Increase kMinRequiredSamples (5 -> 200) for updating histogram stats from OnFrameBufferTimingsUpdat…

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 128 matching lines @@
     int key_frames_permille =
         (num_key_frames * 1000.0f / num_total_frames + 0.5f);
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
                               key_frames_permille);
   }
 
   int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
   if (qp != -1)
     RTC_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);
+  int decode_ms = decode_time_counter_.Avg(kMinRequiredSamples);
   if (decode_ms != -1)
     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
 
-  int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredDecodeSamples);
+  int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredSamples);
   if (jb_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
                                jb_delay_ms);
   }
 
-  int target_delay_ms = target_delay_counter_.Avg(kMinRequiredDecodeSamples);
+  int target_delay_ms = target_delay_counter_.Avg(kMinRequiredSamples);
   if (target_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
   }
-  int current_delay_ms = current_delay_counter_.Avg(kMinRequiredDecodeSamples);
+  int current_delay_ms = current_delay_counter_.Avg(kMinRequiredSamples);
   if (current_delay_ms != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
                                current_delay_ms);
   }
-  int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples);
+  int delay_ms = delay_counter_.Avg(kMinRequiredSamples);
   if (delay_ms != -1)
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
 
   int e2e_delay_ms = e2e_delay_counter_.Avg(kMinRequiredSamples);
   if (e2e_delay_ms != -1)
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.EndToEndDelayInMs", e2e_delay_ms);
 
   StreamDataCounters rtp = stats_.rtp_stats;
   StreamDataCounters rtx;
   for (auto it : rtx_stats_)
@@ skipping 369 matching lines @@
   sum = 0;
 }
 
 void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms,
                                          int64_t max_rtt_ms) {
   rtc::CritScope lock(&crit_);
   avg_rtt_ms_ = avg_rtt_ms;
 }
 
 }  // namespace webrtc