| /* |
| * 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 <sstream> |
| #include <string> |
| |
| #include "testing/gtest/include/gtest/gtest.h" |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/scoped_ptr.h" |
| #include "webrtc/base/thread_annotations.h" |
| #include "webrtc/call.h" |
| #include "webrtc/call/transport_adapter.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/rtp_to_ntp.h" |
| #include "webrtc/test/call_test.h" |
| #include "webrtc/test/direct_transport.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_network.h" |
| #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" |
| #include "webrtc/voice_engine/include/voe_video_sync.h" |
| |
| namespace webrtc { |
| |
| class CallPerfTest : public test::CallTest { |
| protected: |
| void TestAudioVideoSync(bool fec, bool create_audio_first); |
| |
| 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 SyncRtcpObserver : public test::RtpRtcpObserver { |
| public: |
| SyncRtcpObserver() : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs) {} |
| |
| Action OnSendRtcp(const uint8_t* packet, size_t length) override { |
| RTCPUtility::RTCPParserV2 parser(packet, length, true); |
| EXPECT_TRUE(parser.IsValid()); |
| |
| for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); |
| packet_type != RTCPUtility::RTCPPacketTypes::kInvalid; |
| packet_type = parser.Iterate()) { |
| if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) { |
| const RTCPUtility::RTCPPacket& packet = parser.Packet(); |
| RtcpMeasurement ntp_rtp_pair( |
| packet.SR.NTPMostSignificant, |
| packet.SR.NTPLeastSignificant, |
| packet.SR.RTPTimestamp); |
| StoreNtpRtpPair(ntp_rtp_pair); |
| } |
| } |
| return SEND_PACKET; |
| } |
| |
| int64_t RtpTimestampToNtp(uint32_t timestamp) const { |
| rtc::CritScope lock(&crit_); |
| int64_t timestamp_in_ms = -1; |
| if (ntp_rtp_pairs_.size() == 2) { |
| // TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the |
| // RTCP sender where it sends RTCP SR before any RTP packets, which leads |
| // to a bogus NTP/RTP mapping. |
| RtpToNtpMs(timestamp, ntp_rtp_pairs_, ×tamp_in_ms); |
| return timestamp_in_ms; |
| } |
| return -1; |
| } |
| |
| private: |
| void StoreNtpRtpPair(RtcpMeasurement ntp_rtp_pair) { |
| rtc::CritScope lock(&crit_); |
| for (RtcpList::iterator it = ntp_rtp_pairs_.begin(); |
| it != ntp_rtp_pairs_.end(); |
| ++it) { |
| if (ntp_rtp_pair.ntp_secs == it->ntp_secs && |
| ntp_rtp_pair.ntp_frac == it->ntp_frac) { |
| // This RTCP has already been added to the list. |
| return; |
| } |
| } |
| // We need two RTCP SR reports to map between RTP and NTP. More than two |
| // will not improve the mapping. |
| if (ntp_rtp_pairs_.size() == 2) { |
| ntp_rtp_pairs_.pop_back(); |
| } |
| ntp_rtp_pairs_.push_front(ntp_rtp_pair); |
| } |
| |
| mutable rtc::CriticalSection crit_; |
| RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_); |
| }; |
| |
| class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer { |
| static const int kInSyncThresholdMs = 50; |
| static const int kStartupTimeMs = 2000; |
| static const int kMinRunTimeMs = 30000; |
| |
| public: |
| VideoRtcpAndSyncObserver(Clock* clock, |
| int voe_channel, |
| VoEVideoSync* voe_sync, |
| SyncRtcpObserver* audio_observer) |
| : clock_(clock), |
| voe_channel_(voe_channel), |
| voe_sync_(voe_sync), |
| audio_observer_(audio_observer), |
| creation_time_ms_(clock_->TimeInMilliseconds()), |
| first_time_in_sync_(-1) {} |
| |
| void RenderFrame(const VideoFrame& video_frame, |
| int time_to_render_ms) override { |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| uint32_t playout_timestamp = 0; |
| if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0) |
| return; |
| int64_t latest_audio_ntp = |
| audio_observer_->RtpTimestampToNtp(playout_timestamp); |
| int64_t latest_video_ntp = RtpTimestampToNtp(video_frame.timestamp()); |
| if (latest_audio_ntp < 0 || latest_video_ntp < 0) |
| return; |
| int time_until_render_ms = |
| std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms)); |
| latest_video_ntp += time_until_render_ms; |
| int64_t stream_offset = latest_audio_ntp - latest_video_ntp; |
| std::stringstream ss; |
| ss << stream_offset; |
| webrtc::test::PrintResult("stream_offset", |
| "", |
| "synchronization", |
| ss.str(), |
| "ms", |
| false); |
| 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(latest_audio_ntp - latest_video_ntp) < 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(); |
| } |
| } |
| |
| bool IsTextureSupported() const override { return false; } |
| |
| private: |
| Clock* const clock_; |
| const int voe_channel_; |
| VoEVideoSync* const voe_sync_; |
| SyncRtcpObserver* const audio_observer_; |
| const int64_t creation_time_ms_; |
| int64_t first_time_in_sync_; |
| }; |
| |
| void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) { |
| const char* kSyncGroup = "av_sync"; |
| class AudioPacketReceiver : public PacketReceiver { |
| public: |
| AudioPacketReceiver(int channel, VoENetwork* voe_network) |
| : channel_(channel), |
| voe_network_(voe_network), |
| parser_(RtpHeaderParser::Create()) {} |
| DeliveryStatus DeliverPacket(MediaType media_type, |
| const uint8_t* packet, |
| size_t length, |
| const PacketTime& packet_time) override { |
| EXPECT_TRUE(media_type == MediaType::ANY || |
| media_type == MediaType::AUDIO); |
| int ret; |
| if (parser_->IsRtcp(packet, length)) { |
| ret = voe_network_->ReceivedRTCPPacket(channel_, packet, length); |
| } else { |
| ret = voe_network_->ReceivedRTPPacket(channel_, packet, length, |
| PacketTime()); |
| } |
| return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR; |
| } |
| |
| private: |
| int channel_; |
| VoENetwork* voe_network_; |
| rtc::scoped_ptr<RtpHeaderParser> parser_; |
| }; |
| |
| VoiceEngine* voice_engine = VoiceEngine::Create(); |
| VoEBase* voe_base = VoEBase::GetInterface(voice_engine); |
| VoECodec* voe_codec = VoECodec::GetInterface(voice_engine); |
| VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine); |
| VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine); |
| const std::string audio_filename = |
| test::ResourcePath("voice_engine/audio_long16", "pcm"); |
| ASSERT_STRNE("", audio_filename.c_str()); |
| test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), |
| audio_filename); |
| EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr)); |
| int channel = voe_base->CreateChannel(); |
| |
| SyncRtcpObserver audio_observer; |
| |
| AudioState::Config audio_state_config; |
| audio_state_config.voice_engine = voice_engine; |
| Call::Config receiver_config; |
| receiver_config.audio_state = AudioState::Create(audio_state_config); |
| CreateCalls(Call::Config(), receiver_config); |
| |
| CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000}; |
| EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac)); |
| |
| AudioPacketReceiver voe_packet_receiver(channel, voe_network); |
| |
| FakeNetworkPipe::Config net_config; |
| net_config.queue_delay_ms = 500; |
| net_config.loss_percent = 5; |
| test::PacketTransport audio_send_transport( |
| nullptr, &audio_observer, test::PacketTransport::kSender, net_config); |
| audio_send_transport.SetReceiver(&voe_packet_receiver); |
| test::PacketTransport audio_receive_transport( |
| nullptr, &audio_observer, test::PacketTransport::kReceiver, net_config); |
| audio_receive_transport.SetReceiver(&voe_packet_receiver); |
| |
| internal::TransportAdapter transport_adapter(&audio_send_transport); |
| transport_adapter.Enable(); |
| EXPECT_EQ(0, |
| voe_network->RegisterExternalTransport(channel, transport_adapter)); |
| |
| VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), channel, |
| voe_sync, &audio_observer); |
| |
| test::PacketTransport sync_send_transport(sender_call_.get(), &observer, |
| test::PacketTransport::kSender, |
| FakeNetworkPipe::Config()); |
| sync_send_transport.SetReceiver(receiver_call_->Receiver()); |
| test::PacketTransport sync_receive_transport(receiver_call_.get(), &observer, |
| test::PacketTransport::kReceiver, |
| FakeNetworkPipe::Config()); |
| sync_receive_transport.SetReceiver(sender_call_->Receiver()); |
| |
| test::FakeDecoder fake_decoder; |
| |
| CreateSendConfig(1, &sync_send_transport); |
| CreateMatchingReceiveConfigs(&sync_receive_transport); |
| |
| send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| if (fec) { |
| send_config_.rtp.fec.red_payload_type = kRedPayloadType; |
| send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; |
| receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType; |
| receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; |
| } |
| receive_configs_[0].rtp.nack.rtp_history_ms = 1000; |
| receive_configs_[0].renderer = &observer; |
| receive_configs_[0].sync_group = kSyncGroup; |
| |
| AudioReceiveStream::Config audio_config; |
| audio_config.voe_channel_id = channel; |
| audio_config.sync_group = kSyncGroup; |
| |
| AudioReceiveStream* audio_receive_stream = nullptr; |
| |
| if (create_audio_first) { |
| audio_receive_stream = |
| receiver_call_->CreateAudioReceiveStream(audio_config); |
| CreateStreams(); |
| } else { |
| CreateStreams(); |
| audio_receive_stream = |
| receiver_call_->CreateAudioReceiveStream(audio_config); |
| } |
| |
| CreateFrameGeneratorCapturer(); |
| |
| Start(); |
| |
| fake_audio_device.Start(); |
| EXPECT_EQ(0, voe_base->StartPlayout(channel)); |
| EXPECT_EQ(0, voe_base->StartReceive(channel)); |
| EXPECT_EQ(0, voe_base->StartSend(channel)); |
| |
| EXPECT_EQ(kEventSignaled, observer.Wait()) |
| << "Timed out while waiting for audio and video to be synchronized."; |
| |
| EXPECT_EQ(0, voe_base->StopSend(channel)); |
| EXPECT_EQ(0, voe_base->StopReceive(channel)); |
| EXPECT_EQ(0, voe_base->StopPlayout(channel)); |
| fake_audio_device.Stop(); |
| |
| Stop(); |
| sync_send_transport.StopSending(); |
| sync_receive_transport.StopSending(); |
| audio_send_transport.StopSending(); |
| audio_receive_transport.StopSending(); |
| |
| voe_base->DeleteChannel(channel); |
| voe_base->Release(); |
| voe_codec->Release(); |
| voe_network->Release(); |
| voe_sync->Release(); |
| |
| DestroyStreams(); |
| |
| receiver_call_->DestroyAudioReceiveStream(audio_receive_stream); |
| |
| DestroyCalls(); |
| |
| VoiceEngine::Delete(voice_engine); |
| } |
| |
| TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioCreatedFirst) { |
| TestAudioVideoSync(false, true); |
| } |
| |
| TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoCreatedFirst) { |
| TestAudioVideoSync(false, false); |
| } |
| |
| TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) { |
| TestAudioVideoSync(true, false); |
| } |
| |
| 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 VideoRenderer { |
| public: |
| CaptureNtpTimeObserver(int threshold_ms, int start_time_ms, int run_time_ms) |
| : EndToEndTest(kLongTimeoutMs), |
| 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: |
| void RenderFrame(const VideoFrame& video_frame, |
| int time_to_render_ms) 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; |
| std::stringstream ss; |
| ss << time_offset_ms; |
| |
| webrtc::test::PrintResult( |
| "capture_ntp_time", "", "real - estimated", ss.str(), "ms", true); |
| EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_); |
| } |
| |
| bool IsTextureSupported() const override { return false; } |
| |
| virtual Action OnSendRtp(const uint8_t* packet, size_t length) { |
| 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 ModifyConfigs(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_EQ(kEventSignaled, Wait()) << "Timed out while waiting for " |
| "estimated capture NTP time to be " |
| "within bounds."; |
| } |
| |
| rtc::CriticalSection crit_; |
| 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_); |
| } test(threshold_ms, start_time_ms, run_time_ms); |
| |
| RunBaseTest(&test, net_config); |
| } |
| |
| 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 ModifyConfigs(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_EQ(kEventSignaled, Wait()) |
| << "Timed out before receiving an overuse callback."; |
| } |
| |
| LoadObserver::Load tested_load_; |
| test::DelayedEncoder encoder_; |
| } test(tested_load, encode_delay_ms); |
| |
| RunBaseTest(&test, FakeNetworkPipe::Config()); |
| } |
| |
| 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), |
| pad_to_min_bitrate_(using_min_transmit_bitrate), |
| 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 > 0) { |
| test::PrintResult( |
| "bitrate_stats_", |
| (pad_to_min_bitrate_ ? "min_transmit_bitrate" |
| : "without_min_transmit_bitrate"), |
| "bitrate_kbps", |
| static_cast<size_t>(bitrate_kbps), |
| "kbps", |
| false); |
| if (pad_to_min_bitrate_) { |
| if (bitrate_kbps > kMinAcceptableTransmitBitrate && |
| bitrate_kbps < kMaxAcceptableTransmitBitrate) { |
| ++num_bitrate_observations_in_range_; |
| } |
| } else { |
| // Expect bitrate stats to roughly match the max encode bitrate. |
| if (bitrate_kbps > (kMaxEncodeBitrateKbps - |
| kAcceptableBitrateErrorMargin / 2) && |
| bitrate_kbps < (kMaxEncodeBitrateKbps + |
| kAcceptableBitrateErrorMargin / 2)) { |
| ++num_bitrate_observations_in_range_; |
| } |
| } |
| if (num_bitrate_observations_in_range_ == |
| kNumBitrateObservationsInRange) |
| observation_complete_->Set(); |
| } |
| } |
| return SEND_PACKET; |
| } |
| |
| void OnStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& receive_streams) override { |
| send_stream_ = send_stream; |
| } |
| |
| void ModifyConfigs(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_EQ(kEventSignaled, Wait()) |
| << "Timeout while waiting for send-bitrate stats."; |
| } |
| |
| VideoSendStream* send_stream_; |
| const bool pad_to_min_bitrate_; |
| int num_bitrate_observations_in_range_; |
| } test(pad_to_min_bitrate); |
| |
| fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps); |
| RunBaseTest(&test, FakeNetworkPipe::Config()); |
| } |
| |
| 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_(webrtc::EventWrapper::Create()), |
| 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 ModifyConfigs(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; |
| } |
| |
| void OnStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& receive_streams) override { |
| send_stream_ = send_stream; |
| } |
| |
| void PerformTest() override { |
| ASSERT_EQ(kEventSignaled, 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; |
| EXPECT_TRUE(send_stream_->ReconfigureVideoEncoder(encoder_config_)); |
| EXPECT_EQ(kEventSignaled, Wait()) |
| << "Timed out while waiting for a couple of high bitrate estimates " |
| "after reconfiguring the send stream."; |
| } |
| |
| private: |
| rtc::scoped_ptr<webrtc::EventWrapper> time_to_reconfigure_; |
| int encoder_inits_; |
| uint32_t last_set_bitrate_; |
| VideoSendStream* send_stream_; |
| VideoEncoderConfig encoder_config_; |
| } test; |
| |
| RunBaseTest(&test, FakeNetworkPipe::Config()); |
| } |
| |
| } // namespace webrtc |
