/* * 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 "video/receive_statistics_proxy.h" #include #include #include #include #include "modules/pacing/alr_detector.h" #include "modules/video_coding/include/video_codec_interface.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "system_wrappers/include/clock.h" #include "system_wrappers/include/metrics.h" namespace webrtc { namespace { // Periodic time interval for processing samples for |freq_offset_counter_|. const int64_t kFreqOffsetProcessIntervalMs = 40000; // Configuration for bad call detection. const int kBadCallMinRequiredSamples = 10; const int kMinSampleLengthMs = 990; const int kNumMeasurements = 10; const int kNumMeasurementsVariance = kNumMeasurements * 1.5; const float kBadFraction = 0.8f; // For fps: // Low means low enough to be bad, high means high enough to be good const int kLowFpsThreshold = 12; const int kHighFpsThreshold = 14; // For qp and fps variance: // Low means low enough to be good, high means high enough to be bad const int kLowQpThresholdVp8 = 60; const int kHighQpThresholdVp8 = 70; const int kLowVarianceThreshold = 1; const int kHighVarianceThreshold = 2; // Some metrics are reported as a maximum over this period. // This should be synchronized with a typical getStats polling interval in // the clients. const int kMovingMaxWindowMs = 1000; // How large window we use to calculate the framerate/bitrate. const int kRateStatisticsWindowSizeMs = 1000; // Some sane ballpark estimate for maximum common value of inter-frame delay. // Values below that will be stored explicitly in the array, // values above - in the map. const int kMaxCommonInterframeDelayMs = 500; std::string UmaPrefixForContentType(VideoContentType content_type) { std::stringstream ss; ss << "WebRTC.Video"; if (videocontenttypehelpers::IsScreenshare(content_type)) { ss << ".Screenshare"; } return ss.str(); } std::string UmaSuffixForContentType(VideoContentType content_type) { std::stringstream ss; int simulcast_id = videocontenttypehelpers::GetSimulcastId(content_type); if (simulcast_id > 0) { ss << ".S" << simulcast_id - 1; } int experiment_id = videocontenttypehelpers::GetExperimentId(content_type); if (experiment_id > 0) { ss << ".ExperimentGroup" << experiment_id - 1; } return ss.str(); } } // namespace ReceiveStatisticsProxy::ReceiveStatisticsProxy( const VideoReceiveStream::Config* config, Clock* clock) : clock_(clock), config_(*config), start_ms_(clock->TimeInMilliseconds()), last_sample_time_(clock->TimeInMilliseconds()), fps_threshold_(kLowFpsThreshold, kHighFpsThreshold, kBadFraction, kNumMeasurements), qp_threshold_(kLowQpThresholdVp8, kHighQpThresholdVp8, kBadFraction, kNumMeasurements), variance_threshold_(kLowVarianceThreshold, kHighVarianceThreshold, kBadFraction, kNumMeasurementsVariance), num_bad_states_(0), num_certain_states_(0), // 1000ms window, scale 1000 for ms to s. decode_fps_estimator_(1000, 1000), renders_fps_estimator_(1000, 1000), render_fps_tracker_(100, 10u), render_pixel_tracker_(100, 10u), total_byte_tracker_(100, 10u), // bucket_interval_ms, bucket_count interframe_delay_max_moving_(kMovingMaxWindowMs), freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs), first_report_block_time_ms_(-1), avg_rtt_ms_(0), last_content_type_(VideoContentType::UNSPECIFIED), timing_frame_info_counter_(kMovingMaxWindowMs) { stats_.ssrc = config_.rtp.remote_ssrc; // TODO(brandtr): Replace |rtx_stats_| with a single instance of // StreamDataCounters. if (config_.rtp.rtx_ssrc) { rtx_stats_[config_.rtp.rtx_ssrc] = StreamDataCounters(); } } ReceiveStatisticsProxy::~ReceiveStatisticsProxy() { UpdateHistograms(); } void ReceiveStatisticsProxy::UpdateHistograms() { int stream_duration_sec = (clock_->TimeInMilliseconds() - start_ms_) / 1000; if (stats_.frame_counts.key_frames > 0 || stats_.frame_counts.delta_frames > 0) { RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.ReceiveStreamLifetimeInSeconds", stream_duration_sec); RTC_LOG(LS_INFO) << "WebRTC.Video.ReceiveStreamLifetimeInSeconds " << stream_duration_sec; } if (first_report_block_time_ms_ != -1 && ((clock_->TimeInMilliseconds() - first_report_block_time_ms_) / 1000) >= metrics::kMinRunTimeInSeconds) { int fraction_lost = report_block_stats_.FractionLostInPercent(); if (fraction_lost != -1) { RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent", fraction_lost); RTC_LOG(LS_INFO) << "WebRTC.Video.ReceivedPacketsLostInPercent " << fraction_lost; } } const int kMinRequiredSamples = 200; int samples = static_cast(render_fps_tracker_.TotalSampleCount()); if (samples >= kMinRequiredSamples) { RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond", round(render_fps_tracker_.ComputeTotalRate())); RTC_HISTOGRAM_COUNTS_100000( "WebRTC.Video.RenderSqrtPixelsPerSecond", round(render_pixel_tracker_.ComputeTotalRate())); } int sync_offset_ms = sync_offset_counter_.Avg(kMinRequiredSamples); if (sync_offset_ms != -1) { RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs", sync_offset_ms); RTC_LOG(LS_INFO) << "WebRTC.Video.AVSyncOffsetInMs " << sync_offset_ms; } AggregatedStats freq_offset_stats = freq_offset_counter_.GetStats(); if (freq_offset_stats.num_samples > 0) { RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtpToNtpFreqOffsetInKhz", freq_offset_stats.average); RTC_LOG(LS_INFO) << "WebRTC.Video.RtpToNtpFreqOffsetInKhz, " << freq_offset_stats.ToString(); } int num_total_frames = stats_.frame_counts.key_frames + stats_.frame_counts.delta_frames; if (num_total_frames >= kMinRequiredSamples) { int num_key_frames = stats_.frame_counts.key_frames; int key_frames_permille = (num_key_frames * 1000 + num_total_frames / 2) / num_total_frames; RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille", key_frames_permille); RTC_LOG(LS_INFO) << "WebRTC.Video.KeyFramesReceivedInPermille " << key_frames_permille; } int qp = qp_counters_.vp8.Avg(kMinRequiredSamples); if (qp != -1) { RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp); RTC_LOG(LS_INFO) << "WebRTC.Video.Decoded.Vp8.Qp " << qp; } int decode_ms = decode_time_counter_.Avg(kMinRequiredSamples); if (decode_ms != -1) { RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms); RTC_LOG(LS_INFO) << "WebRTC.Video.DecodeTimeInMs " << decode_ms; } int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredSamples); if (jb_delay_ms != -1) { RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs", jb_delay_ms); RTC_LOG(LS_INFO) << "WebRTC.Video.JitterBufferDelayInMs " << jb_delay_ms; } int target_delay_ms = target_delay_counter_.Avg(kMinRequiredSamples); if (target_delay_ms != -1) { RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms); RTC_LOG(LS_INFO) << "WebRTC.Video.TargetDelayInMs " << target_delay_ms; } int current_delay_ms = current_delay_counter_.Avg(kMinRequiredSamples); if (current_delay_ms != -1) { RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs", current_delay_ms); RTC_LOG(LS_INFO) << "WebRTC.Video.CurrentDelayInMs " << current_delay_ms; } int delay_ms = delay_counter_.Avg(kMinRequiredSamples); if (delay_ms != -1) RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms); // Aggregate content_specific_stats_ by removing experiment or simulcast // information; std::map aggregated_stats; for (auto it : content_specific_stats_) { // Calculate simulcast specific metrics (".S0" ... ".S2" suffixes). VideoContentType content_type = it.first; if (videocontenttypehelpers::GetSimulcastId(content_type) > 0) { // Aggregate on experiment id. videocontenttypehelpers::SetExperimentId(&content_type, 0); aggregated_stats[content_type].Add(it.second); } // Calculate experiment specific metrics (".ExperimentGroup[0-7]" suffixes). content_type = it.first; if (videocontenttypehelpers::GetExperimentId(content_type) > 0) { // Aggregate on simulcast id. videocontenttypehelpers::SetSimulcastId(&content_type, 0); aggregated_stats[content_type].Add(it.second); } // Calculate aggregated metrics (no suffixes. Aggregated on everything). content_type = it.first; videocontenttypehelpers::SetSimulcastId(&content_type, 0); videocontenttypehelpers::SetExperimentId(&content_type, 0); aggregated_stats[content_type].Add(it.second); } for (auto it : aggregated_stats) { // For the metric Foo we report the following slices: // WebRTC.Video.Foo, // WebRTC.Video.Screenshare.Foo, // WebRTC.Video.Foo.S[0-3], // WebRTC.Video.Foo.ExperimentGroup[0-7], // WebRTC.Video.Screenshare.Foo.S[0-3], // WebRTC.Video.Screenshare.Foo.ExperimentGroup[0-7]. auto content_type = it.first; auto stats = it.second; std::string uma_prefix = UmaPrefixForContentType(content_type); std::string uma_suffix = UmaSuffixForContentType(content_type); // Metrics can be sliced on either simulcast id or experiment id but not // both. RTC_DCHECK(videocontenttypehelpers::GetExperimentId(content_type) == 0 || videocontenttypehelpers::GetSimulcastId(content_type) == 0); int e2e_delay_ms = stats.e2e_delay_counter.Avg(kMinRequiredSamples); if (e2e_delay_ms != -1) { RTC_HISTOGRAM_COUNTS_SPARSE_10000( uma_prefix + ".EndToEndDelayInMs" + uma_suffix, e2e_delay_ms); RTC_LOG(LS_INFO) << uma_prefix << ".EndToEndDelayInMs" << uma_suffix << " " << e2e_delay_ms; } int e2e_delay_max_ms = stats.e2e_delay_counter.Max(); if (e2e_delay_max_ms != -1 && e2e_delay_ms != -1) { RTC_HISTOGRAM_COUNTS_SPARSE_100000( uma_prefix + ".EndToEndDelayMaxInMs" + uma_suffix, e2e_delay_max_ms); RTC_LOG(LS_INFO) << uma_prefix << ".EndToEndDelayMaxInMs" << uma_suffix << " " << e2e_delay_max_ms; } int interframe_delay_ms = stats.interframe_delay_counter.Avg(kMinRequiredSamples); if (interframe_delay_ms != -1) { RTC_HISTOGRAM_COUNTS_SPARSE_10000( uma_prefix + ".InterframeDelayInMs" + uma_suffix, interframe_delay_ms); RTC_LOG(LS_INFO) << uma_prefix << ".InterframeDelayInMs" << uma_suffix << " " << interframe_delay_ms; } int interframe_delay_max_ms = stats.interframe_delay_counter.Max(); if (interframe_delay_max_ms != -1 && interframe_delay_ms != -1) { RTC_HISTOGRAM_COUNTS_SPARSE_10000( uma_prefix + ".InterframeDelayMaxInMs" + uma_suffix, interframe_delay_max_ms); RTC_LOG(LS_INFO) << uma_prefix << ".InterframeDelayMaxInMs" << uma_suffix << " " << interframe_delay_max_ms; } rtc::Optional interframe_delay_95p_ms = stats.interframe_delay_percentiles.GetPercentile(0.95f); if (interframe_delay_95p_ms && interframe_delay_ms != -1) { RTC_HISTOGRAM_COUNTS_SPARSE_10000( uma_prefix + ".InterframeDelay95PercentileInMs" + uma_suffix, *interframe_delay_95p_ms); RTC_LOG(LS_INFO) << uma_prefix << ".InterframeDelay95PercentileInMs" << uma_suffix << " " << *interframe_delay_95p_ms; } int width = stats.received_width.Avg(kMinRequiredSamples); if (width != -1) { RTC_HISTOGRAM_COUNTS_SPARSE_10000( uma_prefix + ".ReceivedWidthInPixels" + uma_suffix, width); RTC_LOG(LS_INFO) << uma_prefix << ".ReceivedWidthInPixels" << uma_suffix << " " << width; } int height = stats.received_height.Avg(kMinRequiredSamples); if (height != -1) { RTC_HISTOGRAM_COUNTS_SPARSE_10000( uma_prefix + ".ReceivedHeightInPixels" + uma_suffix, height); RTC_LOG(LS_INFO) << uma_prefix << ".ReceivedHeightInPixels" << uma_suffix << " " << height; } if (content_type != VideoContentType::UNSPECIFIED) { // Don't report these 3 metrics unsliced, as more precise variants // are reported separately in this method. float flow_duration_sec = stats.flow_duration_ms / 1000.0; if (flow_duration_sec >= metrics::kMinRunTimeInSeconds) { int media_bitrate_kbps = static_cast(stats.total_media_bytes * 8 / flow_duration_sec / 1000); RTC_HISTOGRAM_COUNTS_SPARSE_10000( uma_prefix + ".MediaBitrateReceivedInKbps" + uma_suffix, media_bitrate_kbps); RTC_LOG(LS_INFO) << uma_prefix << ".MediaBitrateReceivedInKbps" << uma_suffix << " " << media_bitrate_kbps; } int num_total_frames = stats.frame_counts.key_frames + stats.frame_counts.delta_frames; if (num_total_frames >= kMinRequiredSamples) { int num_key_frames = stats.frame_counts.key_frames; int key_frames_permille = (num_key_frames * 1000 + num_total_frames / 2) / num_total_frames; RTC_HISTOGRAM_COUNTS_SPARSE_1000( uma_prefix + ".KeyFramesReceivedInPermille" + uma_suffix, key_frames_permille); RTC_LOG(LS_INFO) << uma_prefix << ".KeyFramesReceivedInPermille" << uma_suffix << " " << key_frames_permille; } int qp = stats.qp_counter.Avg(kMinRequiredSamples); if (qp != -1) { RTC_HISTOGRAM_COUNTS_SPARSE_200( uma_prefix + ".Decoded.Vp8.Qp" + uma_suffix, qp); RTC_LOG(LS_INFO) << uma_prefix << ".Decoded.Vp8.Qp" << uma_suffix << " " << qp; } } } StreamDataCounters rtp = stats_.rtp_stats; StreamDataCounters rtx; for (auto it : rtx_stats_) rtx.Add(it.second); StreamDataCounters rtp_rtx = rtp; rtp_rtx.Add(rtx); int64_t elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(clock_->TimeInMilliseconds()) / 1000; if (elapsed_sec >= metrics::kMinRunTimeInSeconds) { RTC_HISTOGRAM_COUNTS_10000( "WebRTC.Video.BitrateReceivedInKbps", static_cast(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000)); int media_bitrate_kbs = static_cast(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000); RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps", media_bitrate_kbs); RTC_LOG(LS_INFO) << "WebRTC.Video.MediaBitrateReceivedInKbps " << media_bitrate_kbs; RTC_HISTOGRAM_COUNTS_10000( "WebRTC.Video.PaddingBitrateReceivedInKbps", static_cast(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / 1000)); RTC_HISTOGRAM_COUNTS_10000( "WebRTC.Video.RetransmittedBitrateReceivedInKbps", static_cast(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec / 1000)); if (!rtx_stats_.empty()) { RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps", static_cast(rtx.transmitted.TotalBytes() * 8 / elapsed_sec / 1000)); } if (config_.rtp.ulpfec_payload_type != -1) { RTC_HISTOGRAM_COUNTS_10000( "WebRTC.Video.FecBitrateReceivedInKbps", static_cast(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000)); } const RtcpPacketTypeCounter& counters = stats_.rtcp_packet_type_counts; RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute", counters.nack_packets * 60 / elapsed_sec); RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute", counters.fir_packets * 60 / elapsed_sec); RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute", counters.pli_packets * 60 / elapsed_sec); if (counters.nack_requests > 0) { RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent", counters.UniqueNackRequestsInPercent()); } } if (num_certain_states_ >= kBadCallMinRequiredSamples) { RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.Any", 100 * num_bad_states_ / num_certain_states_); } rtc::Optional fps_fraction = fps_threshold_.FractionHigh(kBadCallMinRequiredSamples); if (fps_fraction) { RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.FrameRate", static_cast(100 * (1 - *fps_fraction))); } rtc::Optional variance_fraction = variance_threshold_.FractionHigh(kBadCallMinRequiredSamples); if (variance_fraction) { RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.FrameRateVariance", static_cast(100 * *variance_fraction)); } rtc::Optional qp_fraction = qp_threshold_.FractionHigh(kBadCallMinRequiredSamples); if (qp_fraction) { RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.Qp", static_cast(100 * *qp_fraction)); } } void ReceiveStatisticsProxy::QualitySample() { int64_t now = clock_->TimeInMilliseconds(); if (last_sample_time_ + kMinSampleLengthMs > now) return; double fps = render_fps_tracker_.ComputeRateForInterval(now - last_sample_time_); int qp = qp_sample_.Avg(1); bool prev_fps_bad = !fps_threshold_.IsHigh().value_or(true); bool prev_qp_bad = qp_threshold_.IsHigh().value_or(false); bool prev_variance_bad = variance_threshold_.IsHigh().value_or(false); bool prev_any_bad = prev_fps_bad || prev_qp_bad || prev_variance_bad; fps_threshold_.AddMeasurement(static_cast(fps)); if (qp != -1) qp_threshold_.AddMeasurement(qp); rtc::Optional fps_variance_opt = fps_threshold_.CalculateVariance(); double fps_variance = fps_variance_opt.value_or(0); if (fps_variance_opt) { variance_threshold_.AddMeasurement(static_cast(fps_variance)); } bool fps_bad = !fps_threshold_.IsHigh().value_or(true); bool qp_bad = qp_threshold_.IsHigh().value_or(false); bool variance_bad = variance_threshold_.IsHigh().value_or(false); bool any_bad = fps_bad || qp_bad || variance_bad; if (!prev_any_bad && any_bad) { RTC_LOG(LS_INFO) << "Bad call (any) start: " << now; } else if (prev_any_bad && !any_bad) { RTC_LOG(LS_INFO) << "Bad call (any) end: " << now; } if (!prev_fps_bad && fps_bad) { RTC_LOG(LS_INFO) << "Bad call (fps) start: " << now; } else if (prev_fps_bad && !fps_bad) { RTC_LOG(LS_INFO) << "Bad call (fps) end: " << now; } if (!prev_qp_bad && qp_bad) { RTC_LOG(LS_INFO) << "Bad call (qp) start: " << now; } else if (prev_qp_bad && !qp_bad) { RTC_LOG(LS_INFO) << "Bad call (qp) end: " << now; } if (!prev_variance_bad && variance_bad) { RTC_LOG(LS_INFO) << "Bad call (variance) start: " << now; } else if (prev_variance_bad && !variance_bad) { RTC_LOG(LS_INFO) << "Bad call (variance) end: " << now; } RTC_LOG(LS_VERBOSE) << "SAMPLE: sample_length: " << (now - last_sample_time_) << " fps: " << fps << " fps_bad: " << fps_bad << " qp: " << qp << " qp_bad: " << qp_bad << " variance_bad: " << variance_bad << " fps_variance: " << fps_variance; last_sample_time_ = now; qp_sample_.Reset(); if (fps_threshold_.IsHigh() || variance_threshold_.IsHigh() || qp_threshold_.IsHigh()) { if (any_bad) ++num_bad_states_; ++num_certain_states_; } } void ReceiveStatisticsProxy::UpdateFramerate(int64_t now_ms) const { int64_t old_frames_ms = now_ms - kRateStatisticsWindowSizeMs; while (!frame_window_.empty() && frame_window_.begin()->first < old_frames_ms) { frame_window_.erase(frame_window_.begin()); } size_t framerate = (frame_window_.size() * 1000 + 500) / kRateStatisticsWindowSizeMs; stats_.network_frame_rate = static_cast(framerate); } VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const { rtc::CritScope lock(&crit_); // Get current frame rates here, as only updating them on new frames prevents // us from ever correctly displaying frame rate of 0. int64_t now_ms = clock_->TimeInMilliseconds(); UpdateFramerate(now_ms); stats_.render_frame_rate = renders_fps_estimator_.Rate(now_ms).value_or(0); stats_.decode_frame_rate = decode_fps_estimator_.Rate(now_ms).value_or(0); stats_.total_bitrate_bps = static_cast(total_byte_tracker_.ComputeRate() * 8); stats_.interframe_delay_max_ms = interframe_delay_max_moving_.Max(now_ms).value_or(-1); stats_.timing_frame_info = timing_frame_info_counter_.Max(now_ms); stats_.content_type = last_content_type_; return stats_; } void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) { rtc::CritScope lock(&crit_); stats_.current_payload_type = payload_type; } void ReceiveStatisticsProxy::OnDecoderImplementationName( const char* implementation_name) { rtc::CritScope lock(&crit_); stats_.decoder_implementation_name = implementation_name; } void ReceiveStatisticsProxy::OnIncomingRate(unsigned int framerate, unsigned int bitrate_bps) { rtc::CritScope lock(&crit_); if (stats_.rtp_stats.first_packet_time_ms != -1) QualitySample(); } void ReceiveStatisticsProxy::OnFrameBufferTimingsUpdated( int decode_ms, int max_decode_ms, int current_delay_ms, int target_delay_ms, int jitter_buffer_ms, int min_playout_delay_ms, int render_delay_ms) { rtc::CritScope lock(&crit_); stats_.decode_ms = decode_ms; stats_.max_decode_ms = max_decode_ms; stats_.current_delay_ms = current_delay_ms; stats_.target_delay_ms = target_delay_ms; stats_.jitter_buffer_ms = jitter_buffer_ms; stats_.min_playout_delay_ms = min_playout_delay_ms; stats_.render_delay_ms = render_delay_ms; decode_time_counter_.Add(decode_ms); jitter_buffer_delay_counter_.Add(jitter_buffer_ms); target_delay_counter_.Add(target_delay_ms); current_delay_counter_.Add(current_delay_ms); // Network delay (rtt/2) + target_delay_ms (jitter delay + decode time + // render delay). delay_counter_.Add(target_delay_ms + avg_rtt_ms_ / 2); } void ReceiveStatisticsProxy::OnTimingFrameInfoUpdated( const TimingFrameInfo& info) { rtc::CritScope lock(&crit_); int64_t now_ms = clock_->TimeInMilliseconds(); timing_frame_info_counter_.Add(info, now_ms); } void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated( uint32_t ssrc, const RtcpPacketTypeCounter& packet_counter) { rtc::CritScope lock(&crit_); if (stats_.ssrc != ssrc) return; stats_.rtcp_packet_type_counts = packet_counter; } void ReceiveStatisticsProxy::StatisticsUpdated( const webrtc::RtcpStatistics& statistics, uint32_t ssrc) { rtc::CritScope lock(&crit_); // TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we // receive stats from one of them. if (stats_.ssrc != ssrc) return; stats_.rtcp_stats = statistics; report_block_stats_.Store(statistics, ssrc, 0); if (first_report_block_time_ms_ == -1) first_report_block_time_ms_ = clock_->TimeInMilliseconds(); } void ReceiveStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) { rtc::CritScope lock(&crit_); // TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we // receive stats from one of them. if (stats_.ssrc != ssrc) return; stats_.c_name = cname; } void ReceiveStatisticsProxy::DataCountersUpdated( const webrtc::StreamDataCounters& counters, uint32_t ssrc) { size_t last_total_bytes = 0; size_t total_bytes = 0; rtc::CritScope lock(&crit_); if (ssrc == stats_.ssrc) { last_total_bytes = stats_.rtp_stats.transmitted.TotalBytes(); total_bytes = counters.transmitted.TotalBytes(); stats_.rtp_stats = counters; } else { auto it = rtx_stats_.find(ssrc); if (it != rtx_stats_.end()) { last_total_bytes = it->second.transmitted.TotalBytes(); total_bytes = counters.transmitted.TotalBytes(); it->second = counters; } else { RTC_NOTREACHED() << "Unexpected stream ssrc: " << ssrc; } } if (total_bytes > last_total_bytes) total_byte_tracker_.AddSamples(total_bytes - last_total_bytes); } void ReceiveStatisticsProxy::OnDecodedFrame(rtc::Optional qp, VideoContentType content_type) { rtc::CritScope lock(&crit_); uint64_t now = clock_->TimeInMilliseconds(); ContentSpecificStats* content_specific_stats = &content_specific_stats_[content_type]; ++stats_.frames_decoded; if (qp) { if (!stats_.qp_sum) { if (stats_.frames_decoded != 1) { RTC_LOG(LS_WARNING) << "Frames decoded was not 1 when first qp value was received."; stats_.frames_decoded = 1; } stats_.qp_sum = rtc::Optional(0); } *stats_.qp_sum += *qp; content_specific_stats->qp_counter.Add(*qp); } else if (stats_.qp_sum) { RTC_LOG(LS_WARNING) << "QP sum was already set and no QP was given for a frame."; stats_.qp_sum = rtc::Optional(); } last_content_type_ = content_type; decode_fps_estimator_.Update(1, now); if (last_decoded_frame_time_ms_) { int64_t interframe_delay_ms = now - *last_decoded_frame_time_ms_; RTC_DCHECK_GE(interframe_delay_ms, 0); interframe_delay_max_moving_.Add(interframe_delay_ms, now); content_specific_stats->interframe_delay_counter.Add(interframe_delay_ms); content_specific_stats->interframe_delay_percentiles.Add( interframe_delay_ms); content_specific_stats->flow_duration_ms += interframe_delay_ms; } last_decoded_frame_time_ms_.emplace(now); } void ReceiveStatisticsProxy::OnRenderedFrame(const VideoFrame& frame) { int width = frame.width(); int height = frame.height(); RTC_DCHECK_GT(width, 0); RTC_DCHECK_GT(height, 0); uint64_t now = clock_->TimeInMilliseconds(); rtc::CritScope lock(&crit_); ContentSpecificStats* content_specific_stats = &content_specific_stats_[last_content_type_]; renders_fps_estimator_.Update(1, now); ++stats_.frames_rendered; stats_.width = width; stats_.height = height; render_fps_tracker_.AddSamples(1); render_pixel_tracker_.AddSamples(sqrt(width * height)); content_specific_stats->received_width.Add(width); content_specific_stats->received_height.Add(height); if (frame.ntp_time_ms() > 0) { int64_t delay_ms = clock_->CurrentNtpInMilliseconds() - frame.ntp_time_ms(); if (delay_ms >= 0) { content_specific_stats->e2e_delay_counter.Add(delay_ms); } } } void ReceiveStatisticsProxy::OnSyncOffsetUpdated(int64_t sync_offset_ms, double estimated_freq_khz) { rtc::CritScope lock(&crit_); sync_offset_counter_.Add(std::abs(sync_offset_ms)); stats_.sync_offset_ms = sync_offset_ms; const double kMaxFreqKhz = 10000.0; int offset_khz = kMaxFreqKhz; // Should not be zero or negative. If so, report max. if (estimated_freq_khz < kMaxFreqKhz && estimated_freq_khz > 0.0) offset_khz = static_cast(std::fabs(estimated_freq_khz - 90.0) + 0.5); freq_offset_counter_.Add(offset_khz); } void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate, uint32_t frameRate) { } void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe, size_t size_bytes, VideoContentType content_type) { rtc::CritScope lock(&crit_); if (is_keyframe) { ++stats_.frame_counts.key_frames; } else { ++stats_.frame_counts.delta_frames; } ContentSpecificStats* content_specific_stats = &content_specific_stats_[content_type]; content_specific_stats->total_media_bytes += size_bytes; if (is_keyframe) { ++content_specific_stats->frame_counts.key_frames; } else { ++content_specific_stats->frame_counts.delta_frames; } int64_t now_ms = clock_->TimeInMilliseconds(); frame_window_.insert(std::make_pair(now_ms, size_bytes)); UpdateFramerate(now_ms); } void ReceiveStatisticsProxy::OnFrameCountsUpdated( const FrameCounts& frame_counts) { rtc::CritScope lock(&crit_); stats_.frame_counts = frame_counts; } void ReceiveStatisticsProxy::OnDiscardedPacketsUpdated(int discarded_packets) { rtc::CritScope lock(&crit_); stats_.discarded_packets = discarded_packets; } void ReceiveStatisticsProxy::OnPreDecode( const EncodedImage& encoded_image, const CodecSpecificInfo* codec_specific_info) { if (!codec_specific_info || encoded_image.qp_ == -1) { return; } if (codec_specific_info->codecType == kVideoCodecVP8) { qp_counters_.vp8.Add(encoded_image.qp_); rtc::CritScope lock(&crit_); qp_sample_.Add(encoded_image.qp_); } } void ReceiveStatisticsProxy::OnStreamInactive() { // TODO(sprang): Figure out any other state that should be reset. rtc::CritScope lock(&crit_); // Don't report inter-frame delay if stream was paused. last_decoded_frame_time_ms_.reset(); } void ReceiveStatisticsProxy::SampleCounter::Add(int sample) { sum += sample; ++num_samples; if (!max || sample > *max) { max.emplace(sample); } } void ReceiveStatisticsProxy::SampleCounter::Add(const SampleCounter& other) { sum += other.sum; num_samples += other.num_samples; if (other.max && (!max || *max < *other.max)) max = other.max; } int ReceiveStatisticsProxy::SampleCounter::Avg( int64_t min_required_samples) const { if (num_samples < min_required_samples || num_samples == 0) return -1; return static_cast(sum / num_samples); } int ReceiveStatisticsProxy::SampleCounter::Max() const { return max.value_or(-1); } void ReceiveStatisticsProxy::SampleCounter::Reset() { num_samples = 0; sum = 0; max.reset(); } void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { rtc::CritScope lock(&crit_); avg_rtt_ms_ = avg_rtt_ms; } ReceiveStatisticsProxy::ContentSpecificStats::ContentSpecificStats() : interframe_delay_percentiles(kMaxCommonInterframeDelayMs) {} void ReceiveStatisticsProxy::ContentSpecificStats::Add( const ContentSpecificStats& other) { e2e_delay_counter.Add(other.e2e_delay_counter); interframe_delay_counter.Add(other.interframe_delay_counter); flow_duration_ms += other.flow_duration_ms; total_media_bytes += other.total_media_bytes; received_height.Add(other.received_height); received_width.Add(other.received_width); qp_counter.Add(other.qp_counter); frame_counts.key_frames += other.frame_counts.key_frames; frame_counts.delta_frames += other.frame_counts.delta_frames; interframe_delay_percentiles.Add(other.interframe_delay_percentiles); } } // namespace webrtc