Fix TSAN race in webrtc::voe::Channel

webrtc/voice_engine/channel.h

 /*
  *  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 398 matching lines @@
   int ResendPackets(const uint16_t* sequence_numbers, int length);
   int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
   int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
   void UpdatePlayoutTimestamp(bool rtcp);
   void RegisterReceiveCodecsToRTPModule();
 
   int SetSendRtpHeaderExtension(bool enable,
                                 RTPExtensionType type,
                                 unsigned char id);
 
-  void UpdateOverheadForEncoder();
+  void UpdateOverheadForEncoder(size_t overhead_per_packet);

[hta-webrtc, 2017/02/28 12:21:20]

Should there be a lock notation here saying "must be called with
overhead_per_packet_lock held"?

It's private, so I guess it's OK to leave it out.

[hbos, 2017/02/28 12:32:53]

Good idea! There's EXCLUSIVE_LOCKS_REQUIRED. Now using that, which also meant I
could remove the function parameter and calculate it inside of the function
instead knowing the lock is already held.

 
   int GetRtpTimestampRateHz() const;
   int64_t GetRTT(bool allow_associate_channel) const;
 
   rtc::CriticalSection _fileCritSect;
   rtc::CriticalSection _callbackCritSect;
   rtc::CriticalSection volume_settings_critsect_;
   uint32_t _instanceId;
   int32_t _channelId;
 
@@ skipping 60 matching lines @@
   // VoEVolumeControl
   bool input_mute_ GUARDED_BY(volume_settings_critsect_);
   bool previous_frame_muted_;  // Only accessed from PrepareEncodeAndSend().
   float _panLeft GUARDED_BY(volume_settings_critsect_);
   float _panRight GUARDED_BY(volume_settings_critsect_);
   float _outputGain GUARDED_BY(volume_settings_critsect_);
   // VoeRTP_RTCP
   uint32_t _lastLocalTimeStamp;
   int8_t _lastPayloadType;
   bool _includeAudioLevelIndication;
-  size_t transport_overhead_per_packet_;
-  size_t rtp_overhead_per_packet_;
+  size_t transport_overhead_per_packet_ GUARDED_BY(overhead_per_packet_lock_);
+  size_t rtp_overhead_per_packet_ GUARDED_BY(overhead_per_packet_lock_);
+  rtc::CriticalSection overhead_per_packet_lock_;
   // VoENetwork
   AudioFrame::SpeechType _outputSpeechType;
   // VoEVideoSync
   rtc::CriticalSection video_sync_lock_;
   // VoEAudioProcessing
   bool restored_packet_in_use_;
   // RtcpBandwidthObserver
   std::unique_ptr<VoERtcpObserver> rtcp_observer_;
   // An associated send channel.
   rtc::CriticalSection assoc_send_channel_lock_;
   ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 
   bool pacing_enabled_;
   PacketRouter* packet_router_ = nullptr;
   std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
   std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
   std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_