| /* |
| * 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 "webrtc/video/receive_statistics_proxy.h" |
| |
| #include <algorithm> |
| #include <cmath> |
| #include <sstream> |
| #include <utility> |
| |
| #include "webrtc/modules/pacing/alr_detector.h" |
| #include "webrtc/modules/video_coding/include/video_codec_interface.h" |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/rtc_base/logging.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| #include "webrtc/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. |
| const int kMovingMaxWindowMs = 10000; |
| |
| // How large window we use to calculate the framerate/bitrate. |
| const int kRateStatisticsWindowSizeMs = 1000; |
| |
| 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); |
| 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); |
| LOG(LS_INFO) << "WebRTC.Video.ReceivedPacketsLostInPercent " |
| << fraction_lost; |
| } |
| } |
| |
| const int kMinRequiredSamples = 200; |
| int samples = static_cast<int>(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); |
| 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); |
| 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); |
| 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); |
| 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); |
| 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); |
| 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); |
| 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); |
| 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<VideoContentType, ContentSpecificStats> 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); |
| 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); |
| 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); |
| 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); |
| LOG(LS_INFO) << uma_prefix << ".InterframeDelayMaxInMs" << uma_suffix |
| << " " << interframe_delay_max_ms; |
| } |
| |
| int width = stats.received_width.Avg(kMinRequiredSamples); |
| if (width != -1) { |
| RTC_HISTOGRAM_COUNTS_SPARSE_10000( |
| uma_prefix + ".ReceivedWidthInPixels" + uma_suffix, width); |
| 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); |
| 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<int>(stats.total_media_bytes * 8 / |
| flow_duration_sec / 1000); |
| RTC_HISTOGRAM_COUNTS_SPARSE_10000( |
| uma_prefix + ".MediaBitrateReceivedInKbps" + uma_suffix, |
| media_bitrate_kbps); |
| 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); |
| 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); |
| 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<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / |
| 1000)); |
| int media_bitrate_kbs = |
| static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000); |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps", |
| media_bitrate_kbs); |
| LOG(LS_INFO) << "WebRTC.Video.MediaBitrateReceivedInKbps " |
| << media_bitrate_kbs; |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.PaddingBitrateReceivedInKbps", |
| static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / |
| 1000)); |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.RetransmittedBitrateReceivedInKbps", |
| static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec / |
| 1000)); |
| if (!rtx_stats_.empty()) { |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps", |
| static_cast<int>(rtx.transmitted.TotalBytes() * |
| 8 / elapsed_sec / 1000)); |
| } |
| if (config_.rtp.ulpfec.ulpfec_payload_type != -1) { |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.FecBitrateReceivedInKbps", |
| static_cast<int>(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<double> fps_fraction = |
| fps_threshold_.FractionHigh(kBadCallMinRequiredSamples); |
| if (fps_fraction) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.FrameRate", |
| static_cast<int>(100 * (1 - *fps_fraction))); |
| } |
| rtc::Optional<double> variance_fraction = |
| variance_threshold_.FractionHigh(kBadCallMinRequiredSamples); |
| if (variance_fraction) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.FrameRateVariance", |
| static_cast<int>(100 * *variance_fraction)); |
| } |
| rtc::Optional<double> qp_fraction = |
| qp_threshold_.FractionHigh(kBadCallMinRequiredSamples); |
| if (qp_fraction) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.Qp", |
| static_cast<int>(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<int>(fps)); |
| if (qp != -1) |
| qp_threshold_.AddMeasurement(qp); |
| rtc::Optional<double> fps_variance_opt = fps_threshold_.CalculateVariance(); |
| double fps_variance = fps_variance_opt.value_or(0); |
| if (fps_variance_opt) { |
| variance_threshold_.AddMeasurement(static_cast<int>(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) { |
| LOG(LS_INFO) << "Bad call (any) start: " << now; |
| } else if (prev_any_bad && !any_bad) { |
| LOG(LS_INFO) << "Bad call (any) end: " << now; |
| } |
| |
| if (!prev_fps_bad && fps_bad) { |
| LOG(LS_INFO) << "Bad call (fps) start: " << now; |
| } else if (prev_fps_bad && !fps_bad) { |
| LOG(LS_INFO) << "Bad call (fps) end: " << now; |
| } |
| |
| if (!prev_qp_bad && qp_bad) { |
| LOG(LS_INFO) << "Bad call (qp) start: " << now; |
| } else if (prev_qp_bad && !qp_bad) { |
| LOG(LS_INFO) << "Bad call (qp) end: " << now; |
| } |
| |
| if (!prev_variance_bad && variance_bad) { |
| LOG(LS_INFO) << "Bad call (variance) start: " << now; |
| } else if (prev_variance_bad && !variance_bad) { |
| LOG(LS_INFO) << "Bad call (variance) end: " << now; |
| } |
| |
| 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<int>(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<int>(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) { |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| rtc::CritScope lock(&crit_); |
| 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<uint8_t> qp, |
| VideoContentType content_type) { |
| uint64_t now = clock_->TimeInMilliseconds(); |
| |
| rtc::CritScope lock(&crit_); |
| |
| ContentSpecificStats* content_specific_stats = |
| &content_specific_stats_[content_type]; |
| ++stats_.frames_decoded; |
| if (qp) { |
| if (!stats_.qp_sum) { |
| if (stats_.frames_decoded != 1) { |
| LOG(LS_WARNING) |
| << "Frames decoded was not 1 when first qp value was received."; |
| stats_.frames_decoded = 1; |
| } |
| stats_.qp_sum = rtc::Optional<uint64_t>(0); |
| } |
| *stats_.qp_sum += *qp; |
| content_specific_stats->qp_counter.Add(*qp); |
| } else if (stats_.qp_sum) { |
| LOG(LS_WARNING) |
| << "QP sum was already set and no QP was given for a frame."; |
| stats_.qp_sum = rtc::Optional<uint64_t>(); |
| } |
| 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->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<int>(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<int>(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; |
| } |
| |
| 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; |
| } |
| |
| } // namespace webrtc |
