Add encode and decode time to histograms stats:

webrtc/video/send_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 28 matching lines @@
 void SendStatisticsProxy::UpdateHistograms() {
   int input_fps =
       static_cast<int>(input_frame_rate_tracker_total_.units_second());
   if (input_fps > 0)
     RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.InputFramesPerSecond", input_fps);
   int sent_fps =
       static_cast<int>(sent_frame_rate_tracker_total_.units_second());
   if (sent_fps > 0)
     RTC_HISTOGRAM_COUNTS_100("WebRTC.Video.SentFramesPerSecond", sent_fps);
 
-  const int kMinRequiredSamples = 100;
+  const int kMinRequiredSamples = 200;
   int in_width = input_width_counter_.Avg(kMinRequiredSamples);
   int in_height = input_height_counter_.Avg(kMinRequiredSamples);
   if (in_width != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.InputWidthInPixels", in_width);
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.InputHeightInPixels", in_height);
   }
   int sent_width = sent_width_counter_.Avg(kMinRequiredSamples);
   int sent_height = sent_height_counter_.Avg(kMinRequiredSamples);
   if (sent_width != -1) {
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SentWidthInPixels", sent_width);
     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.SentHeightInPixels", sent_height);
   }
+  int encode_ms = encode_time_counter_.Avg(kMinRequiredSamples);
+  if (encode_ms != -1)
+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.EncodeTimeInMs", encode_ms);
 }
 
 void SendStatisticsProxy::OutgoingRate(const int video_channel,
                                        const unsigned int framerate,
                                        const unsigned int bitrate) {
   rtc::CritScope lock(&crit_);
   stats_.encode_frame_rate = framerate;
   stats_.media_bitrate_bps = bitrate;
 }
 
 void SendStatisticsProxy::CpuOveruseMetricsUpdated(
     const CpuOveruseMetrics& metrics) {
   rtc::CritScope lock(&crit_);
+  // TODO(asapersson): Change to use OnEncodedFrame() for avg_encode_time_ms.
   stats_.avg_encode_time_ms = metrics.avg_encode_time_ms;
   stats_.encode_usage_percent = metrics.encode_usage_percent;
 }
 
 void SendStatisticsProxy::SuspendChange(int video_channel, bool is_suspended) {
   rtc::CritScope lock(&crit_);
   stats_.suspended = is_suspended;
 }
 
 VideoSendStream::Stats SendStatisticsProxy::GetStats() {
@@ skipping 94 matching lines @@
 }
 
 void SendStatisticsProxy::OnIncomingFrame(int width, int height) {
   rtc::CritScope lock(&crit_);
   input_frame_rate_tracker_.Update(1);
   input_frame_rate_tracker_total_.Update(1);
   input_width_counter_.Add(width);
   input_height_counter_.Add(height);
 }
 
+void SendStatisticsProxy::OnEncodedFrame(int encode_time_ms) {
+  rtc::CritScope lock(&crit_);
+  encode_time_counter_.Add(encode_time_ms);
+}
+
 void SendStatisticsProxy::RtcpPacketTypesCounterUpdated(
     uint32_t ssrc,
     const RtcpPacketTypeCounter& packet_counter) {
   rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
   if (stats == nullptr)
     return;
 
   stats->rtcp_packet_type_counts = packet_counter;
 }
@@ skipping 60 matching lines @@
   ++num_samples;
 }
 
 int SendStatisticsProxy::SampleCounter::Avg(int min_required_samples) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return sum / num_samples;
 }
 
 }  // namespace webrtc