| /* |
| * Copyright 2018 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 <memory> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/types/optional.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "api/test/simulated_network.h" |
| #include "api/test/video/function_video_encoder_factory.h" |
| #include "call/fake_network_pipe.h" |
| #include "call/simulated_network.h" |
| #include "modules/rtp_rtcp/source/rtp_utility.h" |
| #include "modules/video_coding/include/video_coding_defines.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/task_queue_for_test.h" |
| #include "system_wrappers/include/metrics.h" |
| #include "system_wrappers/include/sleep.h" |
| #include "test/call_test.h" |
| #include "test/fake_encoder.h" |
| #include "test/gtest.h" |
| #include "test/rtcp_packet_parser.h" |
| |
| namespace webrtc { |
| namespace { |
| enum : int { // The first valid value is 1. |
| kVideoContentTypeExtensionId = 1, |
| }; |
| } // namespace |
| |
| class StatsEndToEndTest : public test::CallTest { |
| public: |
| StatsEndToEndTest() { |
| RegisterRtpExtension(RtpExtension(RtpExtension::kVideoContentTypeUri, |
| kVideoContentTypeExtensionId)); |
| } |
| }; |
| |
| TEST_F(StatsEndToEndTest, GetStats) { |
| static const int kStartBitrateBps = 3000000; |
| static const int kExpectedRenderDelayMs = 20; |
| |
| class ReceiveStreamRenderer : public rtc::VideoSinkInterface<VideoFrame> { |
| public: |
| ReceiveStreamRenderer() {} |
| |
| private: |
| void OnFrame(const VideoFrame& video_frame) override {} |
| }; |
| |
| class StatsObserver : public test::EndToEndTest { |
| public: |
| StatsObserver() |
| : EndToEndTest(kLongTimeoutMs), |
| encoder_factory_([]() { |
| return std::make_unique<test::DelayedEncoder>( |
| Clock::GetRealTimeClock(), 10); |
| }), |
| send_stream_(nullptr), |
| expected_send_ssrcs_() {} |
| |
| private: |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| // Drop every 25th packet => 4% loss. |
| static const int kPacketLossFrac = 25; |
| RTPHeader header; |
| RtpUtility::RtpHeaderParser parser(packet, length); |
| if (parser.Parse(&header) && |
| expected_send_ssrcs_.find(header.ssrc) != |
| expected_send_ssrcs_.end() && |
| header.sequenceNumber % kPacketLossFrac == 0) { |
| return DROP_PACKET; |
| } |
| check_stats_event_.Set(); |
| return SEND_PACKET; |
| } |
| |
| Action OnSendRtcp(const uint8_t* packet, size_t length) override { |
| check_stats_event_.Set(); |
| return SEND_PACKET; |
| } |
| |
| Action OnReceiveRtp(const uint8_t* packet, size_t length) override { |
| check_stats_event_.Set(); |
| return SEND_PACKET; |
| } |
| |
| Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { |
| check_stats_event_.Set(); |
| return SEND_PACKET; |
| } |
| |
| bool CheckReceiveStats() { |
| for (size_t i = 0; i < receive_streams_.size(); ++i) { |
| VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats(); |
| EXPECT_EQ(expected_receive_ssrcs_[i], stats.ssrc); |
| |
| // Make sure all fields have been populated. |
| // TODO(pbos): Use CompoundKey if/when we ever know that all stats are |
| // always filled for all receivers. |
| receive_stats_filled_["IncomingRate"] |= |
| stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0; |
| |
| send_stats_filled_["DecoderImplementationName"] |= |
| stats.decoder_implementation_name == |
| test::FakeDecoder::kImplementationName; |
| receive_stats_filled_["RenderDelayAsHighAsExpected"] |= |
| stats.render_delay_ms >= kExpectedRenderDelayMs; |
| |
| receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0; |
| |
| receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0; |
| |
| receive_stats_filled_["StatisticsUpdated"] |= |
| stats.rtp_stats.packets_lost != 0 || stats.rtp_stats.jitter != 0; |
| |
| receive_stats_filled_["DataCountersUpdated"] |= |
| stats.rtp_stats.packet_counter.payload_bytes != 0 || |
| stats.rtp_stats.packet_counter.header_bytes != 0 || |
| stats.rtp_stats.packet_counter.packets != 0 || |
| stats.rtp_stats.packet_counter.padding_bytes != 0; |
| |
| receive_stats_filled_["CodecStats"] |= stats.target_delay_ms != 0; |
| |
| receive_stats_filled_["FrameCounts"] |= |
| stats.frame_counts.key_frames != 0 || |
| stats.frame_counts.delta_frames != 0; |
| |
| receive_stats_filled_["CName"] |= !stats.c_name.empty(); |
| |
| receive_stats_filled_["RtcpPacketTypeCount"] |= |
| stats.rtcp_packet_type_counts.fir_packets != 0 || |
| stats.rtcp_packet_type_counts.nack_packets != 0 || |
| stats.rtcp_packet_type_counts.pli_packets != 0 || |
| stats.rtcp_packet_type_counts.nack_requests != 0 || |
| stats.rtcp_packet_type_counts.unique_nack_requests != 0; |
| |
| assert(stats.current_payload_type == -1 || |
| stats.current_payload_type == kFakeVideoSendPayloadType); |
| receive_stats_filled_["IncomingPayloadType"] |= |
| stats.current_payload_type == kFakeVideoSendPayloadType; |
| } |
| |
| return AllStatsFilled(receive_stats_filled_); |
| } |
| |
| bool CheckSendStats() { |
| RTC_DCHECK(send_stream_); |
| VideoSendStream::Stats stats = send_stream_->GetStats(); |
| |
| size_t expected_num_streams = |
| kNumSimulcastStreams + expected_send_ssrcs_.size(); |
| send_stats_filled_["NumStreams"] |= |
| stats.substreams.size() == expected_num_streams; |
| |
| send_stats_filled_["CpuOveruseMetrics"] |= |
| stats.avg_encode_time_ms != 0 && stats.encode_usage_percent != 0 && |
| stats.total_encode_time_ms != 0; |
| |
| send_stats_filled_["EncoderImplementationName"] |= |
| stats.encoder_implementation_name == |
| test::FakeEncoder::kImplementationName; |
| |
| for (const auto& kv : stats.substreams) { |
| if (expected_send_ssrcs_.find(kv.first) == expected_send_ssrcs_.end()) |
| continue; // Probably RTX. |
| |
| send_stats_filled_[CompoundKey("CapturedFrameRate", kv.first)] |= |
| stats.input_frame_rate != 0; |
| |
| const VideoSendStream::StreamStats& stream_stats = kv.second; |
| |
| send_stats_filled_[CompoundKey("StatisticsUpdated", kv.first)] |= |
| stream_stats.rtcp_stats.packets_lost != 0 || |
| stream_stats.rtcp_stats.extended_highest_sequence_number != 0 || |
| stream_stats.rtcp_stats.fraction_lost != 0; |
| |
| send_stats_filled_[CompoundKey("DataCountersUpdated", kv.first)] |= |
| stream_stats.rtp_stats.fec.packets != 0 || |
| stream_stats.rtp_stats.transmitted.padding_bytes != 0 || |
| stream_stats.rtp_stats.retransmitted.packets != 0 || |
| stream_stats.rtp_stats.transmitted.packets != 0; |
| |
| send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Total", |
| kv.first)] |= |
| stream_stats.total_bitrate_bps != 0; |
| |
| send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Retransmit", |
| kv.first)] |= |
| stream_stats.retransmit_bitrate_bps != 0; |
| |
| send_stats_filled_[CompoundKey("FrameCountObserver", kv.first)] |= |
| stream_stats.frame_counts.delta_frames != 0 || |
| stream_stats.frame_counts.key_frames != 0; |
| |
| send_stats_filled_[CompoundKey("OutgoingRate", kv.first)] |= |
| stats.encode_frame_rate != 0; |
| |
| send_stats_filled_[CompoundKey("Delay", kv.first)] |= |
| stream_stats.avg_delay_ms != 0 || stream_stats.max_delay_ms != 0; |
| |
| // TODO(pbos): Use CompoundKey when the test makes sure that all SSRCs |
| // report dropped packets. |
| send_stats_filled_["RtcpPacketTypeCount"] |= |
| stream_stats.rtcp_packet_type_counts.fir_packets != 0 || |
| stream_stats.rtcp_packet_type_counts.nack_packets != 0 || |
| stream_stats.rtcp_packet_type_counts.pli_packets != 0 || |
| stream_stats.rtcp_packet_type_counts.nack_requests != 0 || |
| stream_stats.rtcp_packet_type_counts.unique_nack_requests != 0; |
| } |
| |
| return AllStatsFilled(send_stats_filled_); |
| } |
| |
| std::string CompoundKey(const char* name, uint32_t ssrc) { |
| rtc::StringBuilder oss; |
| oss << name << "_" << ssrc; |
| return oss.Release(); |
| } |
| |
| bool AllStatsFilled(const std::map<std::string, bool>& stats_map) { |
| for (const auto& stat : stats_map) { |
| if (!stat.second) |
| return false; |
| } |
| return true; |
| } |
| |
| std::unique_ptr<test::PacketTransport> CreateSendTransport( |
| TaskQueueBase* task_queue, |
| Call* sender_call) override { |
| BuiltInNetworkBehaviorConfig network_config; |
| network_config.loss_percent = 5; |
| 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>(network_config))); |
| } |
| |
| void ModifySenderBitrateConfig( |
| BitrateConstraints* bitrate_config) override { |
| bitrate_config->start_bitrate_bps = kStartBitrateBps; |
| } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStream::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| // Set low simulcast bitrates to not have to wait for bandwidth ramp-up. |
| encoder_config->max_bitrate_bps = 50000; |
| for (auto& layer : encoder_config->simulcast_layers) { |
| layer.min_bitrate_bps = 10000; |
| layer.target_bitrate_bps = 15000; |
| layer.max_bitrate_bps = 20000; |
| } |
| |
| send_config->rtp.c_name = "SomeCName"; |
| send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| send_config->rtp.rtx.payload_type = kSendRtxPayloadType; |
| |
| const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs; |
| for (size_t i = 0; i < ssrcs.size(); ++i) { |
| expected_send_ssrcs_.insert(ssrcs[i]); |
| expected_receive_ssrcs_.push_back( |
| (*receive_configs)[i].rtp.remote_ssrc); |
| (*receive_configs)[i].render_delay_ms = kExpectedRenderDelayMs; |
| (*receive_configs)[i].renderer = &receive_stream_renderer_; |
| (*receive_configs)[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| |
| (*receive_configs)[i].rtp.rtx_ssrc = kSendRtxSsrcs[i]; |
| (*receive_configs)[i] |
| .rtp.rtx_associated_payload_types[kSendRtxPayloadType] = |
| kFakeVideoSendPayloadType; |
| } |
| |
| for (size_t i = 0; i < kNumSimulcastStreams; ++i) |
| send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]); |
| |
| // Use a delayed encoder to make sure we see CpuOveruseMetrics stats that |
| // are non-zero. |
| send_config->encoder_settings.encoder_factory = &encoder_factory_; |
| } |
| |
| size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; } |
| |
| void OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& receive_streams) override { |
| send_stream_ = send_stream; |
| receive_streams_ = receive_streams; |
| task_queue_ = TaskQueueBase::Current(); |
| } |
| |
| void PerformTest() override { |
| Clock* clock = Clock::GetRealTimeClock(); |
| int64_t now_ms = clock->TimeInMilliseconds(); |
| int64_t stop_time_ms = now_ms + test::CallTest::kLongTimeoutMs; |
| bool receive_ok = false; |
| bool send_ok = false; |
| |
| while (now_ms < stop_time_ms) { |
| if (!receive_ok && task_queue_) { |
| SendTask(RTC_FROM_HERE, task_queue_, |
| [&]() { receive_ok = CheckReceiveStats(); }); |
| } |
| if (!send_ok) |
| send_ok = CheckSendStats(); |
| |
| if (receive_ok && send_ok) |
| return; |
| |
| int64_t time_until_timeout_ms = stop_time_ms - now_ms; |
| if (time_until_timeout_ms > 0) |
| check_stats_event_.Wait(time_until_timeout_ms); |
| now_ms = clock->TimeInMilliseconds(); |
| } |
| |
| ADD_FAILURE() << "Timed out waiting for filled stats."; |
| for (const auto& kv : receive_stats_filled_) { |
| if (!kv.second) { |
| ADD_FAILURE() << "Missing receive stats: " << kv.first; |
| } |
| } |
| for (const auto& kv : send_stats_filled_) { |
| if (!kv.second) { |
| ADD_FAILURE() << "Missing send stats: " << kv.first; |
| } |
| } |
| } |
| |
| test::FunctionVideoEncoderFactory encoder_factory_; |
| std::vector<VideoReceiveStream*> receive_streams_; |
| std::map<std::string, bool> receive_stats_filled_; |
| |
| VideoSendStream* send_stream_; |
| std::map<std::string, bool> send_stats_filled_; |
| |
| std::vector<uint32_t> expected_receive_ssrcs_; |
| std::set<uint32_t> expected_send_ssrcs_; |
| |
| rtc::Event check_stats_event_; |
| ReceiveStreamRenderer receive_stream_renderer_; |
| TaskQueueBase* task_queue_ = nullptr; |
| } test; |
| |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(StatsEndToEndTest, TimingFramesAreReported) { |
| static const int kExtensionId = 5; |
| |
| class StatsObserver : public test::EndToEndTest { |
| public: |
| StatsObserver() : EndToEndTest(kLongTimeoutMs) {} |
| |
| private: |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStream::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| send_config->rtp.extensions.clear(); |
| send_config->rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId)); |
| for (auto& receive_config : *receive_configs) { |
| receive_config.rtp.extensions.clear(); |
| receive_config.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId)); |
| } |
| } |
| |
| void OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& receive_streams) override { |
| receive_streams_ = receive_streams; |
| task_queue_ = TaskQueueBase::Current(); |
| } |
| |
| void PerformTest() override { |
| // No frames reported initially. |
| SendTask(RTC_FROM_HERE, task_queue_, [&]() { |
| for (const auto& receive_stream : receive_streams_) { |
| EXPECT_FALSE(receive_stream->GetStats().timing_frame_info); |
| } |
| }); |
| // Wait for at least one timing frame to be sent with 100ms grace period. |
| SleepMs(kDefaultTimingFramesDelayMs + 100); |
| // Check that timing frames are reported for each stream. |
| SendTask(RTC_FROM_HERE, task_queue_, [&]() { |
| for (const auto& receive_stream : receive_streams_) { |
| EXPECT_TRUE(receive_stream->GetStats().timing_frame_info); |
| } |
| }); |
| } |
| |
| std::vector<VideoReceiveStream*> receive_streams_; |
| TaskQueueBase* task_queue_ = nullptr; |
| } test; |
| |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(StatsEndToEndTest, TestReceivedRtpPacketStats) { |
| static const size_t kNumRtpPacketsToSend = 5; |
| class ReceivedRtpStatsObserver : public test::EndToEndTest, |
| public QueuedTask { |
| public: |
| ReceivedRtpStatsObserver() |
| : EndToEndTest(kDefaultTimeoutMs), |
| receive_stream_(nullptr), |
| sent_rtp_(0) {} |
| |
| private: |
| void OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& receive_streams) override { |
| receive_stream_ = receive_streams[0]; |
| task_queue_ = TaskQueueBase::Current(); |
| EXPECT_TRUE(task_queue_ != nullptr); |
| } |
| |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| if (sent_rtp_ >= kNumRtpPacketsToSend) { |
| // Need to check the stats on the correct thread. |
| task_queue_->PostTask(std::unique_ptr<QueuedTask>(this)); |
| return DROP_PACKET; |
| } |
| ++sent_rtp_; |
| return SEND_PACKET; |
| } |
| |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) |
| << "Timed out while verifying number of received RTP packets."; |
| } |
| |
| bool Run() override { |
| VideoReceiveStream::Stats stats = receive_stream_->GetStats(); |
| if (kNumRtpPacketsToSend == stats.rtp_stats.packet_counter.packets) { |
| observation_complete_.Set(); |
| } |
| return false; |
| } |
| |
| VideoReceiveStream* receive_stream_; |
| uint32_t sent_rtp_; |
| TaskQueueBase* task_queue_ = nullptr; |
| } test; |
| |
| RunBaseTest(&test); |
| } |
| |
| #if defined(WEBRTC_WIN) |
| // Disabled due to flakiness on Windows (bugs.webrtc.org/7483). |
| #define MAYBE_ContentTypeSwitches DISABLED_ContentTypeSwitches |
| #else |
| #define MAYBE_ContentTypeSwitches ContentTypeSwitches |
| #endif |
| TEST_F(StatsEndToEndTest, MAYBE_ContentTypeSwitches) { |
| class StatsObserver : public test::BaseTest, |
| public rtc::VideoSinkInterface<VideoFrame> { |
| public: |
| StatsObserver() : BaseTest(kLongTimeoutMs), num_frames_received_(0) {} |
| |
| bool ShouldCreateReceivers() const override { return true; } |
| |
| void OnFrame(const VideoFrame& video_frame) override { |
| // The RTT is needed to estimate |ntp_time_ms| which is used by |
| // end-to-end delay stats. Therefore, start counting received frames once |
| // |ntp_time_ms| is valid. |
| if (video_frame.ntp_time_ms() > 0 && |
| Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() >= |
| video_frame.ntp_time_ms()) { |
| rtc::CritScope lock(&crit_); |
| ++num_frames_received_; |
| } |
| } |
| |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| if (MinNumberOfFramesReceived()) |
| observation_complete_.Set(); |
| return SEND_PACKET; |
| } |
| |
| bool MinNumberOfFramesReceived() const { |
| // Have some room for frames with wrong content type during switch. |
| const int kMinRequiredHistogramSamples = 200 + 50; |
| rtc::CritScope lock(&crit_); |
| return num_frames_received_ > kMinRequiredHistogramSamples; |
| } |
| |
| // May be called several times. |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) << "Timed out waiting for enough packets."; |
| // Reset frame counter so next PerformTest() call will do something. |
| { |
| rtc::CritScope lock(&crit_); |
| num_frames_received_ = 0; |
| } |
| } |
| |
| rtc::CriticalSection crit_; |
| int num_frames_received_ RTC_GUARDED_BY(&crit_); |
| } test; |
| |
| metrics::Reset(); |
| |
| Call::Config send_config(send_event_log_.get()); |
| test.ModifySenderBitrateConfig(&send_config.bitrate_config); |
| Call::Config recv_config(recv_event_log_.get()); |
| test.ModifyReceiverBitrateConfig(&recv_config.bitrate_config); |
| |
| VideoEncoderConfig encoder_config_with_screenshare; |
| |
| SendTask( |
| RTC_FROM_HERE, task_queue(), |
| [this, &test, &send_config, &recv_config, |
| &encoder_config_with_screenshare]() { |
| CreateSenderCall(send_config); |
| CreateReceiverCall(recv_config); |
| |
| receive_transport_ = test.CreateReceiveTransport(task_queue()); |
| send_transport_ = |
| test.CreateSendTransport(task_queue(), sender_call_.get()); |
| send_transport_->SetReceiver(receiver_call_->Receiver()); |
| receive_transport_->SetReceiver(sender_call_->Receiver()); |
| |
| receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp); |
| CreateSendConfig(1, 0, 0, send_transport_.get()); |
| CreateMatchingReceiveConfigs(receive_transport_.get()); |
| |
| // Modify send and receive configs. |
| GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| video_receive_configs_[0].renderer = &test; |
| // RTT needed for RemoteNtpTimeEstimator for the receive stream. |
| video_receive_configs_[0].rtp.rtcp_xr.receiver_reference_time_report = |
| true; |
| // Start with realtime video. |
| GetVideoEncoderConfig()->content_type = |
| VideoEncoderConfig::ContentType::kRealtimeVideo; |
| // Encoder config for the second part of the test uses screenshare. |
| encoder_config_with_screenshare = GetVideoEncoderConfig()->Copy(); |
| encoder_config_with_screenshare.content_type = |
| VideoEncoderConfig::ContentType::kScreen; |
| |
| CreateVideoStreams(); |
| CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth, |
| kDefaultHeight); |
| Start(); |
| }); |
| |
| test.PerformTest(); |
| |
| // Replace old send stream. |
| SendTask(RTC_FROM_HERE, task_queue(), |
| [this, &encoder_config_with_screenshare]() { |
| DestroyVideoSendStreams(); |
| CreateVideoSendStream(encoder_config_with_screenshare); |
| SetVideoDegradation(DegradationPreference::BALANCED); |
| GetVideoSendStream()->Start(); |
| }); |
| |
| // Continue to run test but now with screenshare. |
| test.PerformTest(); |
| |
| SendTask(RTC_FROM_HERE, task_queue(), [this]() { |
| Stop(); |
| DestroyStreams(); |
| send_transport_.reset(); |
| receive_transport_.reset(); |
| DestroyCalls(); |
| }); |
| |
| // Verify that stats have been updated for both screenshare and video. |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayInMs")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayInMs")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayMaxInMs")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayMaxInMs")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.InterframeDelayInMs")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.Screenshare.InterframeDelayInMs")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples("WebRTC.Video.InterframeDelayMaxInMs")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples( |
| "WebRTC.Video.Screenshare.InterframeDelayMaxInMs")); |
| } |
| |
| TEST_F(StatsEndToEndTest, VerifyNackStats) { |
| static const int kPacketNumberToDrop = 200; |
| class NackObserver : public test::EndToEndTest, public QueuedTask { |
| public: |
| NackObserver() |
| : EndToEndTest(kLongTimeoutMs), |
| sent_rtp_packets_(0), |
| dropped_rtp_packet_(0), |
| dropped_rtp_packet_requested_(false), |
| send_stream_(nullptr) {} |
| |
| private: |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| rtc::CritScope lock(&crit_); |
| if (++sent_rtp_packets_ == kPacketNumberToDrop) { |
| std::unique_ptr<RtpHeaderParser> parser( |
| RtpHeaderParser::CreateForTest()); |
| RTPHeader header; |
| EXPECT_TRUE(parser->Parse(packet, length, &header)); |
| dropped_rtp_packet_ = header.sequenceNumber; |
| return DROP_PACKET; |
| } |
| task_queue_->PostTask(std::unique_ptr<QueuedTask>(this)); |
| return SEND_PACKET; |
| } |
| |
| Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { |
| rtc::CritScope lock(&crit_); |
| test::RtcpPacketParser rtcp_parser; |
| rtcp_parser.Parse(packet, length); |
| const std::vector<uint16_t>& nacks = rtcp_parser.nack()->packet_ids(); |
| if (!nacks.empty() && absl::c_linear_search(nacks, dropped_rtp_packet_)) { |
| dropped_rtp_packet_requested_ = true; |
| } |
| return SEND_PACKET; |
| } |
| |
| void VerifyStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(&crit_) { |
| if (!dropped_rtp_packet_requested_) |
| return; |
| int send_stream_nack_packets = 0; |
| int receive_stream_nack_packets = 0; |
| VideoSendStream::Stats stats = send_stream_->GetStats(); |
| for (const auto& kv : stats.substreams) { |
| const VideoSendStream::StreamStats& stream_stats = kv.second; |
| send_stream_nack_packets += |
| stream_stats.rtcp_packet_type_counts.nack_packets; |
| } |
| for (const auto& receive_stream : receive_streams_) { |
| VideoReceiveStream::Stats stats = receive_stream->GetStats(); |
| receive_stream_nack_packets += |
| stats.rtcp_packet_type_counts.nack_packets; |
| } |
| if (send_stream_nack_packets >= 1 && receive_stream_nack_packets >= 1) { |
| // NACK packet sent on receive stream and received on sent stream. |
| if (MinMetricRunTimePassed()) |
| observation_complete_.Set(); |
| } |
| } |
| |
| bool MinMetricRunTimePassed() { |
| int64_t now_ms = Clock::GetRealTimeClock()->TimeInMilliseconds(); |
| if (!start_runtime_ms_) |
| start_runtime_ms_ = now_ms; |
| |
| int64_t elapsed_sec = (now_ms - *start_runtime_ms_) / 1000; |
| return elapsed_sec > metrics::kMinRunTimeInSeconds; |
| } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStream::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| (*receive_configs)[0].renderer = &fake_renderer_; |
| } |
| |
| void OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& receive_streams) override { |
| send_stream_ = send_stream; |
| receive_streams_ = receive_streams; |
| task_queue_ = TaskQueueBase::Current(); |
| EXPECT_TRUE(task_queue_ != nullptr); |
| } |
| |
| bool Run() override { |
| rtc::CritScope lock(&crit_); |
| VerifyStats(); |
| return false; |
| } |
| |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed."; |
| } |
| |
| test::FakeVideoRenderer fake_renderer_; |
| rtc::CriticalSection crit_; |
| uint64_t sent_rtp_packets_; |
| uint16_t dropped_rtp_packet_ RTC_GUARDED_BY(&crit_); |
| bool dropped_rtp_packet_requested_ RTC_GUARDED_BY(&crit_); |
| std::vector<VideoReceiveStream*> receive_streams_; |
| VideoSendStream* send_stream_; |
| absl::optional<int64_t> start_runtime_ms_; |
| TaskQueueBase* task_queue_ = nullptr; |
| } test; |
| |
| metrics::Reset(); |
| RunBaseTest(&test); |
| |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.UniqueNackRequestsSentInPercent")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples( |
| "WebRTC.Video.UniqueNackRequestsReceivedInPercent")); |
| EXPECT_METRIC_GT(metrics::MinSample("WebRTC.Video.NackPacketsSentPerMinute"), |
| 0); |
| } |
| |
| TEST_F(StatsEndToEndTest, CallReportsRttForSender) { |
| static const int kSendDelayMs = 30; |
| static const int kReceiveDelayMs = 70; |
| |
| std::unique_ptr<test::DirectTransport> sender_transport; |
| std::unique_ptr<test::DirectTransport> receiver_transport; |
| |
| SendTask(RTC_FROM_HERE, task_queue(), |
| [this, &sender_transport, &receiver_transport]() { |
| BuiltInNetworkBehaviorConfig config; |
| config.queue_delay_ms = kSendDelayMs; |
| CreateCalls(); |
| sender_transport = std::make_unique<test::DirectTransport>( |
| task_queue(), |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>(config)), |
| sender_call_.get(), payload_type_map_); |
| config.queue_delay_ms = kReceiveDelayMs; |
| receiver_transport = std::make_unique<test::DirectTransport>( |
| task_queue(), |
| std::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| std::make_unique<SimulatedNetwork>(config)), |
| receiver_call_.get(), payload_type_map_); |
| sender_transport->SetReceiver(receiver_call_->Receiver()); |
| receiver_transport->SetReceiver(sender_call_->Receiver()); |
| |
| CreateSendConfig(1, 0, 0, sender_transport.get()); |
| CreateMatchingReceiveConfigs(receiver_transport.get()); |
| |
| CreateVideoStreams(); |
| CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth, |
| kDefaultHeight); |
| Start(); |
| }); |
| |
| int64_t start_time_ms = clock_->TimeInMilliseconds(); |
| while (true) { |
| Call::Stats stats; |
| SendTask(RTC_FROM_HERE, task_queue(), |
| [this, &stats]() { stats = sender_call_->GetStats(); }); |
| ASSERT_GE(start_time_ms + kDefaultTimeoutMs, clock_->TimeInMilliseconds()) |
| << "No RTT stats before timeout!"; |
| if (stats.rtt_ms != -1) { |
| // To avoid failures caused by rounding or minor ntp clock adjustments, |
| // relax expectation by 1ms. |
| constexpr int kAllowedErrorMs = 1; |
| EXPECT_GE(stats.rtt_ms, kSendDelayMs + kReceiveDelayMs - kAllowedErrorMs); |
| break; |
| } |
| SleepMs(10); |
| } |
| |
| SendTask(RTC_FROM_HERE, task_queue(), |
| [this, &sender_transport, &receiver_transport]() { |
| Stop(); |
| DestroyStreams(); |
| sender_transport.reset(); |
| receiver_transport.reset(); |
| DestroyCalls(); |
| }); |
| } |
| } // namespace webrtc |
