webrtc/video/receive_statistics_proxy.cc

/*
 *  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 <cmath>

#include "webrtc/base/checks.h"
#include "webrtc/modules/video_coding/include/video_codec_interface.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/metrics.h"

namespace webrtc {

ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc, Clock* clock)
    : clock_(clock),
      // 1000ms window, scale 1000 for ms to s.
      decode_fps_estimator_(1000, 1000),
      renders_fps_estimator_(1000, 1000),
      render_fps_tracker_(100u, 10u),
      render_pixel_tracker_(100u, 10u) {
  stats_.ssrc = ssrc;
}

ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
  UpdateHistograms();
}

void ReceiveStatisticsProxy::UpdateHistograms() {
  int fraction_lost = report_block_stats_.FractionLostInPercent();
  if (fraction_lost != -1) {
    RTC_HISTOGRAM_PERCENTAGE("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 width = render_width_counter_.Avg(kMinRequiredSamples);
  int height = render_height_counter_.Avg(kMinRequiredSamples);
  if (width != -1) {
    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels", width);
    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels", height);
  }
  int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
  if (qp != -1)
    RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);

  // TODO(asapersson): DecoderTiming() is call periodically (each 1000ms) and
  // not per frame. Change decode time to include every frame.
  const int kMinRequiredDecodeSamples = 5;
  int decode_ms = decode_time_counter_.Avg(kMinRequiredDecodeSamples);
  if (decode_ms != -1)
    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);

  int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
  if (delay_ms != -1)
    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
}

VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
  rtc::CritScope lock(&crit_);
  return stats_;
}

void ReceiveStatisticsProxy::OnIncomingPayloadType(int payload_type) {
  rtc::CritScope lock(&crit_);
  stats_.current_payload_type = payload_type;
}

void ReceiveStatisticsProxy::OnIncomingRate(unsigned int framerate,
                                            unsigned int bitrate_bps) {
  rtc::CritScope lock(&crit_);
  stats_.network_frame_rate = framerate;
  stats_.total_bitrate_bps = bitrate_bps;
}

void ReceiveStatisticsProxy::OnDecoderTiming(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,
                                             int64_t rtt_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);
  // Network delay (rtt/2) + target_delay_ms (jitter delay + decode time +
  // render delay).
  delay_counter_.Add(target_delay_ms + rtt_ms / 2);
}

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);
}

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) {
  rtc::CritScope lock(&crit_);
  if (stats_.ssrc != ssrc)
    return;
  stats_.rtp_stats = counters;
}

void ReceiveStatisticsProxy::OnDecodedFrame() {
  uint64_t now = clock_->TimeInMilliseconds();

  rtc::CritScope lock(&crit_);
  decode_fps_estimator_.Update(1, now);
  stats_.decode_frame_rate = decode_fps_estimator_.Rate(now);
}

void ReceiveStatisticsProxy::OnRenderedFrame(int width, int height) {
  RTC_DCHECK_GT(width, 0);
  RTC_DCHECK_GT(height, 0);
  uint64_t now = clock_->TimeInMilliseconds();

  rtc::CritScope lock(&crit_);
  renders_fps_estimator_.Update(1, now);
  stats_.render_frame_rate = renders_fps_estimator_.Rate(now);
  render_width_counter_.Add(width);
  render_height_counter_.Add(height);
  render_fps_tracker_.AddSamples(1);
  render_pixel_tracker_.AddSamples(sqrt(width * height));
}

void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
                                                   uint32_t frameRate) {
}

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 == nullptr || encoded_image.qp_ == -1) {
    return;
  }
  if (codec_specific_info->codecType == kVideoCodecVP8) {
    qp_counters_.vp8.Add(encoded_image.qp_);
  }
}

void ReceiveStatisticsProxy::SampleCounter::Add(int sample) {
  sum += sample;
  ++num_samples;
}

int ReceiveStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
  if (num_samples < min_required_samples || num_samples == 0)
    return -1;
  return sum / num_samples;
}

}  // namespace webrtc