Add histogram for end-to-end delay.

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.
  */
 
@@ skipping 76 matching lines @@
   }
   int target_delay_ms = target_delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (target_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
   }
   int current_delay_ms = current_delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (current_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
                                current_delay_ms);
   }
-
   int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (delay_ms != -1)
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
 
+  int e2e_delay_ms = e2e_delay_counter_.Avg(kMinRequiredSamples);
+  if (e2e_delay_ms != -1)
+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.EndToEndDelayInMs", e2e_delay_ms);
+
   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(
@@ skipping 129 matching lines @@
 }
 
 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).value_or(0);
 }
 
-void ReceiveStatisticsProxy::OnRenderedFrame(int width, int height) {
+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_);
   renders_fps_estimator_.Update(1, now);
   stats_.render_frame_rate = renders_fps_estimator_.Rate(now).value_or(0);
   stats_.width = width;
   stats_.height = height;
   render_width_counter_.Add(width);
   render_height_counter_.Add(height);
   render_fps_tracker_.AddSamples(1);
   render_pixel_tracker_.AddSamples(sqrt(width * height));
+
+  if (frame.ntp_time_ms() > 0) {
+    int64_t delay_ms = clock_->CurrentNtpInMilliseconds() - frame.ntp_time_ms();
+    if (delay_ms >= 0)
+      e2e_delay_counter_.Add(delay_ms);
+  }
 }
 
 void ReceiveStatisticsProxy::OnSyncOffsetUpdated(int64_t sync_offset_ms) {
   rtc::CritScope lock(&crit_);
   sync_offset_counter_.Add(std::abs(sync_offset_ms));
   stats_.sync_offset_ms = sync_offset_ms;
 }
 
 void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
                                                    uint32_t frameRate) {
@@ skipping 26 matching lines @@
   ++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