| /* |
| * Copyright (c) 2020 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_proxy2.h" |
| |
| #include <algorithm> |
| #include <cmath> |
| #include <utility> |
| |
| #include "modules/video_coding/include/video_codec_interface.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/task_utils/to_queued_task.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/time_utils.h" |
| #include "system_wrappers/include/clock.h" |
| #include "system_wrappers/include/metrics.h" |
| #include "video/video_receive_stream2.h" |
| |
| namespace webrtc { |
| namespace internal { |
| 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; |
| |
| const char* UmaPrefixForContentType(VideoContentType content_type) { |
| if (videocontenttypehelpers::IsScreenshare(content_type)) |
| return "WebRTC.Video.Screenshare"; |
| return "WebRTC.Video"; |
| } |
| |
| std::string UmaSuffixForContentType(VideoContentType content_type) { |
| char ss_buf[1024]; |
| rtc::SimpleStringBuilder ss(ss_buf); |
| 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(); |
| } |
| |
| // TODO(https://bugs.webrtc.org/11572): Workaround for an issue with some |
| // rtc::Thread instances and/or implementations that don't register as the |
| // current task queue. |
| bool IsCurrentTaskQueueOrThread(TaskQueueBase* task_queue) { |
| if (task_queue->IsCurrent()) |
| return true; |
| |
| rtc::Thread* current_thread = rtc::ThreadManager::Instance()->CurrentThread(); |
| if (!current_thread) |
| return false; |
| |
| return static_cast<TaskQueueBase*>(current_thread) == task_queue; |
| } |
| |
| } // namespace |
| |
| ReceiveStatisticsProxy::ReceiveStatisticsProxy( |
| uint32_t remote_ssrc, |
| Clock* clock, |
| TaskQueueBase* worker_thread, |
| const FieldTrialsView& field_trials) |
| : clock_(clock), |
| start_ms_(clock->TimeInMilliseconds()), |
| enable_decode_time_histograms_( |
| !field_trials.IsEnabled("WebRTC-DecodeTimeHistogramsKillSwitch")), |
| 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), |
| remote_ssrc_(remote_ssrc), |
| // 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), |
| video_quality_observer_(new VideoQualityObserver()), |
| interframe_delay_max_moving_(kMovingMaxWindowMs), |
| freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs), |
| last_content_type_(VideoContentType::UNSPECIFIED), |
| last_codec_type_(kVideoCodecVP8), |
| num_delayed_frames_rendered_(0), |
| sum_missed_render_deadline_ms_(0), |
| timing_frame_info_counter_(kMovingMaxWindowMs), |
| worker_thread_(worker_thread) { |
| RTC_DCHECK(worker_thread); |
| decode_queue_.Detach(); |
| incoming_render_queue_.Detach(); |
| stats_.ssrc = remote_ssrc_; |
| } |
| |
| ReceiveStatisticsProxy::~ReceiveStatisticsProxy() { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| } |
| |
| void ReceiveStatisticsProxy::UpdateHistograms( |
| absl::optional<int> fraction_lost, |
| const StreamDataCounters& rtp_stats, |
| const StreamDataCounters* rtx_stats) { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| |
| char log_stream_buf[8 * 1024]; |
| rtc::SimpleStringBuilder log_stream(log_stream_buf); |
| |
| 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_stream << "WebRTC.Video.ReceiveStreamLifetimeInSeconds " |
| << stream_duration_sec << '\n'; |
| } |
| |
| log_stream << "Frames decoded " << stats_.frames_decoded << '\n'; |
| |
| if (num_unique_frames_) { |
| int num_dropped_frames = *num_unique_frames_ - stats_.frames_decoded; |
| RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DroppedFrames.Receiver", |
| num_dropped_frames); |
| log_stream << "WebRTC.Video.DroppedFrames.Receiver " << num_dropped_frames |
| << '\n'; |
| } |
| |
| if (fraction_lost && stream_duration_sec >= metrics::kMinRunTimeInSeconds) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent", |
| *fraction_lost); |
| log_stream << "WebRTC.Video.ReceivedPacketsLostInPercent " << *fraction_lost |
| << '\n'; |
| } |
| |
| if (first_decoded_frame_time_ms_) { |
| const int64_t elapsed_ms = |
| (clock_->TimeInMilliseconds() - *first_decoded_frame_time_ms_); |
| if (elapsed_ms >= |
| metrics::kMinRunTimeInSeconds * rtc::kNumMillisecsPerSec) { |
| int decoded_fps = static_cast<int>( |
| (stats_.frames_decoded * 1000.0f / elapsed_ms) + 0.5f); |
| RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.DecodedFramesPerSecond", |
| decoded_fps); |
| log_stream << "WebRTC.Video.DecodedFramesPerSecond " << decoded_fps |
| << '\n'; |
| |
| const uint32_t frames_rendered = stats_.frames_rendered; |
| if (frames_rendered > 0) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DelayedFramesToRenderer", |
| static_cast<int>(num_delayed_frames_rendered_ * |
| 100 / frames_rendered)); |
| if (num_delayed_frames_rendered_ > 0) { |
| RTC_HISTOGRAM_COUNTS_1000( |
| "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs", |
| static_cast<int>(sum_missed_render_deadline_ms_ / |
| num_delayed_frames_rendered_)); |
| } |
| } |
| } |
| } |
| |
| const int kMinRequiredSamples = 200; |
| int samples = static_cast<int>(render_fps_tracker_.TotalSampleCount()); |
| if (samples >= kMinRequiredSamples) { |
| int rendered_fps = round(render_fps_tracker_.ComputeTotalRate()); |
| RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond", |
| rendered_fps); |
| log_stream << "WebRTC.Video.RenderFramesPerSecond " << rendered_fps << '\n'; |
| RTC_HISTOGRAM_COUNTS_100000( |
| "WebRTC.Video.RenderSqrtPixelsPerSecond", |
| round(render_pixel_tracker_.ComputeTotalRate())); |
| } |
| |
| absl::optional<int> sync_offset_ms = |
| sync_offset_counter_.Avg(kMinRequiredSamples); |
| if (sync_offset_ms) { |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs", |
| *sync_offset_ms); |
| log_stream << "WebRTC.Video.AVSyncOffsetInMs " << *sync_offset_ms << '\n'; |
| } |
| 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_stream << "WebRTC.Video.RtpToNtpFreqOffsetInKhz " |
| << freq_offset_stats.ToString() << '\n'; |
| } |
| |
| 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_stream << "WebRTC.Video.KeyFramesReceivedInPermille " |
| << key_frames_permille << '\n'; |
| } |
| |
| absl::optional<int> qp = qp_counters_.vp8.Avg(kMinRequiredSamples); |
| if (qp) { |
| RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", *qp); |
| log_stream << "WebRTC.Video.Decoded.Vp8.Qp " << *qp << '\n'; |
| } |
| |
| absl::optional<int> decode_ms = decode_time_counter_.Avg(kMinRequiredSamples); |
| if (decode_ms) { |
| RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", *decode_ms); |
| log_stream << "WebRTC.Video.DecodeTimeInMs " << *decode_ms << '\n'; |
| } |
| absl::optional<int> jb_delay_ms = |
| jitter_buffer_delay_counter_.Avg(kMinRequiredSamples); |
| if (jb_delay_ms) { |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs", |
| *jb_delay_ms); |
| log_stream << "WebRTC.Video.JitterBufferDelayInMs " << *jb_delay_ms << '\n'; |
| } |
| |
| absl::optional<int> target_delay_ms = |
| target_delay_counter_.Avg(kMinRequiredSamples); |
| if (target_delay_ms) { |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", |
| *target_delay_ms); |
| log_stream << "WebRTC.Video.TargetDelayInMs " << *target_delay_ms << '\n'; |
| } |
| absl::optional<int> current_delay_ms = |
| current_delay_counter_.Avg(kMinRequiredSamples); |
| if (current_delay_ms) { |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs", |
| *current_delay_ms); |
| log_stream << "WebRTC.Video.CurrentDelayInMs " << *current_delay_ms << '\n'; |
| } |
| absl::optional<int> delay_ms = delay_counter_.Avg(kMinRequiredSamples); |
| if (delay_ms) |
| 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 (const 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 (const 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); |
| |
| absl::optional<int> e2e_delay_ms = |
| stats.e2e_delay_counter.Avg(kMinRequiredSamples); |
| if (e2e_delay_ms) { |
| RTC_HISTOGRAM_COUNTS_SPARSE_10000( |
| uma_prefix + ".EndToEndDelayInMs" + uma_suffix, *e2e_delay_ms); |
| log_stream << uma_prefix << ".EndToEndDelayInMs" << uma_suffix << " " |
| << *e2e_delay_ms << '\n'; |
| } |
| absl::optional<int> e2e_delay_max_ms = stats.e2e_delay_counter.Max(); |
| if (e2e_delay_max_ms && e2e_delay_ms) { |
| RTC_HISTOGRAM_COUNTS_SPARSE_100000( |
| uma_prefix + ".EndToEndDelayMaxInMs" + uma_suffix, *e2e_delay_max_ms); |
| log_stream << uma_prefix << ".EndToEndDelayMaxInMs" << uma_suffix << " " |
| << *e2e_delay_max_ms << '\n'; |
| } |
| absl::optional<int> interframe_delay_ms = |
| stats.interframe_delay_counter.Avg(kMinRequiredSamples); |
| if (interframe_delay_ms) { |
| RTC_HISTOGRAM_COUNTS_SPARSE_10000( |
| uma_prefix + ".InterframeDelayInMs" + uma_suffix, |
| *interframe_delay_ms); |
| log_stream << uma_prefix << ".InterframeDelayInMs" << uma_suffix << " " |
| << *interframe_delay_ms << '\n'; |
| } |
| absl::optional<int> interframe_delay_max_ms = |
| stats.interframe_delay_counter.Max(); |
| if (interframe_delay_max_ms && interframe_delay_ms) { |
| RTC_HISTOGRAM_COUNTS_SPARSE_10000( |
| uma_prefix + ".InterframeDelayMaxInMs" + uma_suffix, |
| *interframe_delay_max_ms); |
| log_stream << uma_prefix << ".InterframeDelayMaxInMs" << uma_suffix << " " |
| << *interframe_delay_max_ms << '\n'; |
| } |
| |
| absl::optional<uint32_t> 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); |
| log_stream << uma_prefix << ".InterframeDelay95PercentileInMs" |
| << uma_suffix << " " << *interframe_delay_95p_ms << '\n'; |
| } |
| |
| absl::optional<int> width = stats.received_width.Avg(kMinRequiredSamples); |
| if (width) { |
| RTC_HISTOGRAM_COUNTS_SPARSE_10000( |
| uma_prefix + ".ReceivedWidthInPixels" + uma_suffix, *width); |
| log_stream << uma_prefix << ".ReceivedWidthInPixels" << uma_suffix << " " |
| << *width << '\n'; |
| } |
| |
| absl::optional<int> height = stats.received_height.Avg(kMinRequiredSamples); |
| if (height) { |
| RTC_HISTOGRAM_COUNTS_SPARSE_10000( |
| uma_prefix + ".ReceivedHeightInPixels" + uma_suffix, *height); |
| log_stream << uma_prefix << ".ReceivedHeightInPixels" << uma_suffix << " " |
| << *height << '\n'; |
| } |
| |
| 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_stream << uma_prefix << ".MediaBitrateReceivedInKbps" << uma_suffix |
| << " " << media_bitrate_kbps << '\n'; |
| } |
| |
| 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_stream << uma_prefix << ".KeyFramesReceivedInPermille" << uma_suffix |
| << " " << key_frames_permille << '\n'; |
| } |
| |
| absl::optional<int> qp = stats.qp_counter.Avg(kMinRequiredSamples); |
| if (qp) { |
| RTC_HISTOGRAM_COUNTS_SPARSE_200( |
| uma_prefix + ".Decoded.Vp8.Qp" + uma_suffix, *qp); |
| log_stream << uma_prefix << ".Decoded.Vp8.Qp" << uma_suffix << " " |
| << *qp << '\n'; |
| } |
| } |
| } |
| |
| StreamDataCounters rtp_rtx_stats = rtp_stats; |
| if (rtx_stats) |
| rtp_rtx_stats.Add(*rtx_stats); |
| |
| int64_t elapsed_sec = |
| rtp_rtx_stats.TimeSinceFirstPacketInMs(clock_->TimeInMilliseconds()) / |
| 1000; |
| if (elapsed_sec >= metrics::kMinRunTimeInSeconds) { |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.BitrateReceivedInKbps", |
| static_cast<int>(rtp_rtx_stats.transmitted.TotalBytes() * 8 / |
| elapsed_sec / 1000)); |
| int media_bitrate_kbs = static_cast<int>(rtp_stats.MediaPayloadBytes() * 8 / |
| elapsed_sec / 1000); |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps", |
| media_bitrate_kbs); |
| log_stream << "WebRTC.Video.MediaBitrateReceivedInKbps " |
| << media_bitrate_kbs << '\n'; |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.PaddingBitrateReceivedInKbps", |
| static_cast<int>(rtp_rtx_stats.transmitted.padding_bytes * 8 / |
| elapsed_sec / 1000)); |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.RetransmittedBitrateReceivedInKbps", |
| static_cast<int>(rtp_rtx_stats.retransmitted.TotalBytes() * 8 / |
| elapsed_sec / 1000)); |
| if (rtx_stats) { |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.RtxBitrateReceivedInKbps", |
| static_cast<int>(rtx_stats->transmitted.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_); |
| } |
| absl::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))); |
| } |
| absl::optional<double> variance_fraction = |
| variance_threshold_.FractionHigh(kBadCallMinRequiredSamples); |
| if (variance_fraction) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.FrameRateVariance", |
| static_cast<int>(100 * *variance_fraction)); |
| } |
| absl::optional<double> qp_fraction = |
| qp_threshold_.FractionHigh(kBadCallMinRequiredSamples); |
| if (qp_fraction) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.Qp", |
| static_cast<int>(100 * *qp_fraction)); |
| } |
| |
| RTC_LOG(LS_INFO) << log_stream.str(); |
| video_quality_observer_->UpdateHistograms( |
| videocontenttypehelpers::IsScreenshare(last_content_type_)); |
| } |
| |
| void ReceiveStatisticsProxy::QualitySample(Timestamp now) { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| |
| if (last_sample_time_ + kMinSampleLengthMs > now.ms()) |
| return; |
| |
| double fps = |
| render_fps_tracker_.ComputeRateForInterval(now.ms() - last_sample_time_); |
| absl::optional<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) |
| qp_threshold_.AddMeasurement(*qp); |
| absl::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) { |
| RTC_LOG(LS_INFO) << "Bad call (any) start: " << now.ms(); |
| } else if (prev_any_bad && !any_bad) { |
| RTC_LOG(LS_INFO) << "Bad call (any) end: " << now.ms(); |
| } |
| |
| if (!prev_fps_bad && fps_bad) { |
| RTC_LOG(LS_INFO) << "Bad call (fps) start: " << now.ms(); |
| } else if (prev_fps_bad && !fps_bad) { |
| RTC_LOG(LS_INFO) << "Bad call (fps) end: " << now.ms(); |
| } |
| |
| if (!prev_qp_bad && qp_bad) { |
| RTC_LOG(LS_INFO) << "Bad call (qp) start: " << now.ms(); |
| } else if (prev_qp_bad && !qp_bad) { |
| RTC_LOG(LS_INFO) << "Bad call (qp) end: " << now.ms(); |
| } |
| |
| if (!prev_variance_bad && variance_bad) { |
| RTC_LOG(LS_INFO) << "Bad call (variance) start: " << now.ms(); |
| } else if (prev_variance_bad && !variance_bad) { |
| RTC_LOG(LS_INFO) << "Bad call (variance) end: " << now.ms(); |
| } |
| |
| RTC_LOG(LS_VERBOSE) << "SAMPLE: sample_length: " |
| << (now.ms() - last_sample_time_) << " fps: " << fps |
| << " fps_bad: " << fps_bad << " qp: " << qp.value_or(-1) |
| << " qp_bad: " << qp_bad |
| << " variance_bad: " << variance_bad |
| << " fps_variance: " << fps_variance; |
| |
| last_sample_time_ = now.ms(); |
| 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 { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| |
| 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); |
| } |
| |
| void ReceiveStatisticsProxy::UpdateDecodeTimeHistograms( |
| int width, |
| int height, |
| int decode_time_ms) const { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| |
| bool is_4k = (width == 3840 || width == 4096) && height == 2160; |
| bool is_hd = width == 1920 && height == 1080; |
| // Only update histograms for 4k/HD and VP9/H264. |
| if ((is_4k || is_hd) && (last_codec_type_ == kVideoCodecVP9 || |
| last_codec_type_ == kVideoCodecH264)) { |
| const std::string kDecodeTimeUmaPrefix = |
| "WebRTC.Video.DecodeTimePerFrameInMs."; |
| |
| // Each histogram needs its own line for it to not be reused in the wrong |
| // way when the format changes. |
| if (last_codec_type_ == kVideoCodecVP9) { |
| bool is_sw_decoder = |
| stats_.decoder_implementation_name.compare(0, 6, "libvpx") == 0; |
| if (is_4k) { |
| if (is_sw_decoder) |
| RTC_HISTOGRAM_COUNTS_1000(kDecodeTimeUmaPrefix + "Vp9.4k.Sw", |
| decode_time_ms); |
| else |
| RTC_HISTOGRAM_COUNTS_1000(kDecodeTimeUmaPrefix + "Vp9.4k.Hw", |
| decode_time_ms); |
| } else { |
| if (is_sw_decoder) |
| RTC_HISTOGRAM_COUNTS_1000(kDecodeTimeUmaPrefix + "Vp9.Hd.Sw", |
| decode_time_ms); |
| else |
| RTC_HISTOGRAM_COUNTS_1000(kDecodeTimeUmaPrefix + "Vp9.Hd.Hw", |
| decode_time_ms); |
| } |
| } else { |
| bool is_sw_decoder = |
| stats_.decoder_implementation_name.compare(0, 6, "FFmpeg") == 0; |
| if (is_4k) { |
| if (is_sw_decoder) |
| RTC_HISTOGRAM_COUNTS_1000(kDecodeTimeUmaPrefix + "H264.4k.Sw", |
| decode_time_ms); |
| else |
| RTC_HISTOGRAM_COUNTS_1000(kDecodeTimeUmaPrefix + "H264.4k.Hw", |
| decode_time_ms); |
| |
| } else { |
| if (is_sw_decoder) |
| RTC_HISTOGRAM_COUNTS_1000(kDecodeTimeUmaPrefix + "H264.Hd.Sw", |
| decode_time_ms); |
| else |
| RTC_HISTOGRAM_COUNTS_1000(kDecodeTimeUmaPrefix + "H264.Hd.Hw", |
| decode_time_ms); |
| } |
| } |
| } |
| } |
| |
| absl::optional<int64_t> |
| ReceiveStatisticsProxy::GetCurrentEstimatedPlayoutNtpTimestampMs( |
| int64_t now_ms) const { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| if (!last_estimated_playout_ntp_timestamp_ms_ || |
| !last_estimated_playout_time_ms_) { |
| return absl::nullopt; |
| } |
| int64_t elapsed_ms = now_ms - *last_estimated_playout_time_ms_; |
| return *last_estimated_playout_ntp_timestamp_ms_ + elapsed_ms; |
| } |
| |
| VideoReceiveStreamInterface::Stats ReceiveStatisticsProxy::GetStats() const { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| |
| // Like VideoReceiveStreamInterface::GetStats, called on the worker thread |
| // from StatsCollector::ExtractMediaInfo via worker_thread()->Invoke(). |
| // WebRtcVideoChannel::GetStats(), GetVideoReceiverInfo. |
| |
| // 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); |
| |
| if (last_decoded_frame_time_ms_) { |
| // Avoid using a newer timestamp than might be pending for decoded frames. |
| // If we do use now_ms, we might roll the max window to a value that is |
| // higher than that of a decoded frame timestamp that we haven't yet |
| // captured the data for (i.e. pending call to OnDecodedFrame). |
| stats_.interframe_delay_max_ms = |
| interframe_delay_max_moving_.Max(*last_decoded_frame_time_ms_) |
| .value_or(-1); |
| } else { |
| // We're paused. Avoid changing the state of `interframe_delay_max_moving_`. |
| stats_.interframe_delay_max_ms = -1; |
| } |
| |
| stats_.freeze_count = video_quality_observer_->NumFreezes(); |
| stats_.pause_count = video_quality_observer_->NumPauses(); |
| stats_.total_freezes_duration_ms = |
| video_quality_observer_->TotalFreezesDurationMs(); |
| stats_.total_pauses_duration_ms = |
| video_quality_observer_->TotalPausesDurationMs(); |
| stats_.total_frames_duration_ms = |
| video_quality_observer_->TotalFramesDurationMs(); |
| stats_.sum_squared_frame_durations = |
| video_quality_observer_->SumSquaredFrameDurationsSec(); |
| stats_.content_type = last_content_type_; |
| stats_.timing_frame_info = timing_frame_info_counter_.Max(now_ms); |
| stats_.jitter_buffer_delay_seconds = |
| static_cast<double>(current_delay_counter_.Sum(1).value_or(0)) / |
| rtc::kNumMillisecsPerSec; |
| stats_.jitter_buffer_emitted_count = current_delay_counter_.NumSamples(); |
| stats_.estimated_playout_ntp_timestamp_ms = |
| GetCurrentEstimatedPlayoutNtpTimestampMs(now_ms); |
| return stats_; |
| } |
| |
| void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) { |
| RTC_DCHECK_RUN_ON(&decode_queue_); |
| worker_thread_->PostTask(ToQueuedTask(task_safety_, [payload_type, this]() { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| stats_.current_payload_type = payload_type; |
| })); |
| } |
| |
| void ReceiveStatisticsProxy::OnDecoderImplementationName( |
| const char* implementation_name) { |
| RTC_DCHECK_RUN_ON(&decode_queue_); |
| worker_thread_->PostTask(ToQueuedTask( |
| task_safety_, [name = std::string(implementation_name), this]() { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| stats_.decoder_implementation_name = name; |
| })); |
| } |
| |
| void ReceiveStatisticsProxy::OnFrameBufferTimingsUpdated( |
| 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) { |
| // Only called on main_thread_ with FrameBuffer3 |
| if (!worker_thread_->IsCurrent()) { |
| RTC_DCHECK_RUN_ON(&decode_queue_); |
| worker_thread_->PostTask(ToQueuedTask( |
| task_safety_, |
| [max_decode_ms, current_delay_ms, target_delay_ms, jitter_buffer_ms, |
| min_playout_delay_ms, render_delay_ms, this]() { |
| OnFrameBufferTimingsUpdated(max_decode_ms, current_delay_ms, |
| target_delay_ms, jitter_buffer_ms, |
| min_playout_delay_ms, render_delay_ms); |
| })); |
| return; |
| } |
| |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| 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; |
| 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::OnUniqueFramesCounted(int num_unique_frames) { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| num_unique_frames_.emplace(num_unique_frames); |
| } |
| |
| void ReceiveStatisticsProxy::OnTimingFrameInfoUpdated( |
| const TimingFrameInfo& info) { |
| // Only called on main_thread_ with FrameBuffer3 |
| if (!worker_thread_->IsCurrent()) { |
| RTC_DCHECK_RUN_ON(&decode_queue_); |
| worker_thread_->PostTask(ToQueuedTask( |
| task_safety_, [info, this]() { OnTimingFrameInfoUpdated(info); })); |
| return; |
| } |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| if (info.flags != VideoSendTiming::kInvalid) { |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| timing_frame_info_counter_.Add(info, now_ms); |
| } |
| |
| // Measure initial decoding latency between the first frame arriving and |
| // the first frame being decoded. |
| if (!first_frame_received_time_ms_.has_value()) { |
| first_frame_received_time_ms_ = info.receive_finish_ms; |
| } |
| if (stats_.first_frame_received_to_decoded_ms == -1 && |
| first_decoded_frame_time_ms_) { |
| stats_.first_frame_received_to_decoded_ms = |
| *first_decoded_frame_time_ms_ - *first_frame_received_time_ms_; |
| } |
| } |
| |
| void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated( |
| uint32_t ssrc, |
| const RtcpPacketTypeCounter& packet_counter) { |
| if (ssrc != remote_ssrc_) |
| return; |
| |
| if (!IsCurrentTaskQueueOrThread(worker_thread_)) { |
| // RtpRtcpInterface::Configuration has a single |
| // RtcpPacketTypeCounterObserver and that same configuration may be used for |
| // both receiver and sender (see ModuleRtpRtcpImpl::ModuleRtpRtcpImpl). The |
| // RTCPSender implementation currently makes calls to this function on a |
| // process thread whereas the RTCPReceiver implementation calls back on the |
| // [main] worker thread. |
| // So until the sender implementation has been updated, we work around this |
| // here by posting the update to the expected thread. We make a by value |
| // copy of the `task_safety_` to handle the case if the queued task |
| // runs after the `ReceiveStatisticsProxy` has been deleted. In such a |
| // case the packet_counter update won't be recorded. |
| worker_thread_->PostTask( |
| ToQueuedTask(task_safety_, [ssrc, packet_counter, this]() { |
| RtcpPacketTypesCounterUpdated(ssrc, packet_counter); |
| })); |
| return; |
| } |
| |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| stats_.rtcp_packet_type_counts = packet_counter; |
| } |
| |
| void ReceiveStatisticsProxy::OnCname(uint32_t ssrc, absl::string_view cname) { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| // TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we |
| // receive stats from one of them. |
| if (remote_ssrc_ != ssrc) |
| return; |
| |
| stats_.c_name = std::string(cname); |
| } |
| |
| void ReceiveStatisticsProxy::OnDecodedFrame(const VideoFrame& frame, |
| absl::optional<uint8_t> qp, |
| int32_t decode_time_ms, |
| VideoContentType content_type) { |
| webrtc::TimeDelta processing_delay = webrtc::TimeDelta::Millis(0); |
| webrtc::Timestamp current_time = clock_->CurrentTime(); |
| // TODO(bugs.webrtc.org/13984): some tests do not fill packet_infos(). |
| webrtc::TimeDelta assembly_time = webrtc::TimeDelta::Millis(0); |
| if (frame.packet_infos().size() > 0) { |
| const auto [first_packet, last_packet] = std::minmax_element( |
| frame.packet_infos().cbegin(), frame.packet_infos().cend(), |
| [](const webrtc::RtpPacketInfo& a, const webrtc::RtpPacketInfo& b) { |
| return a.receive_time() < b.receive_time(); |
| }); |
| if (first_packet->receive_time().IsFinite()) { |
| processing_delay = current_time - first_packet->receive_time(); |
| // Extract frame assembly time (i.e. time between earliest and latest |
| // packet arrival). Note: for single-packet frames this will be 0. |
| assembly_time = |
| last_packet->receive_time() - first_packet->receive_time(); |
| } |
| } |
| // See VCMDecodedFrameCallback::Decoded for more info on what thread/queue we |
| // may be on. E.g. on iOS this gets called on |
| // "com.apple.coremedia.decompressionsession.clientcallback" |
| VideoFrameMetaData meta(frame, current_time); |
| worker_thread_->PostTask( |
| ToQueuedTask(task_safety_, [meta, qp, decode_time_ms, processing_delay, |
| assembly_time, content_type, this]() { |
| OnDecodedFrame(meta, qp, decode_time_ms, processing_delay, |
| assembly_time, content_type); |
| })); |
| } |
| |
| void ReceiveStatisticsProxy::OnDecodedFrame( |
| const VideoFrameMetaData& frame_meta, |
| absl::optional<uint8_t> qp, |
| int32_t decode_time_ms, |
| webrtc::TimeDelta processing_delay, |
| webrtc::TimeDelta assembly_time, |
| VideoContentType content_type) { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| |
| const bool is_screenshare = |
| videocontenttypehelpers::IsScreenshare(content_type); |
| const bool was_screenshare = |
| videocontenttypehelpers::IsScreenshare(last_content_type_); |
| |
| if (is_screenshare != was_screenshare) { |
| // Reset the quality observer if content type is switched. But first report |
| // stats for the previous part of the call. |
| video_quality_observer_->UpdateHistograms(was_screenshare); |
| video_quality_observer_.reset(new VideoQualityObserver()); |
| } |
| |
| video_quality_observer_->OnDecodedFrame(frame_meta.rtp_timestamp, qp, |
| last_codec_type_); |
| |
| 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_.qp_sum = 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.reset(); |
| } |
| decode_time_counter_.Add(decode_time_ms); |
| stats_.decode_ms = decode_time_ms; |
| stats_.total_decode_time_ms += decode_time_ms; |
| stats_.total_processing_delay += processing_delay; |
| stats_.total_assembly_time += assembly_time; |
| if (!assembly_time.IsZero()) { |
| ++stats_.frames_assembled_from_multiple_packets; |
| } |
| if (enable_decode_time_histograms_) { |
| UpdateDecodeTimeHistograms(frame_meta.width, frame_meta.height, |
| decode_time_ms); |
| } |
| |
| last_content_type_ = content_type; |
| decode_fps_estimator_.Update(1, frame_meta.decode_timestamp.ms()); |
| |
| if (last_decoded_frame_time_ms_) { |
| int64_t interframe_delay_ms = |
| frame_meta.decode_timestamp.ms() - *last_decoded_frame_time_ms_; |
| RTC_DCHECK_GE(interframe_delay_ms, 0); |
| double interframe_delay = interframe_delay_ms / 1000.0; |
| stats_.total_inter_frame_delay += interframe_delay; |
| stats_.total_squared_inter_frame_delay += |
| interframe_delay * interframe_delay; |
| interframe_delay_max_moving_.Add(interframe_delay_ms, |
| frame_meta.decode_timestamp.ms()); |
| 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; |
| } |
| if (stats_.frames_decoded == 1) { |
| first_decoded_frame_time_ms_.emplace(frame_meta.decode_timestamp.ms()); |
| } |
| last_decoded_frame_time_ms_.emplace(frame_meta.decode_timestamp.ms()); |
| } |
| |
| void ReceiveStatisticsProxy::OnRenderedFrame( |
| const VideoFrameMetaData& frame_meta) { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| // Called from VideoReceiveStream2::OnFrame. |
| |
| RTC_DCHECK_GT(frame_meta.width, 0); |
| RTC_DCHECK_GT(frame_meta.height, 0); |
| |
| video_quality_observer_->OnRenderedFrame(frame_meta); |
| |
| ContentSpecificStats* content_specific_stats = |
| &content_specific_stats_[last_content_type_]; |
| renders_fps_estimator_.Update(1, frame_meta.decode_timestamp.ms()); |
| |
| ++stats_.frames_rendered; |
| stats_.width = frame_meta.width; |
| stats_.height = frame_meta.height; |
| |
| render_fps_tracker_.AddSamples(1); |
| render_pixel_tracker_.AddSamples(sqrt(frame_meta.width * frame_meta.height)); |
| content_specific_stats->received_width.Add(frame_meta.width); |
| content_specific_stats->received_height.Add(frame_meta.height); |
| |
| // Consider taking stats_.render_delay_ms into account. |
| const int64_t time_until_rendering_ms = |
| frame_meta.render_time_ms() - frame_meta.decode_timestamp.ms(); |
| if (time_until_rendering_ms < 0) { |
| sum_missed_render_deadline_ms_ += -time_until_rendering_ms; |
| ++num_delayed_frames_rendered_; |
| } |
| |
| if (frame_meta.ntp_time_ms > 0) { |
| int64_t delay_ms = |
| clock_->CurrentNtpInMilliseconds() - frame_meta.ntp_time_ms; |
| if (delay_ms >= 0) { |
| content_specific_stats->e2e_delay_counter.Add(delay_ms); |
| } |
| } |
| |
| QualitySample(frame_meta.decode_timestamp); |
| } |
| |
| void ReceiveStatisticsProxy::OnSyncOffsetUpdated(int64_t video_playout_ntp_ms, |
| int64_t sync_offset_ms, |
| double estimated_freq_khz) { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| |
| const int64_t now_ms = clock_->TimeInMilliseconds(); |
| sync_offset_counter_.Add(std::abs(sync_offset_ms)); |
| stats_.sync_offset_ms = sync_offset_ms; |
| last_estimated_playout_ntp_timestamp_ms_ = video_playout_ntp_ms; |
| last_estimated_playout_time_ms_ = now_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::OnCompleteFrame(bool is_keyframe, |
| size_t size_bytes, |
| VideoContentType content_type) { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| |
| if (is_keyframe) { |
| ++stats_.frame_counts.key_frames; |
| } else { |
| ++stats_.frame_counts.delta_frames; |
| } |
| |
| // Content type extension is set only for keyframes and should be propagated |
| // for all the following delta frames. Here we may receive frames out of order |
| // and miscategorise some delta frames near the layer switch. |
| // This may slightly offset calculated bitrate and keyframes permille metrics. |
| VideoContentType propagated_content_type = |
| is_keyframe ? content_type : last_content_type_; |
| |
| ContentSpecificStats* content_specific_stats = |
| &content_specific_stats_[propagated_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::OnDroppedFrames(uint32_t frames_dropped) { |
| // Can be called on either the decode queue or the worker thread |
| // See FrameBuffer2 for more details. |
| worker_thread_->PostTask(ToQueuedTask(task_safety_, [frames_dropped, this]() { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| stats_.frames_dropped += frames_dropped; |
| })); |
| } |
| |
| void ReceiveStatisticsProxy::OnPreDecode(VideoCodecType codec_type, int qp) { |
| RTC_DCHECK_RUN_ON(&decode_queue_); |
| worker_thread_->PostTask(ToQueuedTask(task_safety_, [codec_type, qp, this]() { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| last_codec_type_ = codec_type; |
| if (last_codec_type_ == kVideoCodecVP8 && qp != -1) { |
| qp_counters_.vp8.Add(qp); |
| qp_sample_.Add(qp); |
| } |
| })); |
| } |
| |
| void ReceiveStatisticsProxy::OnStreamInactive() { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| |
| // TODO(sprang): Figure out any other state that should be reset. |
| |
| // Don't report inter-frame delay if stream was paused. |
| last_decoded_frame_time_ms_.reset(); |
| |
| video_quality_observer_->OnStreamInactive(); |
| } |
| |
| void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms) { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| avg_rtt_ms_ = avg_rtt_ms; |
| } |
| |
| void ReceiveStatisticsProxy::DecoderThreadStarting() { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| } |
| |
| void ReceiveStatisticsProxy::DecoderThreadStopped() { |
| RTC_DCHECK_RUN_ON(&main_thread_); |
| decode_queue_.Detach(); |
| } |
| |
| ReceiveStatisticsProxy::ContentSpecificStats::ContentSpecificStats() |
| : interframe_delay_percentiles(kMaxCommonInterframeDelayMs) {} |
| |
| ReceiveStatisticsProxy::ContentSpecificStats::~ContentSpecificStats() = default; |
| |
| 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 internal |
| } // namespace webrtc |
