Add histogram stats for jitter buffer delay and target delay for received video streams:

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 59 matching lines @@
   if (qp != -1)
     RTC_LOGGED_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_LOGGED_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
 
+  int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
+  if (jb_delay_ms != -1) {
+    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
+                                      jb_delay_ms);
+  }
+  int target_delay_ms = target_delay_counter_.Avg(kMinRequiredDecodeSamples);
+  if (target_delay_ms != -1) {
+    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs",
+                                      target_delay_ms);
+  }
+  int current_delay_ms = current_delay_counter_.Avg(kMinRequiredDecodeSamples);
+  if (current_delay_ms != -1) {
+    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
+                                      current_delay_ms);
+  }
+
   int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
   if (delay_ms != -1)
     RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", 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);
@@ skipping 73 matching lines @@
                                              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);
+  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 + rtt_ms / 2);
 }
 
 void ReceiveStatisticsProxy::RtcpPacketTypesCounterUpdated(
     uint32_t ssrc,
     const RtcpPacketTypeCounter& packet_counter) {
   rtc::CritScope lock(&crit_);
   if (stats_.ssrc != ssrc)
@@ skipping 97 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