Fix TSAN race in webrtc::voe::Channel

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 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 2760 matching lines @@
 }
 
 void Channel::SetRtcEventLog(RtcEventLog* event_log) {
   event_log_proxy_->SetEventLog(event_log);
 }
 
 void Channel::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
   rtcp_rtt_stats_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
 }
 
-void Channel::UpdateOverheadForEncoder() {
+void Channel::UpdateOverheadForEncoder(size_t overhead_per_packet) {
   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
     if (*encoder) {
-      (*encoder)->OnReceivedOverhead(transport_overhead_per_packet_ +
-                                     rtp_overhead_per_packet_);
+      (*encoder)->OnReceivedOverhead(overhead_per_packet);
     }
   });
 }
 
+// TODO(solenberg): Make AudioSendStream an OverheadObserver instead.
 void Channel::SetTransportOverhead(size_t transport_overhead_per_packet) {
+  rtc::CritScope cs(&overhead_per_packet_lock_);
   transport_overhead_per_packet_ = transport_overhead_per_packet;
-  UpdateOverheadForEncoder();
+  UpdateOverheadForEncoder(transport_overhead_per_packet_ +
+                           rtp_overhead_per_packet_);
 }
 
 void Channel::OnOverheadChanged(size_t overhead_bytes_per_packet) {
+  rtc::CritScope cs(&overhead_per_packet_lock_);
   rtp_overhead_per_packet_ = overhead_bytes_per_packet;
-  UpdateOverheadForEncoder();
+  UpdateOverheadForEncoder(transport_overhead_per_packet_ +
+                           rtp_overhead_per_packet_);
 }
 
 int Channel::GetNetworkStatistics(NetworkStatistics& stats) {
   return audio_coding_->GetNetworkStatistics(&stats);
 }
 
 void Channel::GetDecodingCallStatistics(AudioDecodingCallStats* stats) const {
   audio_coding_->GetDecodingCallStatistics(stats);
 }
 
@@ skipping 268 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc