| /* |
| * 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 "testing/gtest/include/gtest/gtest.h" |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/base/thread_annotations.h" |
| #include "webrtc/call.h" |
| #include "webrtc/call/transport_adapter.h" |
| #include "webrtc/config.h" |
| #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
| #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/include/metrics_default.h" |
| #include "webrtc/system_wrappers/include/rtp_to_ntp.h" |
| #include "webrtc/test/call_test.h" |
| #include "webrtc/test/direct_transport.h" |
| #include "webrtc/test/drifting_clock.h" |
| #include "webrtc/test/encoder_settings.h" |
| #include "webrtc/test/fake_audio_device.h" |
| #include "webrtc/test/fake_decoder.h" |
| #include "webrtc/test/fake_encoder.h" |
| #include "webrtc/test/frame_generator.h" |
| #include "webrtc/test/frame_generator_capturer.h" |
| #include "webrtc/test/rtp_rtcp_observer.h" |
| #include "webrtc/test/testsupport/fileutils.h" |
| #include "webrtc/test/testsupport/perf_test.h" |
| #include "webrtc/voice_engine/include/voe_base.h" |
| #include "webrtc/voice_engine/include/voe_codec.h" |
| #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
| #include "webrtc/voice_engine/include/voe_video_sync.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 TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms); |
| |
| 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); |
| }; |
| |
| 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", |
| test::ValuesToString(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_ GUARDED_BY(crit_); |
| std::vector<int> 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; |
| |
| metrics::Reset(); |
| VoiceEngine* voice_engine = VoiceEngine::Create(); |
| VoEBase* voe_base = VoEBase::GetInterface(voice_engine); |
| VoECodec* voe_codec = VoECodec::GetInterface(voice_engine); |
| const std::string audio_filename = |
| test::ResourcePath("voice_engine/audio_long16", "pcm"); |
| ASSERT_STRNE("", audio_filename.c_str()); |
| FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename, |
| audio_rtp_speed); |
| EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_)); |
| VoEBase::ChannelConfig config; |
| config.enable_voice_pacing = true; |
| int send_channel_id = voe_base->CreateChannel(config); |
| int recv_channel_id = voe_base->CreateChannel(); |
| |
| AudioState::Config send_audio_state_config; |
| send_audio_state_config.voice_engine = voice_engine; |
| Call::Config sender_config; |
| sender_config.audio_state = AudioState::Create(send_audio_state_config); |
| Call::Config receiver_config; |
| receiver_config.audio_state = sender_config.audio_state; |
| CreateCalls(sender_config, receiver_config); |
| |
| |
| VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock()); |
| |
| // Helper class to ensure we deliver correct media_type to the receiving call. |
| class MediaTypePacketReceiver : public PacketReceiver { |
| public: |
| MediaTypePacketReceiver(PacketReceiver* packet_receiver, |
| MediaType media_type) |
| : packet_receiver_(packet_receiver), media_type_(media_type) {} |
| |
| DeliveryStatus DeliverPacket(MediaType media_type, |
| const uint8_t* packet, |
| size_t length, |
| const PacketTime& packet_time) override { |
| return packet_receiver_->DeliverPacket(media_type_, packet, length, |
| packet_time); |
| } |
| private: |
| PacketReceiver* packet_receiver_; |
| const MediaType media_type_; |
| |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaTypePacketReceiver); |
| }; |
| |
| FakeNetworkPipe::Config audio_net_config; |
| audio_net_config.queue_delay_ms = 500; |
| audio_net_config.loss_percent = 5; |
| test::PacketTransport audio_send_transport(sender_call_.get(), &observer, |
| test::PacketTransport::kSender, |
| audio_net_config); |
| MediaTypePacketReceiver audio_receiver(receiver_call_->Receiver(), |
| MediaType::AUDIO); |
| audio_send_transport.SetReceiver(&audio_receiver); |
| |
| test::PacketTransport video_send_transport(sender_call_.get(), &observer, |
| test::PacketTransport::kSender, |
| FakeNetworkPipe::Config()); |
| MediaTypePacketReceiver video_receiver(receiver_call_->Receiver(), |
| MediaType::VIDEO); |
| video_send_transport.SetReceiver(&video_receiver); |
| |
| test::PacketTransport receive_transport( |
| receiver_call_.get(), &observer, test::PacketTransport::kReceiver, |
| FakeNetworkPipe::Config()); |
| receive_transport.SetReceiver(sender_call_->Receiver()); |
| |
| test::FakeDecoder fake_decoder; |
| |
| CreateSendConfig(1, 0, &video_send_transport); |
| CreateMatchingReceiveConfigs(&receive_transport); |
| |
| AudioSendStream::Config audio_send_config(&audio_send_transport); |
| audio_send_config.voe_channel_id = send_channel_id; |
| audio_send_config.rtp.ssrc = kAudioSendSsrc; |
| AudioSendStream* audio_send_stream = |
| sender_call_->CreateAudioSendStream(audio_send_config); |
| |
| CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000}; |
| EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac)); |
| |
| video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| if (fec == FecMode::kOn) { |
| video_send_config_.rtp.fec.red_payload_type = kRedPayloadType; |
| video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; |
| video_receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType; |
| video_receive_configs_[0].rtp.fec.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_; |
| |
| AudioReceiveStream* audio_receive_stream; |
| |
| 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]); |
| DriftingClock drifting_clock(clock_, video_ntp_speed); |
| CreateFrameGeneratorCapturerWithDrift(&drifting_clock, video_rtp_speed); |
| |
| Start(); |
| |
| fake_audio_device.Start(); |
| EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id)); |
| EXPECT_EQ(0, voe_base->StartReceive(recv_channel_id)); |
| EXPECT_EQ(0, voe_base->StartSend(send_channel_id)); |
| |
| EXPECT_TRUE(observer.Wait()) |
| << "Timed out while waiting for audio and video to be synchronized."; |
| |
| EXPECT_EQ(0, voe_base->StopSend(send_channel_id)); |
| EXPECT_EQ(0, voe_base->StopReceive(recv_channel_id)); |
| EXPECT_EQ(0, voe_base->StopPlayout(recv_channel_id)); |
| fake_audio_device.Stop(); |
| |
| Stop(); |
| video_send_transport.StopSending(); |
| audio_send_transport.StopSending(); |
| receive_transport.StopSending(); |
| |
| DestroyStreams(); |
| |
| 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(); |
| voe_codec->Release(); |
| |
| DestroyCalls(); |
| |
| VoiceEngine::Delete(voice_engine); |
| |
| observer.PrintResults(); |
| 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(Call* sender_call) override { |
| return new test::PacketTransport( |
| sender_call, this, test::PacketTransport::kSender, net_config_); |
| } |
| |
| test::PacketTransport* CreateReceiveTransport() override { |
| return new test::PacketTransport( |
| nullptr, this, test::PacketTransport::kReceiver, 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", |
| test::ValuesToString(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_ GUARDED_BY(&crit_); |
| std::vector<int> time_offset_ms_list_; |
| } test(net_config, threshold_ms, start_time_ms, run_time_ms); |
| |
| RunBaseTest(&test); |
| } |
| |
| 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); |
| } |
| |
| void CallPerfTest::TestCpuOveruse(LoadObserver::Load tested_load, |
| int encode_delay_ms) { |
| class LoadObserver : public test::SendTest, public webrtc::LoadObserver { |
| public: |
| LoadObserver(LoadObserver::Load tested_load, int encode_delay_ms) |
| : SendTest(kLongTimeoutMs), |
| tested_load_(tested_load), |
| encoder_(Clock::GetRealTimeClock(), encode_delay_ms) {} |
| |
| void OnLoadUpdate(Load load) override { |
| if (load == tested_load_) |
| observation_complete_.Set(); |
| } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStream::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| send_config->overuse_callback = this; |
| send_config->encoder_settings.encoder = &encoder_; |
| } |
| |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback."; |
| } |
| |
| LoadObserver::Load tested_load_; |
| test::DelayedEncoder encoder_; |
| } test(tested_load, encode_delay_ms); |
| |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(CallPerfTest, ReceivesCpuUnderuse) { |
| const int kEncodeDelayMs = 2; |
| TestCpuOveruse(LoadObserver::kUnderuse, kEncodeDelayMs); |
| } |
| |
| TEST_F(CallPerfTest, ReceivesCpuOveruse) { |
| const int kEncodeDelayMs = 35; |
| TestCpuOveruse(LoadObserver::kOveruse, kEncodeDelayMs); |
| } |
| |
| 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(1u, 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", test::ValuesToString(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<size_t> 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 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_(0), |
| send_stream_(nullptr) {} |
| |
| int32_t InitEncode(const VideoCodec* config, |
| int32_t number_of_cores, |
| size_t max_payload_size) override { |
| if (encoder_inits_ == 0) { |
| EXPECT_EQ(kInitialBitrateKbps, config->startBitrate) |
| << "Encoder not initialized at expected bitrate."; |
| } |
| ++encoder_inits_; |
| if (encoder_inits_ == 2) { |
| EXPECT_GE(last_set_bitrate_, kReconfigureThresholdKbps); |
| EXPECT_NEAR(config->startBitrate, |
| last_set_bitrate_, |
| 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 SetRates(uint32_t new_target_bitrate_kbps, |
| uint32_t framerate) override { |
| last_set_bitrate_ = new_target_bitrate_kbps; |
| if (encoder_inits_ == 1 && |
| new_target_bitrate_kbps > kReconfigureThresholdKbps) { |
| time_to_reconfigure_.Set(); |
| } |
| return FakeEncoder::SetRates(new_target_bitrate_kbps, framerate); |
| } |
| |
| Call::Config GetSenderCallConfig() override { |
| Call::Config config = EndToEndTest::GetSenderCallConfig(); |
| 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->streams[0].min_bitrate_bps = 50000; |
| encoder_config->streams[0].target_bitrate_bps = |
| encoder_config->streams[0].max_bitrate_bps = 2000000; |
| |
| encoder_config_ = encoder_config->Copy(); |
| } |
| |
| void OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& receive_streams) override { |
| send_stream_ = send_stream; |
| } |
| |
| void PerformTest() override { |
| ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs)) |
| << "Timed out before receiving an initial high bitrate."; |
| encoder_config_.streams[0].width *= 2; |
| encoder_config_.streams[0].height *= 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_; |
| VideoSendStream* send_stream_; |
| VideoEncoderConfig encoder_config_; |
| } test; |
| |
| RunBaseTest(&test); |
| } |
| |
| } // namespace webrtc |
