Fix uploading of available send bitrate statistics.

webrtc/pc/statscollector.cc

 /*
  *  Copyright 2012 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 762 matching lines @@
   if (pc_->session()->state() == WebRtcSession::State::STATE_CLOSED)
     return;
 
   webrtc::Call::Stats call_stats = pc_->session()->GetCallStats();
   cricket::BandwidthEstimationInfo bwe_info;
   bwe_info.available_send_bandwidth = call_stats.send_bandwidth_bps;
   bwe_info.available_recv_bandwidth = call_stats.recv_bandwidth_bps;
   bwe_info.bucket_delay = call_stats.pacer_delay_ms;
   // Fill in target encoder bitrate, actual encoder bitrate, rtx bitrate, etc.
   // TODO(holmer): Also fill this in for audio.
-  if (!pc_->session()->video_channel()) {
-    return;
+  if (pc_->session()->video_channel()) {
+    pc_->session()->video_channel()->FillBitrateInfo(&bwe_info);
   }
-  pc_->session()->video_channel()->FillBitrateInfo(&bwe_info);
   StatsReport::Id report_id(StatsReport::NewBandwidthEstimationId());
   StatsReport* report = reports_.FindOrAddNew(report_id);
   ExtractStats(bwe_info, stats_gathering_started_, report);
 }
 
 void StatsCollector::ExtractVoiceInfo() {
   RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent());
 
   if (!pc_->session()->voice_channel()) {
     return;
@@ skipping 205 matching lines @@
     StatsReport* report = entry.second;
     report->set_timestamp(stats_gathering_started_);
   }
 }
 
 void StatsCollector::ClearUpdateStatsCacheForTest() {
   stats_gathering_started_ = 0;
 }
 
 }  // namespace webrtc