| /* |
| * 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 "absl/types/optional.h" |
| #include "api/test/video/function_video_encoder_factory.h" |
| #include "modules/video_coding/codecs/vp8/include/vp8.h" |
| #include "rtc_base/synchronization/mutex.h" |
| #include "system_wrappers/include/metrics.h" |
| #include "test/call_test.h" |
| #include "test/gtest.h" |
| |
| namespace webrtc { |
| namespace { |
| enum : int { // The first valid value is 1. |
| kTransportSequenceNumberExtensionId = 1, |
| kVideoContentTypeExtensionId, |
| }; |
| } // namespace |
| |
| class HistogramTest : public test::CallTest { |
| public: |
| HistogramTest() { |
| RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
| kTransportSequenceNumberExtensionId)); |
| RegisterRtpExtension(RtpExtension(RtpExtension::kVideoContentTypeUri, |
| kVideoContentTypeExtensionId)); |
| } |
| |
| protected: |
| void VerifyHistogramStats(bool use_rtx, bool use_fec, bool screenshare); |
| }; |
| |
| void HistogramTest::VerifyHistogramStats(bool use_rtx, |
| bool use_fec, |
| bool screenshare) { |
| class FrameObserver : public test::EndToEndTest, |
| public rtc::VideoSinkInterface<VideoFrame> { |
| public: |
| FrameObserver(bool use_rtx, bool use_fec, bool screenshare) |
| : EndToEndTest(kLongTimeoutMs), |
| use_rtx_(use_rtx), |
| use_fec_(use_fec), |
| screenshare_(screenshare), |
| // This test uses NACK, so to send FEC we can't use a fake encoder. |
| encoder_factory_([]() { return VP8Encoder::Create(); }), |
| num_frames_received_(0) {} |
| |
| private: |
| 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()) { |
| MutexLock lock(&mutex_); |
| ++num_frames_received_; |
| } |
| } |
| |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| if (MinMetricRunTimePassed() && MinNumberOfFramesReceived()) |
| observation_complete_.Set(); |
| |
| return SEND_PACKET; |
| } |
| |
| 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 * 2; |
| } |
| |
| bool MinNumberOfFramesReceived() const { |
| const int kMinRequiredHistogramSamples = 200; |
| MutexLock lock(&mutex_); |
| return num_frames_received_ > kMinRequiredHistogramSamples; |
| } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStreamInterface::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| // NACK |
| send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| (*receive_configs)[0].renderer = this; |
| // FEC |
| if (use_fec_) { |
| send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType; |
| send_config->rtp.ulpfec.red_payload_type = kRedPayloadType; |
| send_config->encoder_settings.encoder_factory = &encoder_factory_; |
| send_config->rtp.payload_name = "VP8"; |
| encoder_config->codec_type = kVideoCodecVP8; |
| (*receive_configs)[0].decoders[0].video_format = SdpVideoFormat("VP8"); |
| (*receive_configs)[0].rtp.red_payload_type = kRedPayloadType; |
| (*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType; |
| } |
| // RTX |
| if (use_rtx_) { |
| send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]); |
| send_config->rtp.rtx.payload_type = kSendRtxPayloadType; |
| (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0]; |
| (*receive_configs)[0] |
| .rtp.rtx_associated_payload_types[kSendRtxPayloadType] = |
| kFakeVideoSendPayloadType; |
| if (use_fec_) { |
| send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType; |
| (*receive_configs)[0] |
| .rtp.rtx_associated_payload_types[kRtxRedPayloadType] = |
| kSendRtxPayloadType; |
| } |
| } |
| // RTT needed for RemoteNtpTimeEstimator for the receive stream. |
| (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true; |
| encoder_config->content_type = |
| screenshare_ ? VideoEncoderConfig::ContentType::kScreen |
| : VideoEncoderConfig::ContentType::kRealtimeVideo; |
| } |
| |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) << "Timed out waiting for min frames to be received."; |
| } |
| |
| mutable Mutex mutex_; |
| const bool use_rtx_; |
| const bool use_fec_; |
| const bool screenshare_; |
| test::FunctionVideoEncoderFactory encoder_factory_; |
| absl::optional<int64_t> start_runtime_ms_; |
| int num_frames_received_ RTC_GUARDED_BY(&mutex_); |
| } test(use_rtx, use_fec, screenshare); |
| |
| metrics::Reset(); |
| RunBaseTest(&test); |
| |
| const std::string video_prefix = |
| screenshare ? "WebRTC.Video.Screenshare." : "WebRTC.Video."; |
| // The content type extension is disabled in non screenshare test, |
| // therefore no slicing on simulcast id should be present. |
| const std::string video_suffix = screenshare ? ".S0" : ""; |
| |
| // Verify that stats have been updated once. |
| EXPECT_METRIC_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples( |
| "WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Call.EstimatedSendBitrateInKbps")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Call.PacerBitrateInKbps")); |
| |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.SendStreamLifetimeInSeconds")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.ReceiveStreamLifetimeInSeconds")); |
| |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.NackPacketsSentPerMinute")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples(video_prefix + "NackPacketsReceivedPerMinute")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples("WebRTC.Video.FirPacketsSentPerMinute")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples(video_prefix + "FirPacketsReceivedPerMinute")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples("WebRTC.Video.PliPacketsSentPerMinute")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples(video_prefix + "PliPacketsReceivedPerMinute")); |
| |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples(video_prefix + "KeyFramesSentInPermille")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.KeyFramesReceivedInPermille")); |
| |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples(video_prefix + "SentPacketsLostInPercent")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.ReceivedPacketsLostInPercent")); |
| |
| EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "InputWidthInPixels")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples(video_prefix + "InputHeightInPixels")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "SentWidthInPixels")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "SentHeightInPixels")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples(video_prefix + "ReceivedWidthInPixels")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples(video_prefix + "ReceivedHeightInPixels")); |
| |
| EXPECT_METRIC_EQ(1, metrics::NumEvents(video_prefix + "InputWidthInPixels", |
| kDefaultWidth)); |
| EXPECT_METRIC_EQ(1, metrics::NumEvents(video_prefix + "InputHeightInPixels", |
| kDefaultHeight)); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumEvents(video_prefix + "SentWidthInPixels", kDefaultWidth)); |
| EXPECT_METRIC_EQ(1, metrics::NumEvents(video_prefix + "SentHeightInPixels", |
| kDefaultHeight)); |
| EXPECT_METRIC_EQ(1, metrics::NumEvents(video_prefix + "ReceivedWidthInPixels", |
| kDefaultWidth)); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumEvents(video_prefix + "ReceivedHeightInPixels", |
| kDefaultHeight)); |
| |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples(video_prefix + "InputFramesPerSecond")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples(video_prefix + "SentFramesPerSecond")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples("WebRTC.Video.RenderFramesPerSecond")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer")); |
| |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples("WebRTC.Video.JitterBufferDelayInMs")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.TargetDelayInMs")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.CurrentDelayInMs")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.OnewayDelayInMs")); |
| |
| EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs" + |
| video_suffix)); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs" + |
| video_suffix)); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayInMs" + |
| video_suffix)); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples(video_prefix + "InterframeDelayMaxInMs" + |
| video_suffix)); |
| |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.RenderSqrtPixelsPerSecond")); |
| |
| EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "EncodeTimeInMs")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.DecodeTimeInMs")); |
| |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples(video_prefix + "NumberOfPauseEvents")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples(video_prefix + "PausedTimeInPercent")); |
| |
| EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "BitrateSentInKbps")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples("WebRTC.Video.BitrateReceivedInKbps")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples(video_prefix + "MediaBitrateSentInKbps")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.MediaBitrateReceivedInKbps")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples(video_prefix + "PaddingBitrateSentInKbps")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples("WebRTC.Video.PaddingBitrateReceivedInKbps")); |
| EXPECT_METRIC_EQ( |
| 1, metrics::NumSamples(video_prefix + "RetransmittedBitrateSentInKbps")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples( |
| "WebRTC.Video.RetransmittedBitrateReceivedInKbps")); |
| |
| EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.SendDelayInMs")); |
| EXPECT_METRIC_EQ(1, metrics::NumSamples(video_prefix + "SendSideDelayInMs")); |
| EXPECT_METRIC_EQ(1, |
| metrics::NumSamples(video_prefix + "SendSideDelayMaxInMs")); |
| |
| int num_rtx_samples = use_rtx ? 1 : 0; |
| EXPECT_METRIC_EQ(num_rtx_samples, |
| metrics::NumSamples("WebRTC.Video.RtxBitrateSentInKbps")); |
| EXPECT_METRIC_EQ( |
| num_rtx_samples, |
| metrics::NumSamples("WebRTC.Video.RtxBitrateReceivedInKbps")); |
| |
| int num_red_samples = use_fec ? 1 : 0; |
| EXPECT_METRIC_EQ(num_red_samples, |
| metrics::NumSamples("WebRTC.Video.FecBitrateSentInKbps")); |
| EXPECT_METRIC_EQ( |
| num_red_samples, |
| metrics::NumSamples("WebRTC.Video.FecBitrateReceivedInKbps")); |
| EXPECT_METRIC_EQ( |
| num_red_samples, |
| metrics::NumSamples("WebRTC.Video.ReceivedFecPacketsInPercent")); |
| } |
| |
| TEST_F(HistogramTest, VerifyStatsWithRtx) { |
| const bool kEnabledRtx = true; |
| const bool kEnabledRed = false; |
| const bool kScreenshare = false; |
| VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare); |
| } |
| |
| TEST_F(HistogramTest, VerifyStatsWithRed) { |
| const bool kEnabledRtx = false; |
| const bool kEnabledRed = true; |
| const bool kScreenshare = false; |
| VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare); |
| } |
| |
| TEST_F(HistogramTest, VerifyStatsWithScreenshare) { |
| const bool kEnabledRtx = false; |
| const bool kEnabledRed = false; |
| const bool kScreenshare = true; |
| VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare); |
| } |
| |
| } // namespace webrtc |
