Add delay metric (includes network delay (rtt/2) + jitter delay + decode time +

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 35 matching lines @@
   if (width != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedWidthInPixels", width);
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.ReceivedHeightInPixels", height);
   }
   // 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.ReceiveSideDelayInMs", 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) {
+                                             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);
+  delay_counter_.Add(target_delay_ms + rtt_ms / 2);

[stefan-webrtc, 2015/09/28 15:20:42]

Why are we adding rtt/2? Should at least be explained in the CL description.

[åsapersson, 2015/10/05 13:26:00]

Added comment and updated CL description.

 }
 
 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 67 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