de9e5fffa2
- Add histogram: "WebRTC.Video.RtpToNtpFreqOffsetInKhz" The absolute value of the difference between the estimated frequency during RTP timestamp to NTP time conversion and the actual value (i.e. 90 kHz) is measured per received video frame. The max offset during 40 second intervals is stored. The average of these stored offsets per received video stream is recorded when a stream is removed. Updated rtp_to_ntp.cc: - Add validation for only inserting newer RTCP sender reports to the rtcp list. - Move calculation of frequency/offset (from RTP/NTP timestamp pairs) to UpdateRtcpList. Calculated when a new RTCP SR in inserted (and not in RtpToNtpMs per packet). BUG=webrtc:6579 Review-Url: https://codereview.webrtc.org/2385763002 Cr-Commit-Position: refs/heads/master@{#14891}
342 lines
13 KiB
C++
342 lines
13 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video/receive_statistics_proxy.h"
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#include <cmath>
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#include "webrtc/base/checks.h"
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#include "webrtc/modules/video_coding/include/video_codec_interface.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/system_wrappers/include/metrics.h"
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namespace webrtc {
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namespace {
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// Periodic time interval for processing samples for |freq_offset_counter_|.
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const int64_t kFreqOffsetProcessIntervalMs = 40000;
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} // namespace
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ReceiveStatisticsProxy::ReceiveStatisticsProxy(
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const VideoReceiveStream::Config* config,
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Clock* clock)
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: clock_(clock),
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config_(*config),
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start_ms_(clock->TimeInMilliseconds()),
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// 1000ms window, scale 1000 for ms to s.
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decode_fps_estimator_(1000, 1000),
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renders_fps_estimator_(1000, 1000),
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render_fps_tracker_(100, 10u),
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render_pixel_tracker_(100, 10u),
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freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs) {
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stats_.ssrc = config_.rtp.remote_ssrc;
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for (auto it : config_.rtp.rtx)
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rtx_stats_[it.second.ssrc] = StreamDataCounters();
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}
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ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
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UpdateHistograms();
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}
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void ReceiveStatisticsProxy::UpdateHistograms() {
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RTC_HISTOGRAM_COUNTS_100000(
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"WebRTC.Video.ReceiveStreamLifetimeInSeconds",
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(clock_->TimeInMilliseconds() - start_ms_) / 1000);
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int fraction_lost = report_block_stats_.FractionLostInPercent();
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if (fraction_lost != -1) {
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RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
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fraction_lost);
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}
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const int kMinRequiredSamples = 200;
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int samples = static_cast<int>(render_fps_tracker_.TotalSampleCount());
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if (samples > kMinRequiredSamples) {
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RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond",
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round(render_fps_tracker_.ComputeTotalRate()));
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RTC_HISTOGRAM_COUNTS_100000(
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"WebRTC.Video.RenderSqrtPixelsPerSecond",
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round(render_pixel_tracker_.ComputeTotalRate()));
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}
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int width = render_width_counter_.Avg(kMinRequiredSamples);
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int height = render_height_counter_.Avg(kMinRequiredSamples);
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if (width != -1) {
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels", width);
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels", height);
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}
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int sync_offset_ms = sync_offset_counter_.Avg(kMinRequiredSamples);
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if (sync_offset_ms != -1) {
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs", sync_offset_ms);
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}
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AggregatedStats freq_offset_stats = freq_offset_counter_.GetStats();
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if (freq_offset_stats.num_samples > 0) {
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtpToNtpFreqOffsetInKhz",
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freq_offset_stats.average);
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}
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int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
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if (qp != -1)
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RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
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// TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and
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// not per frame. Change decode time to include every frame.
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const int kMinRequiredDecodeSamples = 5;
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int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
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if (decode_ms != -1)
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RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
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int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
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if (jb_delay_ms != -1) {
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
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jb_delay_ms);
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}
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int target_delay_ms = target_delay_counter_.Avg(kMinRequiredDecodeSamples);
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if (target_delay_ms != -1) {
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
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}
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int current_delay_ms = current_delay_counter_.Avg(kMinRequiredDecodeSamples);
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if (current_delay_ms != -1) {
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
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current_delay_ms);
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}
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int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
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if (delay_ms != -1)
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
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int e2e_delay_ms = e2e_delay_counter_.Avg(kMinRequiredSamples);
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if (e2e_delay_ms != -1)
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.EndToEndDelayInMs", e2e_delay_ms);
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StreamDataCounters rtp = stats_.rtp_stats;
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StreamDataCounters rtx;
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for (auto it : rtx_stats_)
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rtx.Add(it.second);
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StreamDataCounters rtp_rtx = rtp;
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rtp_rtx.Add(rtx);
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int64_t elapsed_sec =
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rtp_rtx.TimeSinceFirstPacketInMs(clock_->TimeInMilliseconds()) / 1000;
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if (elapsed_sec > metrics::kMinRunTimeInSeconds) {
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RTC_HISTOGRAM_COUNTS_10000(
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"WebRTC.Video.BitrateReceivedInKbps",
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static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec /
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1000));
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RTC_HISTOGRAM_COUNTS_10000(
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"WebRTC.Video.MediaBitrateReceivedInKbps",
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static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000));
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RTC_HISTOGRAM_COUNTS_10000(
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"WebRTC.Video.PaddingBitrateReceivedInKbps",
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static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
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1000));
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RTC_HISTOGRAM_COUNTS_10000(
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"WebRTC.Video.RetransmittedBitrateReceivedInKbps",
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static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / elapsed_sec /
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1000));
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if (!rtx_stats_.empty()) {
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtxBitrateReceivedInKbps",
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static_cast<int>(rtx.transmitted.TotalBytes() *
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8 / elapsed_sec / 1000));
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}
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if (config_.rtp.ulpfec.ulpfec_payload_type != -1) {
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RTC_HISTOGRAM_COUNTS_10000(
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"WebRTC.Video.FecBitrateReceivedInKbps",
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static_cast<int>(rtp_rtx.fec.TotalBytes() * 8 / elapsed_sec / 1000));
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}
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const RtcpPacketTypeCounter& counters = stats_.rtcp_packet_type_counts;
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
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counters.nack_packets * 60 / elapsed_sec);
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
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counters.fir_packets * 60 / elapsed_sec);
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RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
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counters.pli_packets * 60 / elapsed_sec);
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if (counters.nack_requests > 0) {
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RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
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counters.UniqueNackRequestsInPercent());
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}
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}
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}
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VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
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rtc::CritScope lock(&crit_);
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return stats_;
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}
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void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) {
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rtc::CritScope lock(&crit_);
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stats_.current_payload_type = payload_type;
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}
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void ReceiveStatisticsProxy::OnDecoderImplementationName(
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const char* implementation_name) {
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rtc::CritScope lock(&crit_);
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stats_.decoder_implementation_name = implementation_name;
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}
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void ReceiveStatisticsProxy::OnIncomingRate(unsigned int framerate,
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unsigned int bitrate_bps) {
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rtc::CritScope lock(&crit_);
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stats_.network_frame_rate = framerate;
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stats_.total_bitrate_bps = bitrate_bps;
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}
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void ReceiveStatisticsProxy::OnDecoderTiming(int decode_ms,
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int max_decode_ms,
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int current_delay_ms,
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int target_delay_ms,
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int jitter_buffer_ms,
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int min_playout_delay_ms,
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int render_delay_ms,
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int64_t rtt_ms) {
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rtc::CritScope lock(&crit_);
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stats_.decode_ms = decode_ms;
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stats_.max_decode_ms = max_decode_ms;
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stats_.current_delay_ms = current_delay_ms;
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stats_.target_delay_ms = target_delay_ms;
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stats_.jitter_buffer_ms = jitter_buffer_ms;
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stats_.min_playout_delay_ms = min_playout_delay_ms;
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stats_.render_delay_ms = render_delay_ms;
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decode_time_counter_.Add(decode_ms);
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jitter_buffer_delay_counter_.Add(jitter_buffer_ms);
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target_delay_counter_.Add(target_delay_ms);
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current_delay_counter_.Add(current_delay_ms);
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// Network delay (rtt/2) + target_delay_ms (jitter delay + decode time +
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// render delay).
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delay_counter_.Add(target_delay_ms + rtt_ms / 2);
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}
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void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
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uint32_t ssrc,
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const RtcpPacketTypeCounter& packet_counter) {
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rtc::CritScope lock(&crit_);
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if (stats_.ssrc != ssrc)
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return;
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stats_.rtcp_packet_type_counts = packet_counter;
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}
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void ReceiveStatisticsProxy::StatisticsUpdated(
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const webrtc::RtcpStatistics& statistics,
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uint32_t ssrc) {
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rtc::CritScope lock(&crit_);
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// TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we
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// receive stats from one of them.
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if (stats_.ssrc != ssrc)
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return;
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stats_.rtcp_stats = statistics;
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report_block_stats_.Store(statistics, ssrc, 0);
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}
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void ReceiveStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {
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rtc::CritScope lock(&crit_);
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// TODO(pbos): Handle both local and remote ssrcs here and RTC_DCHECK that we
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// receive stats from one of them.
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if (stats_.ssrc != ssrc)
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return;
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stats_.c_name = cname;
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}
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void ReceiveStatisticsProxy::DataCountersUpdated(
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const webrtc::StreamDataCounters& counters,
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uint32_t ssrc) {
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rtc::CritScope lock(&crit_);
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if (ssrc == stats_.ssrc) {
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stats_.rtp_stats = counters;
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} else {
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auto it = rtx_stats_.find(ssrc);
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if (it != rtx_stats_.end()) {
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it->second = counters;
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} else {
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RTC_NOTREACHED() << "Unexpected stream ssrc: " << ssrc;
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}
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}
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}
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void ReceiveStatisticsProxy::OnDecodedFrame() {
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uint64_t now = clock_->TimeInMilliseconds();
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rtc::CritScope lock(&crit_);
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++stats_.frames_decoded;
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decode_fps_estimator_.Update(1, now);
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stats_.decode_frame_rate = decode_fps_estimator_.Rate(now).value_or(0);
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}
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void ReceiveStatisticsProxy::OnRenderedFrame(const VideoFrame& frame) {
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int width = frame.width();
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int height = frame.height();
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RTC_DCHECK_GT(width, 0);
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RTC_DCHECK_GT(height, 0);
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uint64_t now = clock_->TimeInMilliseconds();
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rtc::CritScope lock(&crit_);
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renders_fps_estimator_.Update(1, now);
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stats_.render_frame_rate = renders_fps_estimator_.Rate(now).value_or(0);
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stats_.width = width;
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stats_.height = height;
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render_width_counter_.Add(width);
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render_height_counter_.Add(height);
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render_fps_tracker_.AddSamples(1);
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render_pixel_tracker_.AddSamples(sqrt(width * height));
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if (frame.ntp_time_ms() > 0) {
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int64_t delay_ms = clock_->CurrentNtpInMilliseconds() - frame.ntp_time_ms();
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if (delay_ms >= 0)
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e2e_delay_counter_.Add(delay_ms);
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}
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}
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void ReceiveStatisticsProxy::OnSyncOffsetUpdated(int64_t sync_offset_ms,
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double estimated_freq_khz) {
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rtc::CritScope lock(&crit_);
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sync_offset_counter_.Add(std::abs(sync_offset_ms));
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stats_.sync_offset_ms = sync_offset_ms;
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const double kMaxFreqKhz = 10000.0;
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int offset_khz = kMaxFreqKhz;
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// Should not be zero or negative. If so, report max.
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if (estimated_freq_khz < kMaxFreqKhz && estimated_freq_khz > 0.0)
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offset_khz = static_cast<int>(std::fabs(estimated_freq_khz - 90.0) + 0.5);
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freq_offset_counter_.Add(offset_khz);
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}
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void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
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uint32_t frameRate) {
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}
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void ReceiveStatisticsProxy::OnFrameCountsUpdated(
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const FrameCounts& frame_counts) {
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rtc::CritScope lock(&crit_);
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stats_.frame_counts = frame_counts;
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}
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void ReceiveStatisticsProxy::OnDiscardedPacketsUpdated(int discarded_packets) {
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rtc::CritScope lock(&crit_);
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stats_.discarded_packets = discarded_packets;
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}
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void ReceiveStatisticsProxy::OnPreDecode(
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const EncodedImage& encoded_image,
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const CodecSpecificInfo* codec_specific_info) {
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if (!codec_specific_info || encoded_image.qp_ == -1) {
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return;
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}
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if (codec_specific_info->codecType == kVideoCodecVP8) {
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qp_counters_.vp8.Add(encoded_image.qp_);
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}
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}
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void ReceiveStatisticsProxy::SampleCounter::Add(int sample) {
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sum += sample;
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++num_samples;
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}
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int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
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if (num_samples < min_required_samples || num_samples == 0)
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return -1;
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return sum / num_samples;
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}
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} // namespace webrtc
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