Do not update stats for WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video

webrtc/call/call.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 717 matching lines @@
 }
 
 void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
                             int64_t rtt_ms) {
   bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss,
                                        rtt_ms);
 
   // Ignore updates where the bitrate is zero because the aggregate network
   // state is down.
   if (target_bitrate_bps > 0) {
+    {
+      ReadLockScoped read_lock(*send_crit_);
+      // Do not update the stats if we are not sending video.
+      if (video_send_streams_.empty())
+        return;
+    }
     rtc::CritScope lock(&bitrate_crit_);
     // We only update these stats if we have send streams, and assume that
     // OnNetworkChanged is called roughly with a fixed frequency.
     estimated_send_bitrate_sum_kbits_ += target_bitrate_bps / 1000;
     // Pacer bitrate might be higher than bitrate estimate if enforcing min
     // bitrate.
     uint32_t pacer_bitrate_bps =
         std::max(target_bitrate_bps, min_allocated_send_bitrate_bps_);
     pacer_bitrate_sum_kbits_ += pacer_bitrate_bps / 1000;
     ++num_bitrate_updates_;
@@ skipping 153 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc