| /* |
| * 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 <limits> |
| #include <memory> |
| #include <string> |
| |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "api/test/simulated_network.h" |
| #include "api/video/builtin_video_bitrate_allocator_factory.h" |
| #include "api/video/video_bitrate_allocation.h" |
| #include "api/video_codecs/video_encoder.h" |
| #include "api/video_codecs/video_encoder_config.h" |
| #include "call/call.h" |
| #include "call/fake_network_pipe.h" |
| #include "call/simulated_network.h" |
| #include "modules/audio_coding/include/audio_coding_module.h" |
| #include "modules/audio_device/include/test_audio_device.h" |
| #include "modules/audio_mixer/audio_mixer_impl.h" |
| #include "modules/rtp_rtcp/source/rtp_packet.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/synchronization/mutex.h" |
| #include "rtc_base/task_queue_for_test.h" |
| #include "rtc_base/task_utils/pending_task_safety_flag.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/thread_annotations.h" |
| #include "system_wrappers/include/metrics.h" |
| #include "test/call_test.h" |
| #include "test/direct_transport.h" |
| #include "test/drifting_clock.h" |
| #include "test/encoder_settings.h" |
| #include "test/fake_encoder.h" |
| #include "test/field_trial.h" |
| #include "test/frame_generator_capturer.h" |
| #include "test/gtest.h" |
| #include "test/null_transport.h" |
| #include "test/rtp_rtcp_observer.h" |
| #include "test/testsupport/file_utils.h" |
| #include "test/testsupport/perf_test.h" |
| #include "test/video_encoder_proxy_factory.h" |
| #include "video/transport_adapter.h" |
| |
| using webrtc::test::DriftingClock; |
| |
| namespace webrtc { |
| namespace { |
| 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, |
| const std::string& 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); |
| }; |
| |
| class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver, |
| public rtc::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, |
| const std::string& test_label) |
| : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs), |
| 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(ToQueuedTask([this]() { CheckStats(); })); |
| } |
| |
| void CheckStats() { |
| if (!receive_stream_) |
| return; |
| |
| VideoReceiveStream::Stats stats = receive_stream_->GetStats(); |
| if (stats.sync_offset_ms == std::numeric_limits<int>::max()) |
| return; |
| |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| 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; |
| webrtc::test::PrintResult("sync_convergence_time", test_label_, |
| "synchronization", time_since_creation, "ms", |
| false); |
| } |
| if (time_since_creation > kMinRunTimeMs) |
| observation_complete_.Set(); |
| } |
| if (first_time_in_sync_ != -1) |
| sync_offset_ms_list_.push_back(stats.sync_offset_ms); |
| } |
| |
| void set_receive_stream(VideoReceiveStream* receive_stream) { |
| RTC_DCHECK_EQ(task_queue_, TaskQueueBase::Current()); |
| // Note that receive_stream may be nullptr. |
| receive_stream_ = receive_stream; |
| } |
| |
| void PrintResults() { |
| test::PrintResultList("stream_offset", test_label_, "synchronization", |
| sync_offset_ms_list_, "ms", false); |
| } |
| |
| private: |
| Clock* const clock_; |
| std::string test_label_; |
| const int64_t creation_time_ms_; |
| int64_t first_time_in_sync_ = -1; |
| VideoReceiveStream* receive_stream_ = nullptr; |
| std::vector<double> 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, |
| const std::string& test_label) { |
| const char* kSyncGroup = "av_sync"; |
| const uint32_t kAudioSendSsrc = 1234; |
| const uint32_t kAudioRecvSsrc = 5678; |
| |
| 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; |
| AudioReceiveStream* audio_receive_stream; |
| std::unique_ptr<DriftingClock> drifting_clock; |
| |
| SendTask(RTC_FROM_HERE, task_queue(), [&]() { |
| metrics::Reset(); |
| rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device = |
| TestAudioDeviceModule::Create( |
| task_queue_factory_.get(), |
| 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 = |
| AudioProcessingBuilder().Create(); |
| send_audio_state_config.audio_device_module = fake_audio_device; |
| Call::Config sender_config(send_event_log_.get()); |
| |
| auto audio_state = AudioState::Create(send_audio_state_config); |
| fake_audio_device->RegisterAudioCallback(audio_state->audio_transport()); |
| sender_config.audio_state = audio_state; |
| Call::Config receiver_config(recv_event_log_.get()); |
| receiver_config.audio_state = audio_state; |
| CreateCalls(sender_config, 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>( |
| 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))); |
| audio_send_transport->SetReceiver(receiver_call_->Receiver()); |
| |
| video_send_transport = std::make_unique<test::PacketTransport>( |
| task_queue(), sender_call_.get(), observer.get(), |
| test::PacketTransport::kSender, video_pt_map, |
| std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>( |
| BuiltInNetworkBehaviorConfig()))); |
| video_send_transport->SetReceiver(receiver_call_->Receiver()); |
| |
| receive_transport = std::make_unique<test::PacketTransport>( |
| task_queue(), receiver_call_.get(), observer.get(), |
| test::PacketTransport::kReceiver, payload_type_map_, |
| std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>( |
| BuiltInNetworkBehaviorConfig()))); |
| 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; |
| audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec( |
| kAudioSendPayloadType, {"ISAC", 16000, 1}); |
| audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory(); |
| audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config); |
| |
| GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| if (fec == FecMode::kOn) { |
| GetVideoSendConfig()->rtp.ulpfec.red_payload_type = kRedPayloadType; |
| GetVideoSendConfig()->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType; |
| video_receive_configs_[0].rtp.red_payload_type = kRedPayloadType; |
| video_receive_configs_[0].rtp.ulpfec_payload_type = 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; |
| |
| AudioReceiveStream::Config audio_recv_config; |
| audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc; |
| audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc; |
| 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 = { |
| {kAudioSendPayloadType, {"ISAC", 16000, 1}}}; |
| |
| 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>(clock_, video_ntp_speed); |
| CreateFrameGeneratorCapturerWithDrift(drifting_clock.get(), video_rtp_speed, |
| kDefaultFramerate, kDefaultWidth, |
| 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(RTC_FROM_HERE, 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 (!field_trial::IsEnabled("WebRTC-QuickPerfTest")) { |
| // 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( |
| ToQueuedTask([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"); |
| } |
| |
| void CallPerfTest::TestCaptureNtpTime( |
| const BuiltInNetworkBehaviorConfig& net_config, |
| int threshold_ms, |
| int start_time_ms, |
| int run_time_ms) { |
| class CaptureNtpTimeObserver : public test::EndToEndTest, |
| public rtc::VideoSinkInterface<VideoFrame> { |
| public: |
| CaptureNtpTimeObserver(const BuiltInNetworkBehaviorConfig& net_config, |
| int threshold_ms, |
| int start_time_ms, |
| int run_time_ms) |
| : EndToEndTest(kLongTimeoutMs), |
| net_config_(net_config), |
| clock_(Clock::GetRealTimeClock()), |
| threshold_ms_(threshold_ms), |
| start_time_ms_(start_time_ms), |
| run_time_ms_(run_time_ms), |
| creation_time_ms_(clock_->TimeInMilliseconds()), |
| capturer_(nullptr), |
| rtp_start_timestamp_set_(false), |
| rtp_start_timestamp_(0) {} |
| |
| private: |
| std::unique_ptr<test::PacketTransport> CreateSendTransport( |
| TaskQueueBase* task_queue, |
| Call* sender_call) override { |
| return std::make_unique<test::PacketTransport>( |
| task_queue, sender_call, this, test::PacketTransport::kSender, |
| payload_type_map_, |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>(net_config_))); |
| } |
| |
| std::unique_ptr<test::PacketTransport> CreateReceiveTransport( |
| TaskQueueBase* task_queue) override { |
| return std::make_unique<test::PacketTransport>( |
| task_queue, nullptr, this, test::PacketTransport::kReceiver, |
| payload_type_map_, |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>(net_config_))); |
| } |
| |
| void OnFrame(const VideoFrame& video_frame) override { |
| MutexLock lock(&mutex_); |
| if (video_frame.ntp_time_ms() <= 0) { |
| // Haven't got enough RTCP SR in order to calculate the capture ntp |
| // time. |
| return; |
| } |
| |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| int64_t time_since_creation = now_ms - creation_time_ms_; |
| if (time_since_creation < start_time_ms_) { |
| // Wait for `start_time_ms_` before start measuring. |
| return; |
| } |
| |
| if (time_since_creation > run_time_ms_) { |
| observation_complete_.Set(); |
| } |
| |
| FrameCaptureTimeList::iterator iter = |
| capture_time_list_.find(video_frame.timestamp()); |
| EXPECT_TRUE(iter != capture_time_list_.end()); |
| |
| // The real capture time has been wrapped to uint32_t before converted |
| // to rtp timestamp in the sender side. So here we convert the estimated |
| // capture time to a uint32_t 90k timestamp also for comparing. |
| uint32_t estimated_capture_timestamp = |
| 90 * static_cast<uint32_t>(video_frame.ntp_time_ms()); |
| uint32_t real_capture_timestamp = iter->second; |
| int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp; |
| time_offset_ms = time_offset_ms / 90; |
| time_offset_ms_list_.push_back(time_offset_ms); |
| |
| EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_); |
| } |
| |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| MutexLock lock(&mutex_); |
| RtpPacket rtp_packet; |
| EXPECT_TRUE(rtp_packet.Parse(packet, length)); |
| |
| if (!rtp_start_timestamp_set_) { |
| // Calculate the rtp timestamp offset in order to calculate the real |
| // capture time. |
| uint32_t first_capture_timestamp = |
| 90 * static_cast<uint32_t>(capturer_->first_frame_capture_time()); |
| rtp_start_timestamp_ = rtp_packet.Timestamp() - first_capture_timestamp; |
| rtp_start_timestamp_set_ = true; |
| } |
| |
| uint32_t capture_timestamp = |
| rtp_packet.Timestamp() - rtp_start_timestamp_; |
| capture_time_list_.insert( |
| capture_time_list_.end(), |
| std::make_pair(rtp_packet.Timestamp(), capture_timestamp)); |
| return SEND_PACKET; |
| } |
| |
| void OnFrameGeneratorCapturerCreated( |
| test::FrameGeneratorCapturer* frame_generator_capturer) override { |
| capturer_ = frame_generator_capturer; |
| } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStream::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| (*receive_configs)[0].renderer = this; |
| // Enable the receiver side rtt calculation. |
| (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true; |
| } |
| |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) << "Timed out while waiting for " |
| "estimated capture NTP time to be " |
| "within bounds."; |
| test::PrintResultList("capture_ntp_time", "", "real - estimated", |
| time_offset_ms_list_, "ms", true); |
| } |
| |
| Mutex mutex_; |
| const BuiltInNetworkBehaviorConfig net_config_; |
| Clock* const clock_; |
| int threshold_ms_; |
| int start_time_ms_; |
| int run_time_ms_; |
| int64_t creation_time_ms_; |
| test::FrameGeneratorCapturer* capturer_; |
| bool rtp_start_timestamp_set_; |
| uint32_t rtp_start_timestamp_; |
| typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList; |
| FrameCaptureTimeList capture_time_list_ RTC_GUARDED_BY(&mutex_); |
| std::vector<double> time_offset_ms_list_; |
| } test(net_config, threshold_ms, start_time_ms, run_time_ms); |
| |
| RunBaseTest(&test); |
| } |
| |
| // Flaky tests, disabled on Mac and Windows due to webrtc:8291. |
| #if !(defined(WEBRTC_MAC) || defined(WEBRTC_WIN)) |
| TEST_F(CallPerfTest, Real_Estimated_CaptureNtpTimeWithNetworkDelay) { |
| BuiltInNetworkBehaviorConfig net_config; |
| net_config.queue_delay_ms = 100; |
| // TODO(wu): lower the threshold as the calculation/estimatation becomes more |
| // accurate. |
| const int kThresholdMs = 100; |
| const int kStartTimeMs = 10000; |
| const int kRunTimeMs = 20000; |
| TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs); |
| } |
| |
| TEST_F(CallPerfTest, Real_Estimated_CaptureNtpTimeWithNetworkJitter) { |
| BuiltInNetworkBehaviorConfig net_config; |
| net_config.queue_delay_ms = 100; |
| net_config.delay_standard_deviation_ms = 10; |
| // TODO(wu): lower the threshold as the calculation/estimatation becomes more |
| // accurate. |
| const int kThresholdMs = 100; |
| const int kStartTimeMs = 10000; |
| const int kRunTimeMs = 20000; |
| TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs); |
| } |
| #endif |
| |
| 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. |
| test::ScopedFieldTrials fake_overuse_settings( |
| "WebRTC-ForceSimulatedOveruseIntervalMs/1-30000-80000/"); |
| |
| class LoadObserver : public test::SendTest, |
| public test::FrameGeneratorCapturer::SinkWantsObserver { |
| public: |
| LoadObserver() : SendTest(kLongTimeoutMs), 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(rtc::VideoSinkInterface<VideoFrame>* sink, |
| const rtc::VideoSinkWants& wants) override { |
| // 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) { |
| return; |
| } |
| // At kStart expect CPU overuse. Then expect CPU underuse when the encoder |
| // delay has been decreased. |
| switch (test_phase_) { |
| case TestPhase::kInit: |
| // Max framerate should be set initially. |
| if (wants.max_framerate_fps != std::numeric_limits<int>::max() && |
| wants.max_pixel_count == std::numeric_limits<int>::max()) { |
| test_phase_ = TestPhase::kStart; |
| } 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::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<VideoReceiveStream::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: |
| rtc::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(bool using_min_transmit_bitrate, |
| TaskQueueBase* task_queue) |
| : EndToEndTest(kLongTimeoutMs), |
| 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(const uint8_t* packet, size_t length) override { |
| task_queue_->PostTask(ToQueuedTask(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_.push_back(bitrate_kbps); |
| } |
| })); |
| return SEND_PACKET; |
| } |
| |
| void OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& receive_streams) override { |
| send_stream_ = send_stream; |
| } |
| |
| void OnStreamsStopped() override { task_safety_flag_->SetNotAlive(); } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStream::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."; |
| test::PrintResultList( |
| "bitrate_stats_", |
| (pad_to_min_bitrate_ ? "min_transmit_bitrate" |
| : "without_min_transmit_bitrate"), |
| "bitrate_kbps", bitrate_kbps_list_, "kbps", false); |
| } |
| |
| 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_; |
| std::vector<double> bitrate_kbps_list_; |
| TaskQueueBase* task_queue_; |
| rtc::scoped_refptr<PendingTaskSafetyFlag> task_safety_flag_; |
| } test(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 kReconfigureThresholdKbps = 600; |
| |
| // We get lower bitrate than expected by this test if the following field |
| // trial is enabled. |
| test::ScopedFieldTrials field_trials( |
| "WebRTC-SendSideBwe-WithOverhead/Disabled/"); |
| |
| class VideoStreamFactory |
| : public VideoEncoderConfig::VideoStreamFactoryInterface { |
| public: |
| VideoStreamFactory() {} |
| |
| private: |
| std::vector<VideoStream> CreateEncoderStreams( |
| int width, |
| int height, |
| const VideoEncoderConfig& encoder_config) override { |
| std::vector<VideoStream> streams = |
| test::CreateVideoStreams(width, 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(TaskQueueBase* task_queue) |
| : EndToEndTest(kDefaultTimeoutMs), |
| FakeEncoder(Clock::GetRealTimeClock()), |
| 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; |
| EXPECT_EQ(expected_bitrate, config->startBitrate) |
| << "Encoder not initialized at expected bitrate."; |
| EXPECT_EQ(kDefaultWidth, config->width); |
| EXPECT_EQ(kDefaultHeight, config->height); |
| } else if (encoder_inits_ == 2) { |
| EXPECT_EQ(2 * kDefaultWidth, config->width); |
| EXPECT_EQ(2 * 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<VideoReceiveStream::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 = |
| rtc::make_ref_counted<VideoStreamFactory>(); |
| |
| encoder_config_ = encoder_config->Copy(); |
| } |
| |
| void OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& 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(kDefaultTimeoutMs)) |
| << "Timed out before receiving an initial high bitrate."; |
| frame_generator_->ChangeResolution(kDefaultWidth * 2, kDefaultHeight * 2); |
| SendTask(RTC_FROM_HERE, 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: |
| rtc::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(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 kOpusBitrateFbBps = 32000; |
| 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() { |
| BuiltInNetworkBehaviorConfig pipe_config; |
| pipe_config.link_capacity_kbps = test_bitrate_from_; |
| return pipe_config; |
| } |
| |
| std::unique_ptr<test::PacketTransport> CreateSendTransport( |
| TaskQueueBase* task_queue, |
| Call* sender_call) override { |
| auto network = |
| std::make_unique<SimulatedNetwork>(GetFakeNetworkPipeConfig()); |
| send_simulated_network_ = network.get(); |
| return std::make_unique<test::PacketTransport>( |
| task_queue, sender_call, this, test::PacketTransport::kSender, |
| test::CallTest::payload_type_map_, |
| std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(), |
| std::move(network))); |
| } |
| |
| std::unique_ptr<test::PacketTransport> CreateReceiveTransport( |
| TaskQueueBase* task_queue) override { |
| auto network = |
| std::make_unique<SimulatedNetwork>(GetFakeNetworkPipeConfig()); |
| receive_simulated_network_ = network.get(); |
| return std::make_unique<test::PacketTransport>( |
| task_queue, nullptr, this, test::PacketTransport::kReceiver, |
| test::CallTest::payload_type_map_, |
| std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(), |
| std::move(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 = |
| field_trial::IsEnabled("WebRTC-QuickPerfTest"); |
| 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_kbps = test_bitrate; |
| send_simulated_network_->SetConfig(pipe_config); |
| receive_simulated_network_->SetConfig(pipe_config); |
| |
| rtc::Thread::SleepMs(quick_perf_test ? kShortDelayMs |
| : kBitrateStabilizationMs); |
| |
| int64_t avg_rtt = 0; |
| for (int i = 0; i < kBitrateMeasurements; i++) { |
| Call::Stats call_stats; |
| SendTask(RTC_FROM_HERE, task_queue_, [this, &call_stats]() { |
| call_stats = sender_call_->GetStats(); |
| }); |
| avg_rtt += call_stats.rtt_ms; |
| rtc::Thread::SleepMs(quick_perf_test ? kShortDelayMs |
| : kBitrateMeasurementMs); |
| } |
| avg_rtt = avg_rtt / kBitrateMeasurements; |
| if (avg_rtt > kMinGoodRttMs) { |
| break; |
| } else { |
| last_passed_test_bitrate = test_bitrate; |
| } |
| } |
| EXPECT_GT(last_passed_test_bitrate, -1) |
| << "Minimum supported bitrate out of the test scope"; |
| webrtc::test::PrintResult("min_test_bitrate_", "", "min_bitrate", |
| last_passed_test_bitrate, "kbps", false); |
| } |
| |
| 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; } |
| |
| void ModifyAudioConfigs( |
| AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStream::Config>* receive_configs) override { |
| send_config->send_codec_spec->target_bitrate_bps = |
| absl::optional<int>(kOpusBitrateFbBps); |
| } |
| |
| 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_; |
| SimulatedNetwork* send_simulated_network_; |
| SimulatedNetwork* 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); |
| } |
| |
| // TODO(bugs.webrtc.org/8878) |
| #if defined(WEBRTC_MAC) |
| #define MAYBE_Min_Bitrate_VideoAndAudio DISABLED_Min_Bitrate_VideoAndAudio |
| #else |
| #define MAYBE_Min_Bitrate_VideoAndAudio Min_Bitrate_VideoAndAudio |
| #endif |
| TEST_F(CallPerfTest, MAYBE_Min_Bitrate_VideoAndAudio) { |
| TestMinAudioVideoBitrate(110, 40, -10, 10000, 70000, 200000); |
| } |
| |
| } // namespace webrtc |
