diff options
Diffstat (limited to 'third_party/libwebrtc/webrtc/call/call_perf_tests.cc')
| -rw-r--r-- | third_party/libwebrtc/webrtc/call/call_perf_tests.cc | 960 |
1 files changed, 960 insertions, 0 deletions
diff --git a/third_party/libwebrtc/webrtc/call/call_perf_tests.cc b/third_party/libwebrtc/webrtc/call/call_perf_tests.cc new file mode 100644 index 0000000000..90eaa5a4d5 --- /dev/null +++ b/third_party/libwebrtc/webrtc/call/call_perf_tests.cc @@ -0,0 +1,960 @@ +/* + * 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 "call/call.h" +#include "call/video_config.h" +#include "logging/rtc_event_log/rtc_event_log.h" +#include "modules/audio_coding/include/audio_coding_module.h" +#include "modules/audio_mixer/audio_mixer_impl.h" +#include "modules/rtp_rtcp/include/rtp_header_parser.h" +#include "rtc_base/bitrateallocationstrategy.h" +#include "rtc_base/checks.h" +#include "rtc_base/ptr_util.h" +#include "rtc_base/thread_annotations.h" +#include "system_wrappers/include/metrics_default.h" +#include "test/call_test.h" +#include "test/direct_transport.h" +#include "test/drifting_clock.h" +#include "test/encoder_settings.h" +#include "test/fake_audio_device.h" +#include "test/fake_encoder.h" +#include "test/field_trial.h" +#include "test/frame_generator.h" +#include "test/frame_generator_capturer.h" +#include "test/gtest.h" +#include "test/rtp_rtcp_observer.h" +#include "test/single_threaded_task_queue.h" +#include "test/testsupport/fileutils.h" +#include "test/testsupport/perf_test.h" +#include "video/transport_adapter.h" +#include "voice_engine/include/voe_base.h" + +using webrtc::test::DriftingClock; +using webrtc::test::FakeAudioDevice; + +namespace webrtc { + +class CallPerfTest : public test::CallTest { + 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); + + void TestMinTransmitBitrate(bool pad_to_min_bitrate); + + void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, + int threshold_ms, + int start_time_ms, + int run_time_ms); + void TestMinAudioVideoBitrate(bool use_bitrate_allocation_strategy, + 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(Clock* clock) + : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs), + clock_(clock), + creation_time_ms_(clock_->TimeInMilliseconds()), + first_time_in_sync_(-1), + receive_stream_(nullptr) {} + + void OnFrame(const VideoFrame& video_frame) override { + VideoReceiveStream::Stats stats; + { + rtc::CritScope lock(&crit_); + if (receive_stream_) + 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", + "", + "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::CritScope lock(&crit_); + receive_stream_ = receive_stream; + } + + void PrintResults() { + test::PrintResultList("stream_offset", "", "synchronization", + sync_offset_ms_list_, "ms", false); + } + + private: + Clock* const clock_; + const int64_t creation_time_ms_; + int64_t first_time_in_sync_; + rtc::CriticalSection crit_; + VideoReceiveStream* receive_stream_ RTC_GUARDED_BY(crit_); + std::vector<double> sync_offset_ms_list_; +}; + +void CallPerfTest::TestAudioVideoSync(FecMode fec, + CreateOrder create_first, + float video_ntp_speed, + float video_rtp_speed, + float audio_rtp_speed) { + const char* kSyncGroup = "av_sync"; + const uint32_t kAudioSendSsrc = 1234; + const uint32_t kAudioRecvSsrc = 5678; + + int send_channel_id; + int recv_channel_id; + + FakeNetworkPipe::Config audio_net_config; + audio_net_config.queue_delay_ms = 500; + audio_net_config.loss_percent = 5; + + rtc::scoped_refptr<AudioProcessing> audio_processing; + VoiceEngine* voice_engine; + VoEBase* voe_base; + std::unique_ptr<FakeAudioDevice> fake_audio_device; + VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock()); + + 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; + + task_queue_.SendTask([&]() { + metrics::Reset(); + audio_processing = AudioProcessing::Create(); + voice_engine = VoiceEngine::Create(); + voe_base = VoEBase::GetInterface(voice_engine); + fake_audio_device = rtc::MakeUnique<FakeAudioDevice>( + FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000), + FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed); + EXPECT_EQ(0, fake_audio_device->Init()); + EXPECT_EQ(0, voe_base->Init(fake_audio_device.get(), audio_processing.get(), + decoder_factory_)); + VoEBase::ChannelConfig config; + config.enable_voice_pacing = true; + send_channel_id = voe_base->CreateChannel(config); + recv_channel_id = voe_base->CreateChannel(); + + AudioState::Config send_audio_state_config; + send_audio_state_config.voice_engine = voice_engine; + send_audio_state_config.audio_mixer = AudioMixerImpl::Create(); + send_audio_state_config.audio_processing = audio_processing; + Call::Config sender_config(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(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 = rtc::MakeUnique<test::PacketTransport>( + &task_queue_, sender_call_.get(), &observer, + test::PacketTransport::kSender, audio_pt_map, audio_net_config); + audio_send_transport->SetReceiver(receiver_call_->Receiver()); + + video_send_transport = rtc::MakeUnique<test::PacketTransport>( + &task_queue_, sender_call_.get(), &observer, + test::PacketTransport::kSender, video_pt_map, + FakeNetworkPipe::Config()); + video_send_transport->SetReceiver(receiver_call_->Receiver()); + + receive_transport = rtc::MakeUnique<test::PacketTransport>( + &task_queue_, receiver_call_.get(), &observer, + test::PacketTransport::kReceiver, payload_type_map_, + FakeNetworkPipe::Config()); + 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.voe_channel_id = send_channel_id; + audio_send_config.rtp.ssrc = kAudioSendSsrc; + audio_send_config.send_codec_spec = + rtc::Optional<AudioSendStream::Config::SendCodecSpec>( + {kAudioSendPayloadType, {"ISAC", 16000, 1}}); + audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory(); + audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config); + + video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; + if (fec == FecMode::kOn) { + video_send_config_.rtp.ulpfec.red_payload_type = kRedPayloadType; + video_send_config_.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; + 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.voe_channel_id = recv_channel_id; + audio_recv_config.sync_group = kSyncGroup; + audio_recv_config.decoder_factory = 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 = rtc::MakeUnique<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."; + + task_queue_.SendTask([&]() { + audio_send_stream->Stop(); + audio_receive_stream->Stop(); + + Stop(); + + DestroyStreams(); + + video_send_transport.reset(); + audio_send_transport.reset(); + receive_transport.reset(); + + sender_call_->DestroyAudioSendStream(audio_send_stream); + receiver_call_->DestroyAudioReceiveStream(audio_receive_stream); + + voe_base->DeleteChannel(send_channel_id); + voe_base->DeleteChannel(recv_channel_id); + voe_base->Release(); + + DestroyCalls(); + + VoiceEngine::Delete(voice_engine); + + fake_audio_device.reset(); + }); + + observer.PrintResults(); + + // In quick test synchronization may not be achieved in time. + if (!field_trial::IsEnabled("WebRTC-QuickPerfTest")) { + EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs")); + } +} + +TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) { + TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst, + DriftingClock::PercentsFaster(10.0f), + DriftingClock::kNoDrift, DriftingClock::kNoDrift); +} + +TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift) { + TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst, + DriftingClock::kNoDrift, + DriftingClock::PercentsSlower(30.0f), + DriftingClock::PercentsFaster(30.0f)); +} + +TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift) { + TestAudioVideoSync(FecMode::kOn, CreateOrder::kVideoFirst, + DriftingClock::kNoDrift, + DriftingClock::PercentsFaster(30.0f), + DriftingClock::PercentsSlower(30.0f)); +} + +void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& 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 FakeNetworkPipe::Config& 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: + test::PacketTransport* CreateSendTransport( + test::SingleThreadedTaskQueueForTesting* task_queue, + Call* sender_call) override { + return new test::PacketTransport(task_queue, sender_call, this, + test::PacketTransport::kSender, + payload_type_map_, net_config_); + } + + test::PacketTransport* CreateReceiveTransport( + test::SingleThreadedTaskQueueForTesting* task_queue) override { + return new test::PacketTransport(task_queue, nullptr, this, + test::PacketTransport::kReceiver, + payload_type_map_, net_config_); + } + + void OnFrame(const VideoFrame& video_frame) override { + rtc::CritScope lock(&crit_); + 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 { + rtc::CritScope lock(&crit_); + RTPHeader header; + EXPECT_TRUE(parser_->Parse(packet, length, &header)); + + 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_ = header.timestamp - first_capture_timestamp; + rtp_start_timestamp_set_ = true; + } + + uint32_t capture_timestamp = header.timestamp - rtp_start_timestamp_; + capture_time_list_.insert( + capture_time_list_.end(), + std::make_pair(header.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); + } + + rtc::CriticalSection crit_; + const FakeNetworkPipe::Config 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(&crit_); + 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 due to webrtc:8291. +#if !(defined(WEBRTC_MAC)) +TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) { + FakeNetworkPipe::Config 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, CaptureNtpTimeWithNetworkJitter) { + FakeNetworkPipe::Config 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::kStart) {} + + 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 { + // First expect CPU overuse. Then expect CPU underuse when the encoder + // delay has been decreased. + switch (test_phase_) { + 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 { kStart, kAdaptedDown, kAdaptedUp } test_phase_; + } 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) + : 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) {} + + private: + // TODO(holmer): Run this with a timer instead of once per packet. + Action OnSendRtp(const uint8_t* packet, size_t length) override { + VideoSendStream::Stats stats = send_stream_->GetStats(); + if (stats.substreams.size() > 0) { + 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 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_; + } test(pad_to_min_bitrate); + + fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps); + RunBaseTest(&test); +} + +TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); } + +TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) { + TestMinTransmitBitrate(false); +} + +TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) { + static const uint32_t kInitialBitrateKbps = 400; + static const uint32_t kReconfigureThresholdKbps = 600; + static const uint32_t kPermittedReconfiguredBitrateDiffKbps = 100; + + 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: + BitrateObserver() + : EndToEndTest(kDefaultTimeoutMs), + FakeEncoder(Clock::GetRealTimeClock()), + time_to_reconfigure_(false, false), + encoder_inits_(0), + last_set_bitrate_kbps_(0), + send_stream_(nullptr), + frame_generator_(nullptr) {} + + int32_t InitEncode(const VideoCodec* config, + int32_t number_of_cores, + size_t max_payload_size) 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, + last_set_bitrate_kbps_ - kPermittedReconfiguredBitrateDiffKbps) + << "Encoder reconfigured with bitrate too far away from last set."; + observation_complete_.Set(); + } + return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size); + } + + int32_t SetRateAllocation(const BitrateAllocation& rate_allocation, + uint32_t framerate) override { + last_set_bitrate_kbps_ = rate_allocation.get_sum_kbps(); + if (encoder_inits_ == 1 && + rate_allocation.get_sum_kbps() > kReconfigureThresholdKbps) { + time_to_reconfigure_.Set(); + } + return FakeEncoder::SetRateAllocation(rate_allocation, framerate); + } + + Call::Config GetSenderCallConfig() override { + Call::Config config = EndToEndTest::GetSenderCallConfig(); + config.event_log = event_log_.get(); + config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000; + return config; + } + + void ModifyVideoConfigs( + VideoSendStream::Config* send_config, + std::vector<VideoReceiveStream::Config>* receive_configs, + VideoEncoderConfig* encoder_config) override { + send_config->encoder_settings.encoder = this; + encoder_config->max_bitrate_bps = 2 * kReconfigureThresholdKbps * 1000; + encoder_config->video_stream_factory = + new rtc::RefCountedObject<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); + 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_; + VideoEncoderConfig encoder_config_; + } test; + + 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. +// +// |use_bitrate_allocation_strategy| use AudioPriorityBitrateAllocationStrategy +// |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( + bool use_bitrate_allocation_strategy, + 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 uint32_t kSufficientAudioBitrateBps = 16000; + static constexpr int kOpusMinBitrateBps = 6000; + static constexpr int kOpusBitrateFbBps = 32000; + static constexpr int kBitrateStabilizationMs = 10000; + static constexpr int kBitrateMeasurements = 10; + static constexpr int kBitrateMeasurementMs = 1000; + static constexpr int kMinGoodRttMs = 400; + + class MinVideoAndAudioBitrateTester : public test::EndToEndTest { + public: + MinVideoAndAudioBitrateTester(bool use_bitrate_allocation_strategy, + int test_bitrate_from, + int test_bitrate_to, + int test_bitrate_step, + int min_bwe, + int start_bwe, + int max_bwe) + : EndToEndTest(), + allocation_strategy_(new rtc::AudioPriorityBitrateAllocationStrategy( + kAudioTrackId, + kSufficientAudioBitrateBps)), + use_bitrate_allocation_strategy_(use_bitrate_allocation_strategy), + 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) {} + + protected: + FakeNetworkPipe::Config GetFakeNetworkPipeConfig() { + FakeNetworkPipe::Config pipe_config; + pipe_config.link_capacity_kbps = test_bitrate_from_; + return pipe_config; + } + + test::PacketTransport* CreateSendTransport( + test::SingleThreadedTaskQueueForTesting* task_queue, + Call* sender_call) override { + return send_transport_ = new test::PacketTransport( + task_queue, sender_call, this, test::PacketTransport::kSender, + test::CallTest::payload_type_map_, GetFakeNetworkPipeConfig()); + } + + test::PacketTransport* CreateReceiveTransport( + test::SingleThreadedTaskQueueForTesting* task_queue) override { + return receive_transport_ = new test::PacketTransport( + task_queue, nullptr, this, test::PacketTransport::kReceiver, + test::CallTest::payload_type_map_, GetFakeNetworkPipeConfig()); + } + + void PerformTest() override { + 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_) { + FakeNetworkPipe::Config pipe_config; + pipe_config.link_capacity_kbps = test_bitrate; + send_transport_->SetConfig(pipe_config); + receive_transport_->SetConfig(pipe_config); + + rtc::ThreadManager::Instance()->CurrentThread()->SleepMs( + kBitrateStabilizationMs); + + int64_t avg_rtt = 0; + for (int i = 0; i < kBitrateMeasurements; i++) { + Call::Stats call_stats = sender_call_->GetStats(); + avg_rtt += call_stats.rtt_ms; + rtc::ThreadManager::Instance()->CurrentThread()->SleepMs( + 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_", + use_bitrate_allocation_strategy_ + ? "with_allocation_strategy" + : "no_allocation_strategy", + "", last_passed_test_bitrate, "kbps", false); + } + + void OnCallsCreated(Call* sender_call, Call* receiver_call) override { + sender_call_ = sender_call; + Call::Config::BitrateConfig 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->SetBitrateConfig(bitrate_config); + if (use_bitrate_allocation_strategy_) { + sender_call->SetBitrateAllocationStrategy( + std::move(allocation_strategy_)); + } + } + + 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 { + if (use_bitrate_allocation_strategy_) { + send_config->track_id = kAudioTrackId; + send_config->min_bitrate_bps = kOpusMinBitrateBps; + send_config->max_bitrate_bps = kOpusBitrateFbBps; + } else { + send_config->send_codec_spec->target_bitrate_bps = + rtc::Optional<int>(kOpusBitrateFbBps); + } + } + + private: + std::unique_ptr<rtc::BitrateAllocationStrategy> allocation_strategy_; + const bool use_bitrate_allocation_strategy_; + 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_; + test::PacketTransport* send_transport_; + test::PacketTransport* receive_transport_; + Call* sender_call_; + } test(use_bitrate_allocation_strategy, test_bitrate_from, test_bitrate_to, + test_bitrate_step, min_bwe, start_bwe, max_bwe); + + RunBaseTest(&test); +} + +TEST_F(CallPerfTest, MinVideoAndAudioBitrate) { + TestMinAudioVideoBitrate(false, 110, 40, -10, 10000, 70000, 200000); +} +TEST_F(CallPerfTest, MinVideoAndAudioBitrateWStrategy) { + TestMinAudioVideoBitrate(true, 110, 40, -10, 10000, 70000, 200000); +} + +} // namespace webrtc |