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);

[stefan-webrtc, 2016/04/26 08:19:26]

We also have "current delay ms" which refers to the actual delay that we impose
on frames, with the goal of reaching the target delay. It can be different
because we're only allowed to change the current delay with 10% per second.
Maybe we should track that too, or instead?

[åsapersson, 2016/04/26 10:23:58]

Ok, added current delay.

+  if (target_delay_ms != -1) {
+    RTC_LOGGED_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs",
+                                      target_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);

[åsapersson, 2016/04/20 10:48:47]

Let target delay stats replace metric below and then reland
https://codereview.webrtc.org/1905563002/ ?

[åsapersson, 2016/04/26 10:23:58]

Reland CL above?

[stefan-webrtc, 2016/04/26 10:39:42]

Sgtm if Magnus agrees.

   // 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