Adding locking to webrtc::voe::Channel to fix race conditions

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 256 matching lines @@
     int SetChannelOutputVolumeScaling(float scaling);
     int GetChannelOutputVolumeScaling(float& scaling) const;
 
     // VoENetEqStats
     int GetNetworkStatistics(NetworkStatistics& stats);
     void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
     // VoEVideoSync
     bool GetDelayEstimate(int* jitter_buffer_delay_ms,
                           int* playout_buffer_delay_ms) const;
-    int least_required_delay_ms() const { return least_required_delay_ms_; }
+    int LeastRequiredDelayMs() const {
+      return audio_coding_->LeastRequiredDelayMs();

[Henrik Grunell WebRTC, 2015/08/12 15:03:22]

Move to implementation file.

[Taylor Brandstetter, 2015/08/12 19:49:04]

Done.

+    }
     int SetInitialPlayoutDelay(int delay_ms);
     int SetMinimumPlayoutDelay(int delayMs);
     int GetPlayoutTimestamp(unsigned int& timestamp);
-    void UpdatePlayoutTimestamp(bool rtcp);
     int SetInitTimestamp(unsigned int timestamp);
     int SetInitSequenceNumber(short sequenceNumber);
 
     // VoEVideoSyncExtended
     int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
     // VoEDtmf
     int SendTelephoneEventOutband(unsigned char eventCode, int lengthMs,
                                   int attenuationDb, bool playDtmfEvent);
     int SendTelephoneEventInband(unsigned char eventCode, int lengthMs,
@@ skipping 165 matching lines @@
     bool HandleRtxPacket(const uint8_t* packet,
                          size_t packet_length,
                          const RTPHeader& header);
     bool IsPacketInOrder(const RTPHeader& header) const;
     bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
     int ResendPackets(const uint16_t* sequence_numbers, int length);
     int InsertInbandDtmfTone();
     int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
     int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
     int32_t SendPacketRaw(const void *data, size_t len, bool RTCP);
+    void UpdatePlayoutTimestamp(bool rtcp);
     void UpdatePacketDelay(uint32_t timestamp,
                            uint16_t sequenceNumber);
     void RegisterReceiveCodecsToRTPModule();
 
     int SetRedPayloadType(int red_payload_type);
     int SetSendRtpHeaderExtension(bool enable, RTPExtensionType type,
                                   unsigned char id);
 
     int32_t GetPlayoutFrequency();
     int64_t GetRTT(bool allow_associate_channel) const;
@@ skipping 32 matching lines @@
     bool _outputExternalMedia;
     VoEMediaProcess* _inputExternalMediaCallbackPtr;
     VoEMediaProcess* _outputExternalMediaCallbackPtr;
     uint32_t _timeStamp;
     uint8_t _sendTelephoneEventPayloadType;
 
     RemoteNtpTimeEstimator ntp_estimator_ GUARDED_BY(ts_stats_lock_);
 
     // Timestamp of the audio pulled from NetEq.
     uint32_t jitter_buffer_playout_timestamp_;
-    uint32_t playout_timestamp_rtp_;
+    uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
     uint32_t playout_timestamp_rtcp_;

[Henrik Grunell WebRTC, 2015/08/12 15:03:22]

Protect also?

[Taylor Brandstetter, 2015/08/12 19:49:04]

I don't think it needs to be protected (at least not for the same reason as
playout_timestamp_rtp_), since it's not accessed by GetPlayoutTimestamp.

[Henrik Grunell WebRTC, 2015/08/13 08:25:02]

Acknowledged.

-    uint32_t playout_delay_ms_;
+    uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
     uint32_t _numberOfDiscardedPackets;
     uint16_t send_sequence_number_;
     uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
 
     rtc::scoped_ptr<CriticalSectionWrapper> ts_stats_lock_;
 
     rtc::scoped_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
     // The rtp timestamp of the first played out audio frame.
     int64_t capture_start_rtp_time_stamp_;
     // The capture ntp time (in local timebase) of the first played out audio
@@ skipping 25 matching lines @@
     // VoEDtmf
     bool _playOutbandDtmfEvent;
     bool _playInbandDtmfEvent;
     // VoeRTP_RTCP
     uint32_t _lastLocalTimeStamp;
     int8_t _lastPayloadType;
     bool _includeAudioLevelIndication;
     // VoENetwork
     AudioFrame::SpeechType _outputSpeechType;
     // VoEVideoSync
-    uint32_t _average_jitter_buffer_delay_us;
-    int least_required_delay_ms_;
+    rtc::scoped_ptr<CriticalSectionWrapper> video_sync_lock_;
+    uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
     uint32_t _previousTimestamp;
-    uint16_t _recPacketDelayMs;
+    uint16_t _recPacketDelayMs GUARDED_BY(video_sync_lock_);
     // VoEAudioProcessing
     bool _RxVadDetection;
     bool _rxAgcIsEnabled;
     bool _rxNsIsEnabled;
     bool restored_packet_in_use_;
     // RtcpBandwidthObserver
     rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_