| /* | |
| * Copyright (c) 2013 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. | |
| */ | |
| #include <algorithm> | |
| #include <cstddef> | |
| #include <cstdint> | |
| #include <cstdlib> | |
| #include <iterator> | |
| #include <limits> | |
| #include <map> | |
| #include <memory> | |
| #include <string> | |
| #include <utility> | |
| #include <vector> | |
| #include "absl/flags/flag.h" | |
| #include "absl/strings/string_view.h" | |
| #include "api/array_view.h" | |
| #include "api/audio/audio_device.h" | |
| #include "api/audio/builtin_audio_processing_builder.h" | |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" | |
| #include "api/environment/environment.h" | |
| #include "api/field_trials_view.h" | |
| #include "api/make_ref_counted.h" | |
| #include "api/numerics/samples_stats_counter.h" | |
| #include "api/rtp_parameters.h" | |
| #include "api/scoped_refptr.h" | |
| #include "api/task_queue/pending_task_safety_flag.h" | |
| #include "api/task_queue/task_queue_base.h" | |
| #include "api/test/metrics/global_metrics_logger_and_exporter.h" | |
| #include "api/test/metrics/metric.h" | |
| #include "api/test/simulated_network.h" | |
| #include "api/test/video/function_video_encoder_factory.h" | |
| #include "api/transport/bitrate_settings.h" | |
| #include "api/units/data_rate.h" | |
| #include "api/units/time_delta.h" | |
| #include "api/units/timestamp.h" | |
| #include "api/video/builtin_video_bitrate_allocator_factory.h" | |
| #include "api/video/video_bitrate_allocation.h" | |
| #include "api/video/video_bitrate_allocator_factory.h" | |
| #include "api/video/video_frame.h" | |
| #include "api/video/video_sink_interface.h" | |
| #include "api/video/video_source_interface.h" | |
| #include "api/video_codecs/sdp_video_format.h" | |
| #include "api/video_codecs/video_codec.h" | |
| #include "api/video_codecs/video_encoder.h" | |
| #include "api/video_codecs/video_encoder_factory.h" | |
| #include "call/audio_receive_stream.h" | |
| #include "call/audio_send_stream.h" | |
| #include "call/audio_state.h" | |
| #include "call/call.h" | |
| #include "call/call_config.h" | |
| #include "call/fake_network_pipe.h" | |
| #include "call/video_receive_stream.h" | |
| #include "call/video_send_stream.h" | |
| #include "media/engine/internal_encoder_factory.h" | |
| #include "media/engine/simulcast_encoder_adapter.h" | |
| #include "modules/audio_device/include/test_audio_device.h" | |
| #include "modules/audio_mixer/audio_mixer_impl.h" | |
| #include "rtc_base/checks.h" | |
| #include "rtc_base/event.h" | |
| #include "rtc_base/logging.h" | |
| #include "rtc_base/task_queue_for_test.h" | |
| #include "rtc_base/thread.h" | |
| #include "system_wrappers/include/metrics.h" | |
| #include "test/call_test.h" | |
| #include "test/drifting_clock.h" | |
| #include "test/encoder_settings.h" | |
| #include "test/fake_encoder.h" | |
| #include "test/frame_generator_capturer.h" | |
| #include "test/gtest.h" | |
| #include "test/network/simulated_network.h" | |
| #include "test/rtp_rtcp_observer.h" | |
| #include "test/test_flags.h" | |
| #include "test/video_encoder_proxy_factory.h" | |
| #include "test/video_test_constants.h" | |
| #include "video/config/video_encoder_config.h" | |
| using webrtc::test::DriftingClock; | |
| namespace webrtc { | |
| namespace { | |
| using test::GetGlobalMetricsLogger; | |
| using test::ImprovementDirection; | |
| using test::Unit; | |
| enum : int { // The first valid value is 1. | |
| kTransportSequenceNumberExtensionId = 1, | |
| }; | |
| } // namespace | |
| class CallPerfTest : public test::CallTest { | |
| public: | |
| CallPerfTest() { | |
| RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, | |
| kTransportSequenceNumberExtensionId)); | |
| } | |
| protected: | |
| enum class FecMode { kOn, kOff }; | |
| enum class CreateOrder { kAudioFirst, kVideoFirst }; | |
| void TestAudioVideoSync(FecMode fec, | |
| CreateOrder create_first, | |
| float video_ntp_speed, | |
| float video_rtp_speed, | |
| float audio_rtp_speed, | |
| absl::string_view test_label); | |
| void TestMinTransmitBitrate(bool pad_to_min_bitrate); | |
| void TestCaptureNtpTime(const BuiltInNetworkBehaviorConfig& net_config, | |
| int threshold_ms, | |
| int start_time_ms, | |
| int run_time_ms); | |
| void TestMinAudioVideoBitrate(int test_bitrate_from, | |
| int test_bitrate_to, | |
| int test_bitrate_step, | |
| int min_bwe, | |
| int start_bwe, | |
| int max_bwe); | |
| void TestEncodeFramerate(VideoEncoderFactory* encoder_factory, | |
| absl::string_view payload_name, | |
| const std::vector<int>& max_framerates); | |
| }; | |
| class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver, | |
| public VideoSinkInterface<VideoFrame> { | |
| static const int kInSyncThresholdMs = 50; | |
| static const int kStartupTimeMs = 2000; | |
| static const int kMinRunTimeMs = 30000; | |
| public: | |
| explicit VideoRtcpAndSyncObserver(TaskQueueBase* task_queue, | |
| Clock* clock, | |
| absl::string_view test_label) | |
| : test::RtpRtcpObserver(test::VideoTestConstants::kLongTimeout), | |
| clock_(clock), | |
| test_label_(test_label), | |
| creation_time_ms_(clock_->TimeInMilliseconds()), | |
| task_queue_(task_queue) {} | |
| void OnFrame(const VideoFrame& /* video_frame */) override { | |
| task_queue_->PostTask([this]() { CheckStats(); }); | |
| } | |
| void CheckStats() { | |
| if (!receive_stream_) | |
| return; | |
| VideoReceiveStreamInterface::Stats stats = receive_stream_->GetStats(); | |
| if (stats.sync_offset_ms == std::numeric_limits<int>::max()) | |
| return; | |
| Timestamp now = clock_->CurrentTime(); | |
| int64_t time_since_creation = now.ms() - creation_time_ms_; | |
| // During the first couple of seconds audio and video can falsely be | |
| // estimated as being synchronized. We don't want to trigger on those. | |
| if (time_since_creation < kStartupTimeMs) | |
| return; | |
| if (std::abs(stats.sync_offset_ms) < kInSyncThresholdMs) { | |
| if (first_time_in_sync_ == -1) { | |
| first_time_in_sync_ = now.ms(); | |
| GetGlobalMetricsLogger()->LogSingleValueMetric( | |
| "sync_convergence_time" + test_label_, "synchronization", | |
| time_since_creation, Unit::kMilliseconds, | |
| ImprovementDirection::kSmallerIsBetter); | |
| } | |
| if (time_since_creation > kMinRunTimeMs) | |
| observation_complete_.Set(); | |
| } | |
| if (first_time_in_sync_ != -1) | |
| sync_offset_ms_list_.AddSample( | |
| {.value = static_cast<double>(stats.sync_offset_ms), .time = now}); | |
| } | |
| void set_receive_stream(VideoReceiveStreamInterface* receive_stream) { | |
| RTC_DCHECK_EQ(task_queue_, TaskQueueBase::Current()); | |
| // Note that receive_stream may be nullptr. | |
| receive_stream_ = receive_stream; | |
| } | |
| void PrintResults() { | |
| GetGlobalMetricsLogger()->LogMetric( | |
| "stream_offset" + test_label_, "synchronization", sync_offset_ms_list_, | |
| Unit::kMilliseconds, ImprovementDirection::kNeitherIsBetter); | |
| } | |
| private: | |
| Clock* const clock_; | |
| const std::string test_label_; | |
| const int64_t creation_time_ms_; | |
| int64_t first_time_in_sync_ = -1; | |
| VideoReceiveStreamInterface* receive_stream_ = nullptr; | |
| SamplesStatsCounter sync_offset_ms_list_; | |
| TaskQueueBase* const task_queue_; | |
| }; | |
| void CallPerfTest::TestAudioVideoSync(FecMode fec, | |
| CreateOrder create_first, | |
| float video_ntp_speed, | |
| float video_rtp_speed, | |
| float audio_rtp_speed, | |
| absl::string_view test_label) { | |
| const char* kSyncGroup = "av_sync"; | |
| const uint32_t kAudioSendSsrc = 1234; | |
| BuiltInNetworkBehaviorConfig audio_net_config; | |
| audio_net_config.queue_delay_ms = 500; | |
| audio_net_config.loss_percent = 5; | |
| auto observer = std::make_unique<VideoRtcpAndSyncObserver>( | |
| task_queue(), Clock::GetRealTimeClock(), test_label); | |
| std::map<uint8_t, MediaType> audio_pt_map; | |
| std::map<uint8_t, MediaType> video_pt_map; | |
| std::unique_ptr<test::PacketTransport> audio_send_transport; | |
| std::unique_ptr<test::PacketTransport> video_send_transport; | |
| std::unique_ptr<test::PacketTransport> receive_transport; | |
| AudioSendStream* audio_send_stream; | |
| AudioReceiveStreamInterface* audio_receive_stream; | |
| std::unique_ptr<DriftingClock> drifting_clock; | |
| SendTask(task_queue(), [&]() { | |
| metrics::Reset(); | |
| scoped_refptr<AudioDeviceModule> fake_audio_device = | |
| TestAudioDeviceModule::Create( | |
| env(), TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000), | |
| TestAudioDeviceModule::CreateDiscardRenderer(48000), | |
| audio_rtp_speed); | |
| EXPECT_EQ(0, fake_audio_device->Init()); | |
| AudioState::Config send_audio_state_config; | |
| send_audio_state_config.audio_mixer = AudioMixerImpl::Create(); | |
| send_audio_state_config.audio_processing = | |
| BuiltinAudioProcessingBuilder().Build(env()); | |
| send_audio_state_config.audio_device_module = fake_audio_device; | |
| CallConfig sender_config = SendCallConfig(); | |
| auto audio_state = AudioState::Create(send_audio_state_config); | |
| fake_audio_device->RegisterAudioCallback(audio_state->audio_transport()); | |
| sender_config.audio_state = audio_state; | |
| CallConfig receiver_config = RecvCallConfig(); | |
| receiver_config.audio_state = audio_state; | |
| CreateCalls(std::move(sender_config), std::move(receiver_config)); | |
| std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_), | |
| std::inserter(audio_pt_map, audio_pt_map.end()), | |
| [](const std::pair<const uint8_t, MediaType>& pair) { | |
| return pair.second == MediaType::AUDIO; | |
| }); | |
| std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_), | |
| std::inserter(video_pt_map, video_pt_map.end()), | |
| [](const std::pair<const uint8_t, MediaType>& pair) { | |
| return pair.second == MediaType::VIDEO; | |
| }); | |
| audio_send_transport = std::make_unique<test::PacketTransport>( | |
| send_env_, task_queue(), sender_call_.get(), observer.get(), | |
| test::PacketTransport::kSender, audio_pt_map, | |
| std::make_unique<FakeNetworkPipe>( | |
| Clock::GetRealTimeClock(), | |
| std::make_unique<SimulatedNetwork>(audio_net_config)), | |
| GetRegisteredExtensions(), GetRegisteredExtensions()); | |
| audio_send_transport->SetReceiver(receiver_call_->Receiver()); | |
| video_send_transport = std::make_unique<test::PacketTransport>( | |
| send_env_, task_queue(), sender_call_.get(), observer.get(), | |
| test::PacketTransport::kSender, video_pt_map, | |
| std::make_unique<FakeNetworkPipe>( | |
| Clock::GetRealTimeClock(), | |
| std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())), | |
| GetRegisteredExtensions(), GetRegisteredExtensions()); | |
| video_send_transport->SetReceiver(receiver_call_->Receiver()); | |
| receive_transport = std::make_unique<test::PacketTransport>( | |
| recv_env_, task_queue(), receiver_call_.get(), observer.get(), | |
| test::PacketTransport::kReceiver, payload_type_map_, | |
| std::make_unique<FakeNetworkPipe>( | |
| Clock::GetRealTimeClock(), | |
| std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())), | |
| GetRegisteredExtensions(), GetRegisteredExtensions()); | |
| receive_transport->SetReceiver(sender_call_->Receiver()); | |
| CreateSendConfig(1, 0, 0, video_send_transport.get()); | |
| CreateMatchingReceiveConfigs(receive_transport.get()); | |
| AudioSendStream::Config audio_send_config(audio_send_transport.get()); | |
| audio_send_config.rtp.ssrc = kAudioSendSsrc; | |
| // TODO(bugs.webrtc.org/14683): Let the tests fail with invalid config. | |
| audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec( | |
| test::VideoTestConstants::kAudioSendPayloadType, {"OPUS", 48000, 2}); | |
| audio_send_config.min_bitrate_bps = 6000; | |
| audio_send_config.max_bitrate_bps = 510000; | |
| audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory(); | |
| audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config); | |
| GetVideoSendConfig()->rtp.nack.rtp_history_ms = | |
| test::VideoTestConstants::kNackRtpHistoryMs; | |
| if (fec == FecMode::kOn) { | |
| GetVideoSendConfig()->rtp.ulpfec.red_payload_type = | |
| test::VideoTestConstants::kRedPayloadType; | |
| GetVideoSendConfig()->rtp.ulpfec.ulpfec_payload_type = | |
| test::VideoTestConstants::kUlpfecPayloadType; | |
| video_receive_configs_[0].rtp.red_payload_type = | |
| test::VideoTestConstants::kRedPayloadType; | |
| video_receive_configs_[0].rtp.ulpfec_payload_type = | |
| test::VideoTestConstants::kUlpfecPayloadType; | |
| } | |
| video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000; | |
| video_receive_configs_[0].renderer = observer.get(); | |
| video_receive_configs_[0].sync_group = kSyncGroup; | |
| AudioReceiveStreamInterface::Config audio_recv_config; | |
| audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc; | |
| audio_recv_config.rtcp_send_transport = receive_transport.get(); | |
| audio_recv_config.sync_group = kSyncGroup; | |
| audio_recv_config.decoder_factory = audio_decoder_factory_; | |
| audio_recv_config.decoder_map = { | |
| {test::VideoTestConstants::kAudioSendPayloadType, {"OPUS", 48000, 2}}}; | |
| if (create_first == CreateOrder::kAudioFirst) { | |
| audio_receive_stream = | |
| receiver_call_->CreateAudioReceiveStream(audio_recv_config); | |
| CreateVideoStreams(); | |
| } else { | |
| CreateVideoStreams(); | |
| audio_receive_stream = | |
| receiver_call_->CreateAudioReceiveStream(audio_recv_config); | |
| } | |
| EXPECT_EQ(1u, video_receive_streams_.size()); | |
| observer->set_receive_stream(video_receive_streams_[0]); | |
| drifting_clock = | |
| std::make_unique<DriftingClock>(&env().clock(), video_ntp_speed); | |
| CreateFrameGeneratorCapturerWithDrift( | |
| drifting_clock.get(), video_rtp_speed, | |
| test::VideoTestConstants::kDefaultFramerate, | |
| test::VideoTestConstants::kDefaultWidth, | |
| test::VideoTestConstants::kDefaultHeight); | |
| Start(); | |
| audio_send_stream->Start(); | |
| audio_receive_stream->Start(); | |
| }); | |
| EXPECT_TRUE(observer->Wait()) | |
| << "Timed out while waiting for audio and video to be synchronized."; | |
| SendTask(task_queue(), [&]() { | |
| // Clear the pointer to the receive stream since it will now be deleted. | |
| observer->set_receive_stream(nullptr); | |
| audio_send_stream->Stop(); | |
| audio_receive_stream->Stop(); | |
| Stop(); | |
| DestroyStreams(); | |
| sender_call_->DestroyAudioSendStream(audio_send_stream); | |
| receiver_call_->DestroyAudioReceiveStream(audio_receive_stream); | |
| DestroyCalls(); | |
| // Call may post periodic rtcp packet to the transport on the process | |
| // thread, thus transport should be destroyed after the call objects. | |
| // Though transports keep pointers to the call objects, transports handle | |
| // packets on the task_queue() and thus wouldn't create a race while current | |
| // destruction happens in the same task as destruction of the call objects. | |
| video_send_transport.reset(); | |
| audio_send_transport.reset(); | |
| receive_transport.reset(); | |
| }); | |
| observer->PrintResults(); | |
| // In quick test synchronization may not be achieved in time. | |
| if (!absl::GetFlag(FLAGS_webrtc_quick_perf_test)) { | |
| // TODO(bugs.webrtc.org/10417): Reenable this for iOS | |
| #if !defined(WEBRTC_IOS) | |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs")); | |
| #endif | |
| } | |
| task_queue()->PostTask( | |
| [to_delete = observer.release()]() { delete to_delete; }); | |
| } | |
| TEST_F(CallPerfTest, Synchronization_PlaysOutAudioAndVideoWithoutClockDrift) { | |
| TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst, | |
| DriftingClock::kNoDrift, DriftingClock::kNoDrift, | |
| DriftingClock::kNoDrift, "_video_no_drift"); | |
| } | |
| TEST_F(CallPerfTest, Synchronization_PlaysOutAudioAndVideoWithVideoNtpDrift) { | |
| TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst, | |
| DriftingClock::PercentsFaster(10.0f), | |
| DriftingClock::kNoDrift, DriftingClock::kNoDrift, | |
| "_video_ntp_drift"); | |
| } | |
| TEST_F(CallPerfTest, | |
| Synchronization_PlaysOutAudioAndVideoWithAudioFasterThanVideoDrift) { | |
| TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst, | |
| DriftingClock::kNoDrift, | |
| DriftingClock::PercentsSlower(30.0f), | |
| DriftingClock::PercentsFaster(30.0f), "_audio_faster"); | |
| } | |
| TEST_F(CallPerfTest, | |
| Synchronization_PlaysOutAudioAndVideoWithVideoFasterThanAudioDrift) { | |
| TestAudioVideoSync(FecMode::kOn, CreateOrder::kVideoFirst, | |
| DriftingClock::kNoDrift, | |
| DriftingClock::PercentsFaster(30.0f), | |
| DriftingClock::PercentsSlower(30.0f), "_video_faster"); | |
| } | |
| TEST_F(CallPerfTest, ReceivesCpuOveruseAndUnderuse) { | |
| // Minimal normal usage at the start, then 30s overuse to allow filter to | |
| // settle, and then 80s underuse to allow plenty of time for rampup again. | |
| field_trials().Set("WebRTC-ForceSimulatedOveruseIntervalMs", "1-30000-80000"); | |
| class LoadObserver : public test::SendTest, | |
| public test::FrameGeneratorCapturer::SinkWantsObserver { | |
| public: | |
| LoadObserver() | |
| : SendTest(test::VideoTestConstants::kLongTimeout), | |
| test_phase_(TestPhase::kInit) {} | |
| void OnFrameGeneratorCapturerCreated( | |
| test::FrameGeneratorCapturer* frame_generator_capturer) override { | |
| frame_generator_capturer->SetSinkWantsObserver(this); | |
| // Set a high initial resolution to be sure that we can scale down. | |
| frame_generator_capturer->ChangeResolution(1920, 1080); | |
| } | |
| // OnSinkWantsChanged is called when FrameGeneratorCapturer::AddOrUpdateSink | |
| // is called. | |
| // TODO(sprang): Add integration test for maintain-framerate mode? | |
| void OnSinkWantsChanged(VideoSinkInterface<VideoFrame>* /* sink */, | |
| const VideoSinkWants& wants) override { | |
| RTC_LOG(LS_INFO) << "OnSinkWantsChanged fps:" << wants.max_framerate_fps | |
| << " max_pixel_count " << wants.max_pixel_count | |
| << " target_pixel_count" | |
| << wants.target_pixel_count.value_or(-1); | |
| // The sink wants can change either because an adaptation happened | |
| // (i.e. the pixels or frame rate changed) or for other reasons, such | |
| // as encoded resolutions being communicated (happens whenever we | |
| // capture a new frame size). In this test, we only care about | |
| // adaptations. | |
| bool did_adapt = | |
| last_wants_.max_pixel_count != wants.max_pixel_count || | |
| last_wants_.target_pixel_count != wants.target_pixel_count || | |
| last_wants_.max_framerate_fps != wants.max_framerate_fps; | |
| last_wants_ = wants; | |
| if (!did_adapt) { | |
| if (test_phase_ == TestPhase::kInit) { | |
| test_phase_ = TestPhase::kStart; | |
| } | |
| return; | |
| } | |
| // At kStart expect CPU overuse. Then expect CPU underuse when the encoder | |
| // delay has been decreased. | |
| switch (test_phase_) { | |
| case TestPhase::kInit: | |
| ADD_FAILURE() << "Got unexpected adaptation request, max res = " | |
| << wants.max_pixel_count << ", target res = " | |
| << wants.target_pixel_count.value_or(-1) | |
| << ", max fps = " << wants.max_framerate_fps; | |
| break; | |
| case TestPhase::kStart: | |
| if (wants.max_pixel_count < std::numeric_limits<int>::max()) { | |
| // On adapting down, VideoStreamEncoder::VideoSourceProxy will set | |
| // only the max pixel count, leaving the target unset. | |
| test_phase_ = TestPhase::kAdaptedDown; | |
| } else { | |
| ADD_FAILURE() << "Got unexpected adaptation request, max res = " | |
| << wants.max_pixel_count << ", target res = " | |
| << wants.target_pixel_count.value_or(-1) | |
| << ", max fps = " << wants.max_framerate_fps; | |
| } | |
| break; | |
| case TestPhase::kAdaptedDown: | |
| // On adapting up, the adaptation counter will again be at zero, and | |
| // so all constraints will be reset. | |
| if (wants.max_pixel_count == std::numeric_limits<int>::max() && | |
| !wants.target_pixel_count) { | |
| test_phase_ = TestPhase::kAdaptedUp; | |
| observation_complete_.Set(); | |
| } else { | |
| ADD_FAILURE() << "Got unexpected adaptation request, max res = " | |
| << wants.max_pixel_count << ", target res = " | |
| << wants.target_pixel_count.value_or(-1) | |
| << ", max fps = " << wants.max_framerate_fps; | |
| } | |
| break; | |
| case TestPhase::kAdaptedUp: | |
| ADD_FAILURE() << "Got unexpected adaptation request, max res = " | |
| << wants.max_pixel_count << ", target res = " | |
| << wants.target_pixel_count.value_or(-1) | |
| << ", max fps = " << wants.max_framerate_fps; | |
| } | |
| } | |
| void ModifyVideoConfigs( | |
| VideoSendStream::Config* /* send_config */, | |
| std::vector<VideoReceiveStreamInterface::Config>* /* receive_configs */, | |
| VideoEncoderConfig* /* encoder_config */) override {} | |
| void PerformTest() override { | |
| EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback."; | |
| } | |
| enum class TestPhase { | |
| kInit, | |
| kStart, | |
| kAdaptedDown, | |
| kAdaptedUp | |
| } test_phase_; | |
| private: | |
| VideoSinkWants last_wants_; | |
| } test; | |
| RunBaseTest(&test); | |
| } | |
| void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) { | |
| static const int kMaxEncodeBitrateKbps = 30; | |
| static const int kMinTransmitBitrateBps = 150000; | |
| static const int kMinAcceptableTransmitBitrate = 130; | |
| static const int kMaxAcceptableTransmitBitrate = 170; | |
| static const int kNumBitrateObservationsInRange = 100; | |
| static const int kAcceptableBitrateErrorMargin = 15; // +- 7 | |
| class BitrateObserver : public test::EndToEndTest { | |
| public: | |
| explicit BitrateObserver(const Environment& env, | |
| bool using_min_transmit_bitrate, | |
| TaskQueueBase* task_queue) | |
| : EndToEndTest(test::VideoTestConstants::kLongTimeout), | |
| env_(env), | |
| send_stream_(nullptr), | |
| converged_(false), | |
| pad_to_min_bitrate_(using_min_transmit_bitrate), | |
| min_acceptable_bitrate_(using_min_transmit_bitrate | |
| ? kMinAcceptableTransmitBitrate | |
| : (kMaxEncodeBitrateKbps - | |
| kAcceptableBitrateErrorMargin / 2)), | |
| max_acceptable_bitrate_(using_min_transmit_bitrate | |
| ? kMaxAcceptableTransmitBitrate | |
| : (kMaxEncodeBitrateKbps + | |
| kAcceptableBitrateErrorMargin / 2)), | |
| num_bitrate_observations_in_range_(0), | |
| task_queue_(task_queue), | |
| task_safety_flag_(PendingTaskSafetyFlag::CreateDetached()) {} | |
| private: | |
| // TODO(holmer): Run this with a timer instead of once per packet. | |
| Action OnSendRtp(ArrayView<const uint8_t> /* packet */) override { | |
| task_queue_->PostTask(SafeTask(task_safety_flag_, [this]() { | |
| VideoSendStream::Stats stats = send_stream_->GetStats(); | |
| if (!stats.substreams.empty()) { | |
| RTC_DCHECK_EQ(1, stats.substreams.size()); | |
| int bitrate_kbps = | |
| stats.substreams.begin()->second.total_bitrate_bps / 1000; | |
| if (bitrate_kbps > min_acceptable_bitrate_ && | |
| bitrate_kbps < max_acceptable_bitrate_) { | |
| converged_ = true; | |
| ++num_bitrate_observations_in_range_; | |
| if (num_bitrate_observations_in_range_ == | |
| kNumBitrateObservationsInRange) | |
| observation_complete_.Set(); | |
| } | |
| if (converged_) | |
| bitrate_kbps_list_.AddSample( | |
| {.value = static_cast<double>(bitrate_kbps), | |
| .time = env_.clock().CurrentTime()}); | |
| } | |
| })); | |
| return SEND_PACKET; | |
| } | |
| void OnVideoStreamsCreated(VideoSendStream* send_stream, | |
| const std::vector<VideoReceiveStreamInterface*>& | |
| /* receive_streams */) override { | |
| send_stream_ = send_stream; | |
| } | |
| void OnStreamsStopped() override { task_safety_flag_->SetNotAlive(); } | |
| void ModifyVideoConfigs( | |
| VideoSendStream::Config* /* send_config */, | |
| std::vector<VideoReceiveStreamInterface::Config>* /* receive_configs */, | |
| VideoEncoderConfig* encoder_config) override { | |
| if (pad_to_min_bitrate_) { | |
| encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps; | |
| } else { | |
| RTC_DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps); | |
| } | |
| } | |
| void PerformTest() override { | |
| EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats."; | |
| GetGlobalMetricsLogger()->LogMetric( | |
| std::string("bitrate_stats_") + | |
| (pad_to_min_bitrate_ ? "min_transmit_bitrate" | |
| : "without_min_transmit_bitrate"), | |
| "bitrate_kbps", bitrate_kbps_list_, Unit::kUnitless, | |
| ImprovementDirection::kNeitherIsBetter); | |
| } | |
| Environment env_; | |
| VideoSendStream* send_stream_; | |
| bool converged_; | |
| const bool pad_to_min_bitrate_; | |
| const int min_acceptable_bitrate_; | |
| const int max_acceptable_bitrate_; | |
| int num_bitrate_observations_in_range_; | |
| SamplesStatsCounter bitrate_kbps_list_; | |
| TaskQueueBase* task_queue_; | |
| scoped_refptr<PendingTaskSafetyFlag> task_safety_flag_; | |
| } test(env(), pad_to_min_bitrate, task_queue()); | |
| fake_encoder_max_bitrate_ = kMaxEncodeBitrateKbps; | |
| RunBaseTest(&test); | |
| } | |
| TEST_F(CallPerfTest, Bitrate_Kbps_PadsToMinTransmitBitrate) { | |
| TestMinTransmitBitrate(true); | |
| } | |
| TEST_F(CallPerfTest, Bitrate_Kbps_NoPadWithoutMinTransmitBitrate) { | |
| TestMinTransmitBitrate(false); | |
| } | |
| // TODO(bugs.webrtc.org/8878) | |
| #if defined(WEBRTC_MAC) | |
| #define MAYBE_KeepsHighBitrateWhenReconfiguringSender \ | |
| DISABLED_KeepsHighBitrateWhenReconfiguringSender | |
| #else | |
| #define MAYBE_KeepsHighBitrateWhenReconfiguringSender \ | |
| KeepsHighBitrateWhenReconfiguringSender | |
| #endif | |
| TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) { | |
| static const uint32_t kInitialBitrateKbps = 400; | |
| static const uint32_t kInitialBitrateOverheadKpbs = 6; | |
| static const uint32_t kReconfigureThresholdKbps = 600; | |
| class VideoStreamFactory | |
| : public VideoEncoderConfig::VideoStreamFactoryInterface { | |
| public: | |
| VideoStreamFactory() {} | |
| private: | |
| std::vector<VideoStream> CreateEncoderStreams( | |
| const FieldTrialsView& /*field_trials*/, | |
| int frame_width, | |
| int frame_height, | |
| const VideoEncoderConfig& encoder_config) override { | |
| std::vector<VideoStream> streams = | |
| test::CreateVideoStreams(frame_width, frame_height, encoder_config); | |
| streams[0].min_bitrate_bps = 50000; | |
| streams[0].target_bitrate_bps = streams[0].max_bitrate_bps = 2000000; | |
| return streams; | |
| } | |
| }; | |
| class BitrateObserver : public test::EndToEndTest, public test::FakeEncoder { | |
| public: | |
| explicit BitrateObserver(const Environment& env, TaskQueueBase* task_queue) | |
| : EndToEndTest(test::VideoTestConstants::kDefaultTimeout), | |
| FakeEncoder(env), | |
| encoder_inits_(0), | |
| last_set_bitrate_kbps_(0), | |
| send_stream_(nullptr), | |
| frame_generator_(nullptr), | |
| encoder_factory_(this), | |
| bitrate_allocator_factory_( | |
| CreateBuiltinVideoBitrateAllocatorFactory()), | |
| task_queue_(task_queue) {} | |
| int32_t InitEncode(const VideoCodec* config, | |
| const VideoEncoder::Settings& settings) override { | |
| ++encoder_inits_; | |
| if (encoder_inits_ == 1) { | |
| // First time initialization. Frame size is known. | |
| // `expected_bitrate` is affected by bandwidth estimation before the | |
| // first frame arrives to the encoder. | |
| uint32_t expected_bitrate = | |
| last_set_bitrate_kbps_ > 0 | |
| ? last_set_bitrate_kbps_ | |
| : kInitialBitrateKbps - kInitialBitrateOverheadKpbs; | |
| EXPECT_EQ(expected_bitrate, config->startBitrate) | |
| << "Encoder not initialized at expected bitrate."; | |
| EXPECT_EQ(test::VideoTestConstants::kDefaultWidth, config->width); | |
| EXPECT_EQ(test::VideoTestConstants::kDefaultHeight, config->height); | |
| } else if (encoder_inits_ == 2) { | |
| EXPECT_EQ(2 * test::VideoTestConstants::kDefaultWidth, config->width); | |
| EXPECT_EQ(2 * test::VideoTestConstants::kDefaultHeight, config->height); | |
| EXPECT_GE(last_set_bitrate_kbps_, kReconfigureThresholdKbps); | |
| EXPECT_GT(config->startBitrate, kReconfigureThresholdKbps) | |
| << "Encoder reconfigured with bitrate too far away from last set."; | |
| observation_complete_.Set(); | |
| } | |
| return FakeEncoder::InitEncode(config, settings); | |
| } | |
| void SetRates(const RateControlParameters& parameters) override { | |
| last_set_bitrate_kbps_ = parameters.bitrate.get_sum_kbps(); | |
| if (encoder_inits_ == 1 && | |
| parameters.bitrate.get_sum_kbps() > kReconfigureThresholdKbps) { | |
| time_to_reconfigure_.Set(); | |
| } | |
| FakeEncoder::SetRates(parameters); | |
| } | |
| void ModifySenderBitrateConfig( | |
| BitrateConstraints* bitrate_config) override { | |
| bitrate_config->start_bitrate_bps = kInitialBitrateKbps * 1000; | |
| } | |
| void ModifyVideoConfigs( | |
| VideoSendStream::Config* send_config, | |
| std::vector<VideoReceiveStreamInterface::Config>* /* receive_configs */, | |
| VideoEncoderConfig* encoder_config) override { | |
| send_config->encoder_settings.encoder_factory = &encoder_factory_; | |
| send_config->encoder_settings.bitrate_allocator_factory = | |
| bitrate_allocator_factory_.get(); | |
| encoder_config->max_bitrate_bps = 2 * kReconfigureThresholdKbps * 1000; | |
| encoder_config->video_stream_factory = | |
| make_ref_counted<VideoStreamFactory>(); | |
| encoder_config_ = encoder_config->Copy(); | |
| } | |
| void OnVideoStreamsCreated(VideoSendStream* send_stream, | |
| const std::vector<VideoReceiveStreamInterface*>& | |
| /* receive_streams */) override { | |
| send_stream_ = send_stream; | |
| } | |
| void OnFrameGeneratorCapturerCreated( | |
| test::FrameGeneratorCapturer* frame_generator_capturer) override { | |
| frame_generator_ = frame_generator_capturer; | |
| } | |
| void PerformTest() override { | |
| ASSERT_TRUE( | |
| time_to_reconfigure_.Wait(test::VideoTestConstants::kDefaultTimeout)) | |
| << "Timed out before receiving an initial high bitrate."; | |
| frame_generator_->ChangeResolution( | |
| test::VideoTestConstants::kDefaultWidth * 2, | |
| test::VideoTestConstants::kDefaultHeight * 2); | |
| SendTask(task_queue_, [&]() { | |
| send_stream_->ReconfigureVideoEncoder(encoder_config_.Copy()); | |
| }); | |
| EXPECT_TRUE(Wait()) | |
| << "Timed out while waiting for a couple of high bitrate estimates " | |
| "after reconfiguring the send stream."; | |
| } | |
| private: | |
| Event time_to_reconfigure_; | |
| int encoder_inits_; | |
| uint32_t last_set_bitrate_kbps_; | |
| VideoSendStream* send_stream_; | |
| test::FrameGeneratorCapturer* frame_generator_; | |
| test::VideoEncoderProxyFactory encoder_factory_; | |
| std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_; | |
| VideoEncoderConfig encoder_config_; | |
| TaskQueueBase* task_queue_; | |
| } test(env(), task_queue()); | |
| RunBaseTest(&test); | |
| } | |
| // Discovers the minimal supported audio+video bitrate. The test bitrate is | |
| // considered supported if Rtt does not go above 400ms with the network | |
| // contrained to the test bitrate. | |
| // | |
| // |test_bitrate_from test_bitrate_to| bitrate constraint range | |
| // `test_bitrate_step` bitrate constraint update step during the test | |
| // |min_bwe max_bwe| BWE range | |
| // `start_bwe` initial BWE | |
| void CallPerfTest::TestMinAudioVideoBitrate(int test_bitrate_from, | |
| int test_bitrate_to, | |
| int test_bitrate_step, | |
| int min_bwe, | |
| int start_bwe, | |
| int max_bwe) { | |
| static const std::string kAudioTrackId = "audio_track_0"; | |
| static constexpr int kBitrateStabilizationMs = 10000; | |
| static constexpr int kBitrateMeasurements = 10; | |
| static constexpr int kBitrateMeasurementMs = 1000; | |
| static constexpr int kShortDelayMs = 10; | |
| static constexpr int kMinGoodRttMs = 400; | |
| class MinVideoAndAudioBitrateTester : public test::EndToEndTest { | |
| public: | |
| MinVideoAndAudioBitrateTester(int test_bitrate_from, | |
| int test_bitrate_to, | |
| int test_bitrate_step, | |
| int min_bwe, | |
| int start_bwe, | |
| int max_bwe, | |
| TaskQueueBase* task_queue) | |
| : EndToEndTest(), | |
| test_bitrate_from_(test_bitrate_from), | |
| test_bitrate_to_(test_bitrate_to), | |
| test_bitrate_step_(test_bitrate_step), | |
| min_bwe_(min_bwe), | |
| start_bwe_(start_bwe), | |
| max_bwe_(max_bwe), | |
| task_queue_(task_queue) {} | |
| protected: | |
| BuiltInNetworkBehaviorConfig GetFakeNetworkPipeConfig() const { | |
| BuiltInNetworkBehaviorConfig pipe_config; | |
| pipe_config.link_capacity = DataRate::KilobitsPerSec(test_bitrate_from_); | |
| return pipe_config; | |
| } | |
| BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override { | |
| return GetFakeNetworkPipeConfig(); | |
| } | |
| BuiltInNetworkBehaviorConfig GetReceiveTransportConfig() const override { | |
| return GetFakeNetworkPipeConfig(); | |
| } | |
| void OnTransportCreated( | |
| test::PacketTransport* /* to_receiver */, | |
| SimulatedNetworkInterface* sender_network, | |
| test::PacketTransport* /* to_sender */, | |
| SimulatedNetworkInterface* receiver_network) override { | |
| send_simulated_network_ = sender_network; | |
| receive_simulated_network_ = receiver_network; | |
| } | |
| void PerformTest() override { | |
| // Quick test mode, just to exercise all the code paths without actually | |
| // caring about performance measurements. | |
| const bool quick_perf_test = absl::GetFlag(FLAGS_webrtc_quick_perf_test); | |
| int last_passed_test_bitrate = -1; | |
| for (int test_bitrate = test_bitrate_from_; | |
| test_bitrate_from_ < test_bitrate_to_ | |
| ? test_bitrate <= test_bitrate_to_ | |
| : test_bitrate >= test_bitrate_to_; | |
| test_bitrate += test_bitrate_step_) { | |
| BuiltInNetworkBehaviorConfig pipe_config; | |
| pipe_config.link_capacity = DataRate::KilobitsPerSec(test_bitrate); | |
| send_simulated_network_->SetConfig(pipe_config); | |
| receive_simulated_network_->SetConfig(pipe_config); | |
| Thread::SleepMs(quick_perf_test ? kShortDelayMs | |
| : kBitrateStabilizationMs); | |
| int64_t avg_rtt = 0; | |
| for (int i = 0; i < kBitrateMeasurements; i++) { | |
| Call::Stats call_stats; | |
| SendTask(task_queue_, [this, &call_stats]() { | |
| call_stats = sender_call_->GetStats(); | |
| }); | |
| avg_rtt += call_stats.rtt_ms; | |
| Thread::SleepMs(quick_perf_test ? kShortDelayMs | |
| : kBitrateMeasurementMs); | |
| } | |
| avg_rtt = avg_rtt / kBitrateMeasurements; | |
| if (avg_rtt > kMinGoodRttMs) { | |
| RTC_LOG(LS_WARNING) | |
| << "Failed test bitrate: " << test_bitrate << " RTT: " << avg_rtt; | |
| break; | |
| } else { | |
| RTC_LOG(LS_INFO) << "Passed test bitrate: " << test_bitrate | |
| << " RTT: " << avg_rtt; | |
| last_passed_test_bitrate = test_bitrate; | |
| } | |
| } | |
| EXPECT_GT(last_passed_test_bitrate, -1) | |
| << "Minimum supported bitrate out of the test scope"; | |
| GetGlobalMetricsLogger()->LogSingleValueMetric( | |
| "min_test_bitrate_", "min_bitrate", last_passed_test_bitrate, | |
| Unit::kUnitless, ImprovementDirection::kNeitherIsBetter); | |
| } | |
| void OnCallsCreated(Call* sender_call, Call* /* receiver_call */) override { | |
| sender_call_ = sender_call; | |
| BitrateConstraints bitrate_config; | |
| bitrate_config.min_bitrate_bps = min_bwe_; | |
| bitrate_config.start_bitrate_bps = start_bwe_; | |
| bitrate_config.max_bitrate_bps = max_bwe_; | |
| sender_call->GetTransportControllerSend()->SetSdpBitrateParameters( | |
| bitrate_config); | |
| } | |
| size_t GetNumVideoStreams() const override { return 1; } | |
| size_t GetNumAudioStreams() const override { return 1; } | |
| private: | |
| const int test_bitrate_from_; | |
| const int test_bitrate_to_; | |
| const int test_bitrate_step_; | |
| const int min_bwe_; | |
| const int start_bwe_; | |
| const int max_bwe_; | |
| SimulatedNetworkInterface* send_simulated_network_; | |
| SimulatedNetworkInterface* receive_simulated_network_; | |
| Call* sender_call_; | |
| TaskQueueBase* const task_queue_; | |
| } test(test_bitrate_from, test_bitrate_to, test_bitrate_step, min_bwe, | |
| start_bwe, max_bwe, task_queue()); | |
| RunBaseTest(&test); | |
| } | |
| TEST_F(CallPerfTest, Min_Bitrate_VideoAndAudio) { | |
| TestMinAudioVideoBitrate(110, 40, -10, 10000, 70000, 200000); | |
| } | |
| void CallPerfTest::TestEncodeFramerate(VideoEncoderFactory* encoder_factory, | |
| absl::string_view payload_name, | |
| const std::vector<int>& max_framerates) { | |
| static constexpr double kAllowedFpsDiff = 1.5; | |
| static constexpr TimeDelta kMinGetStatsInterval = TimeDelta::Millis(400); | |
| static constexpr TimeDelta kMinRunTime = TimeDelta::Seconds(15); | |
| static constexpr DataRate kMaxBitrate = DataRate::KilobitsPerSec(1000); | |
| class FramerateObserver | |
| : public test::EndToEndTest, | |
| public test::FrameGeneratorCapturer::SinkWantsObserver { | |
| public: | |
| FramerateObserver(VideoEncoderFactory* encoder_factory, | |
| absl::string_view payload_name, | |
| const std::vector<int>& max_framerates, | |
| TaskQueueBase* task_queue) | |
| : EndToEndTest(test::VideoTestConstants::kDefaultTimeout), | |
| clock_(Clock::GetRealTimeClock()), | |
| encoder_factory_(encoder_factory), | |
| payload_name_(payload_name), | |
| max_framerates_(max_framerates), | |
| task_queue_(task_queue), | |
| start_time_(clock_->CurrentTime()), | |
| last_getstats_time_(start_time_), | |
| send_stream_(nullptr) {} | |
| void OnFrameGeneratorCapturerCreated( | |
| test::FrameGeneratorCapturer* frame_generator_capturer) override { | |
| frame_generator_capturer->ChangeResolution(640, 360); | |
| } | |
| void OnSinkWantsChanged(VideoSinkInterface<VideoFrame>* /* sink */, | |
| const VideoSinkWants& /* wants */) override {} | |
| void ModifySenderBitrateConfig( | |
| BitrateConstraints* bitrate_config) override { | |
| bitrate_config->start_bitrate_bps = kMaxBitrate.bps() / 2; | |
| } | |
| void OnVideoStreamsCreated(VideoSendStream* send_stream, | |
| const std::vector<VideoReceiveStreamInterface*>& | |
| /* receive_streams */) override { | |
| send_stream_ = send_stream; | |
| } | |
| size_t GetNumVideoStreams() const override { | |
| return max_framerates_.size(); | |
| } | |
| void ModifyVideoConfigs( | |
| VideoSendStream::Config* send_config, | |
| std::vector<VideoReceiveStreamInterface::Config>* /* receive_configs */, | |
| VideoEncoderConfig* encoder_config) override { | |
| send_config->encoder_settings.encoder_factory = encoder_factory_; | |
| send_config->rtp.payload_name = payload_name_; | |
| send_config->rtp.payload_type = | |
| test::VideoTestConstants::kVideoSendPayloadType; | |
| encoder_config->video_format.name = payload_name_; | |
| encoder_config->codec_type = PayloadStringToCodecType(payload_name_); | |
| encoder_config->max_bitrate_bps = kMaxBitrate.bps(); | |
| for (size_t i = 0; i < max_framerates_.size(); ++i) { | |
| encoder_config->simulcast_layers[i].max_framerate = max_framerates_[i]; | |
| configured_framerates_[send_config->rtp.ssrcs[i]] = max_framerates_[i]; | |
| } | |
| } | |
| void PerformTest() override { | |
| EXPECT_TRUE(Wait()) << "Timeout while waiting for framerate stats."; | |
| } | |
| void VerifyStats() const { | |
| const bool quick_perf_test = absl::GetFlag(FLAGS_webrtc_quick_perf_test); | |
| double input_fps = 0.0; | |
| for (const auto& configured_framerate : configured_framerates_) { | |
| input_fps = std::max(configured_framerate.second, input_fps); | |
| } | |
| for (const auto& encode_frame_rate_list : encode_frame_rate_lists_) { | |
| const SamplesStatsCounter& values = encode_frame_rate_list.second; | |
| GetGlobalMetricsLogger()->LogMetric( | |
| "substream_fps", "encode_frame_rate", values, Unit::kUnitless, | |
| ImprovementDirection::kNeitherIsBetter); | |
| if (values.IsEmpty()) { | |
| continue; | |
| } | |
| double average_fps = values.GetAverage(); | |
| uint32_t ssrc = encode_frame_rate_list.first; | |
| double expected_fps = configured_framerates_.find(ssrc)->second; | |
| if (quick_perf_test && expected_fps != input_fps) | |
| EXPECT_NEAR(expected_fps, average_fps, kAllowedFpsDiff); | |
| } | |
| } | |
| Action OnSendRtp(ArrayView<const uint8_t> /* packet */) override { | |
| const Timestamp now = clock_->CurrentTime(); | |
| if (now - last_getstats_time_ > kMinGetStatsInterval) { | |
| last_getstats_time_ = now; | |
| task_queue_->PostTask([this, now]() { | |
| VideoSendStream::Stats stats = send_stream_->GetStats(); | |
| for (const auto& stat : stats.substreams) { | |
| encode_frame_rate_lists_[stat.first].AddSample( | |
| {.value = stat.second.encode_frame_rate, .time = now}); | |
| } | |
| if (now - start_time_ > kMinRunTime) { | |
| VerifyStats(); | |
| observation_complete_.Set(); | |
| } | |
| }); | |
| } | |
| return SEND_PACKET; | |
| } | |
| Clock* const clock_; | |
| VideoEncoderFactory* const encoder_factory_; | |
| const std::string payload_name_; | |
| const std::vector<int> max_framerates_; | |
| TaskQueueBase* const task_queue_; | |
| const Timestamp start_time_; | |
| Timestamp last_getstats_time_; | |
| VideoSendStream* send_stream_; | |
| std::map<uint32_t, SamplesStatsCounter> encode_frame_rate_lists_; | |
| std::map<uint32_t, double> configured_framerates_; | |
| } test(encoder_factory, payload_name, max_framerates, task_queue()); | |
| RunBaseTest(&test); | |
| } | |
| TEST_F(CallPerfTest, TestEncodeFramerateVp8Simulcast) { | |
| InternalEncoderFactory internal_encoder_factory; | |
| test::FunctionVideoEncoderFactory encoder_factory( | |
| [&internal_encoder_factory](const Environment& env, | |
| const SdpVideoFormat& /* format */) { | |
| return std::make_unique<SimulcastEncoderAdapter>( | |
| env, &internal_encoder_factory, nullptr, SdpVideoFormat::VP8()); | |
| }); | |
| TestEncodeFramerate(&encoder_factory, "VP8", | |
| /*max_framerates=*/{20, 30}); | |
| } | |
| TEST_F(CallPerfTest, TestEncodeFramerateVp8SimulcastLowerInputFps) { | |
| InternalEncoderFactory internal_encoder_factory; | |
| test::FunctionVideoEncoderFactory encoder_factory( | |
| [&internal_encoder_factory](const Environment& env, | |
| const SdpVideoFormat& /* format */) { | |
| return std::make_unique<SimulcastEncoderAdapter>( | |
| env, &internal_encoder_factory, nullptr, SdpVideoFormat::VP8()); | |
| }); | |
| TestEncodeFramerate(&encoder_factory, "VP8", | |
| /*max_framerates=*/{14, 20}); | |
| } | |
| } // namespace webrtc |