Add encode and decode time to histograms stats:

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 21 matching lines @@
   int fraction_lost = report_block_stats_.FractionLostInPercent();
   if (fraction_lost != -1) {
     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
         fraction_lost);
   }
 
   int render_fps = static_cast<int>(render_fps_tracker_total_.units_second());
   if (render_fps > 0)
     RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.RenderFramesPerSecond", render_fps);
 
-  const int kMinRequiredSamples = 100;
+  const int kMinRequiredSamples = 200;
   int width = render_width_counter_.Avg(kMinRequiredSamples);
   int height = render_height_counter_.Avg(kMinRequiredSamples);
   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);
 }
 
 VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
   rtc::CritScope lock(&crit_);
   return stats_;
 }
 
 void ReceiveStatisticsProxy::IncomingRate(const int video_channel,
                                           const unsigned int framerate,
                                           const unsigned int bitrate_bps) {
@@ skipping 10 matching lines @@
                                            int min_playout_delay_ms,
                                            int render_delay_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);
 }
 
 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