Add periodic logging of video 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 377 matching lines @@
   }
 }
 
 VideoSendStream::StreamStats* SendStatisticsProxy::GetStatsEntry(
     uint32_t ssrc) {
   std::map<uint32_t, VideoSendStream::StreamStats>::iterator it =
       stats_.substreams.find(ssrc);
   if (it != stats_.substreams.end())
     return &it->second;
 
+  bool is_rtx = false;
   if (std::find(config_.rtp.ssrcs.begin(), config_.rtp.ssrcs.end(), ssrc) ==
-          config_.rtp.ssrcs.end() &&
-      std::find(config_.rtp.rtx.ssrcs.begin(),
-                config_.rtp.rtx.ssrcs.end(),
-                ssrc) == config_.rtp.rtx.ssrcs.end()) {
-    return nullptr;
+      config_.rtp.ssrcs.end()) {
+    if (std::find(config_.rtp.rtx.ssrcs.begin(), config_.rtp.rtx.ssrcs.end(),
+                  ssrc) == config_.rtp.rtx.ssrcs.end()) {
+      return nullptr;
+    }
+    is_rtx = true;
   }
 
-  return &stats_.substreams[ssrc];  // Insert new entry and return ptr.
+  // Insert new entry and return ptr.
+  VideoSendStream::StreamStats* entry = &stats_.substreams[ssrc];
+  entry->is_rtx = is_rtx;
+
+  return entry;
 }
 
 void SendStatisticsProxy::OnInactiveSsrc(uint32_t ssrc) {
   rtc::CritScope lock(&crit_);
   VideoSendStream::StreamStats* stats = GetStatsEntry(ssrc);
   if (!stats)
     return;
 
   stats->total_bitrate_bps = 0;
   stats->retransmit_bitrate_bps = 0;
@@ skipping 210 matching lines @@
   return Fraction(min_required_samples, 1000.0f);
 }
 
 int SendStatisticsProxy::BoolSampleCounter::Fraction(
     int min_required_samples, float multiplier) const {
   if (num_samples < min_required_samples || num_samples == 0)
     return -1;
   return static_cast<int>((sum * multiplier / num_samples) + 0.5f);
 }
 }  // namespace webrtc