Update histogram "WebRTC.Video.OnewayDelayInMs" to use the estimated one-way 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 91 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);
-  // 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;
 }
@@ skipping 29 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);
 }
 
-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);
   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) {
+    delay_counter_.Add(clock_->CurrentNtpInMilliseconds() -
+                       frame.ntp_time_ms());

[stefan-webrtc, 2016/02/22 16:18:21]

Maybe we should make sure this is positive before logging since ntp_time_ms() is
an estimate based on rtt. Or could you verify that the estimation makes sure it
doesn't happen?

[åsapersson, 2016/03/01 12:01:43]

RemoteNtpTimeEstimator::Estimate returns -1 on error (e.g. if having received
less than two RTCP SR reports).

So ntp_time_ms should be -1 if there is no estimate (or 0 if not set).

So maybe checking if frame.ntp_time_ms() > 0 should be enough?

[stefan-webrtc, 2016/03/01 12:38:20]

I'm not sure that's enough as there are no guarantees that frame.ntp_time_ms() <
clock_->CurrentNtpInMilliseconds():

https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc...

Maybe we can make sure it is in that file though? I.e., add a line:
receiver_capture_ntp_ms = std::max(clock_->CurrentNtpInMilliseconds(),
receiver_capture_ntp_ms);

[åsapersson, 2016/03/01 13:46:42]

Ok. Added a check that the delay is positive before logging.

+  }
 }
 
 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;
@@ skipping 20 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