiplier, std::map<std::string, absl::optional<int>> stream_required_spatial_index, VideoQualityAnalyzerInjectionHelper* video_analyzer_helper, - absl::optional<std::string> audio_output_file_name) { + absl::optional<RemotePeerAudioConfig> remote_audio_config) { cricket::MediaEngineDependencies media_deps; media_deps.task_queue_factory = pcf_dependencies->task_queue_factory.get(); media_deps.adm = CreateAudioDeviceModule(media_deps.task_queue_factory, std::move(audio_config), - std::move(audio_output_file_name)); + std::move(remote_audio_config)); media_deps.audio_processing = audio_processing; media_deps.video_encoder_factory = CreateVideoEncoderFactory( pcf_dependencies, video_analyzer_helper, bitrate_multiplier, @@ -230,7 +235,7 @@ VideoQualityAnalyzerInjectionHelper* video_analyzer_helper, rtc::Thread* network_thread, rtc::Thread* signaling_thread, - absl::optional<std::string> audio_output_file_name, + absl::optional<RemotePeerAudioConfig> remote_audio_config, rtc::TaskQueue* task_queue) { PeerConnectionFactoryDependencies pcf_deps; pcf_deps.network_thread = network_thread; @@ -238,7 +243,7 @@ pcf_deps.media_engine = CreateMediaEngine( pcf_dependencies.get(), std::move(audio_config), bitrate_multiplier, std::move(stream_required_spatial_index), video_analyzer_helper, - std::move(audio_output_file_name)); + std::move(remote_audio_config)); pcf_deps.call_factory = std::move(pcf_dependencies->call_factory); pcf_deps.event_log_factory = std::move(pcf_dependencies->event_log_factory); @@ -293,13 +298,21 @@ } // namespace +absl::optional<RemotePeerAudioConfig> TestPeer::CreateRemoteAudioConfig( + absl::optional<AudioConfig> config) { + if (!config) { + return absl::nullopt; + } + return RemotePeerAudioConfig(config.value()); +} + std::unique_ptr<TestPeer> TestPeer::CreateTestPeer( std::unique_ptr<InjectableComponents> components, std::unique_ptr<Params> params, std::unique_ptr<MockPeerConnectionObserver> observer, VideoQualityAnalyzerInjectionHelper* video_analyzer_helper, rtc::Thread* signaling_thread, - absl::optional<std::string> audio_output_file_name, + absl::optional<RemotePeerAudioConfig> remote_audio_config, double bitrate_multiplier, rtc::TaskQueue* task_queue) { RTC_DCHECK(components); @@ -309,7 +322,7 @@ TestPeerComponents tpc(std::move(components), *params, observer.get(), video_analyzer_helper, signaling_thread, - std::move(audio_output_file_name), bitrate_multiplier, + std::move(remote_audio_config), bitrate_multiplier, task_queue); return absl::WrapUnique(new TestPeer(
diff --git a/test/pc/e2e/test_peer.h b/test/pc/e2e/test_peer.h index df335cc..8cb8415 100644 --- a/test/pc/e2e/test_peer.h +++ b/test/pc/e2e/test_peer.h
@@ -37,6 +37,18 @@ using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig; using AudioConfig = PeerConnectionE2EQualityTestFixture::AudioConfig; + struct RemotePeerAudioConfig { + RemotePeerAudioConfig(AudioConfig config) + : sampling_frequency_in_hz(config.sampling_frequency_in_hz), + output_file_name(config.output_dump_file_name) {} + + int sampling_frequency_in_hz; + absl::optional<std::string> output_file_name; + }; + + static absl::optional<RemotePeerAudioConfig> CreateRemoteAudioConfig( + absl::optional<AudioConfig> config); + // Setups all components, that should be provided to WebRTC // PeerConnectionFactory and PeerConnection creation methods, // also will setup dependencies, that are required for media analyzers @@ -54,7 +66,7 @@ std::unique_ptr<MockPeerConnectionObserver> observer, VideoQualityAnalyzerInjectionHelper* video_analyzer_helper, rtc::Thread* signaling_thread, - absl::optional<std::string> audio_output_file_name, + absl::optional<RemotePeerAudioConfig> remote_audio_config, double bitrate_multiplier, rtc::TaskQueue* task_queue);