Resolves TSan v2 warnings in voe_auto_test.

See bug report for details.

BUG=1590
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5714 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/voice_engine/channel.cc

@@ -410,7 +410,7 @@ Channel::OnPacketTimeout(int32_t id)
     CriticalSectionScoped cs(_callbackCritSectPtr);
     if (_voiceEngineObserverPtr)
     {
-        if (_receiving || _externalTransport)
+        if (channel_state_.Get().receiving || _externalTransport)
         {
             int32_t channel = VoEChannelId(id);
             assert(channel == _channelId);
@@ -488,7 +488,7 @@ Channel::OnPeriodicDeadOrAlive(int32_t id,
     // It is possible that the connection is alive even if no RTP packet has
     // been received for a long time since the other side might use VAD/DTX
     // and a low SID-packet update rate.
-    if ((kRtpNoRtp == alive) && _playing)
+    if ((kRtpNoRtp == alive) && channel_state_.Get().playing)
     {
         // Detect Alive for all NetEQ states except for the case when we are
         // in PLC_CNG state.
@@ -527,7 +527,7 @@ Channel::OnReceivedPayloadData(const uint8_t* payloadData,
 
     _lastRemoteTimeStamp = rtpHeader->header.timestamp;
 
-    if (!_playing)
+    if (!channel_state_.Get().playing)
     {
         // Avoid inserting into NetEQ when we are not playing. Count the
         // packet as discarded.
@@ -612,7 +612,9 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
     // Store speech type for dead-or-alive detection
     _outputSpeechType = audioFrame.speech_type_;
 
-    if (_rxApmIsEnabled) {
+    ChannelState::State state = channel_state_.Get();
+
+    if (state.rx_apm_is_enabled) {
       int err = rx_audioproc_->ProcessStream(&audioFrame);
       if (err) {
         LOG(LS_ERROR) << "ProcessStream() error: " << err;
@@ -656,13 +658,13 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame& audioFrame)
     }
 
     // Mix decoded PCM output with file if file mixing is enabled
-    if (_outputFilePlaying)
+    if (state.output_file_playing)
     {
         MixAudioWithFile(audioFrame, audioFrame.sample_rate_hz_);
     }
 
     // Place channel in on-hold state (~muted) if on-hold is activated
-    if (_outputIsOnHold)
+    if (state.output_is_on_hold)
     {
         AudioFrameOperations::Mute(audioFrame);
     }
@@ -725,10 +727,10 @@ Channel::NeededFrequency(int32_t id)
     // we take that frequency into consideration as well
     // This is not needed on sending side, since the codec will
     // limit the spectrum anyway.
-    if (_outputFilePlaying)
+    if (channel_state_.Get().output_file_playing)
     {
         CriticalSectionScoped cs(&_fileCritSect);
-        if (_outputFilePlayerPtr && _outputFilePlaying)
+        if (_outputFilePlayerPtr)
         {
             if(_outputFilePlayerPtr->Frequency()>highestNeeded)
             {
@@ -790,9 +792,7 @@ Channel::PlayFileEnded(int32_t id)
 
     if (id == _inputFilePlayerId)
     {
-        CriticalSectionScoped cs(&_fileCritSect);
+        channel_state_.SetInputFilePlaying(false);
-
-        _inputFilePlaying = false;
         WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
                      VoEId(_instanceId,_channelId),
                      "Channel::PlayFileEnded() => input file player module is"
@@ -800,9 +800,7 @@ Channel::PlayFileEnded(int32_t id)
     }
     else if (id == _outputFilePlayerId)
     {
-        CriticalSectionScoped cs(&_fileCritSect);
+        channel_state_.SetOutputFilePlaying(false);
-
-        _outputFilePlaying = false;
         WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
                      VoEId(_instanceId,_channelId),
                      "Channel::PlayFileEnded() => output file player module is"
@@ -860,12 +858,9 @@ Channel::Channel(int32_t channelId,
     _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
     _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
     _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
-    _inputFilePlaying(false),
-    _outputFilePlaying(false),
     _outputFileRecording(false),
     _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
     _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
-    _inputExternalMedia(false),
     _outputExternalMedia(false),
     _inputExternalMediaCallbackPtr(NULL),
     _outputExternalMediaCallbackPtr(NULL),
@@ -891,13 +886,9 @@ Channel::Channel(int32_t channelId,
     _sendFrameType(0),
     _rtpObserverPtr(NULL),
     _rtcpObserverPtr(NULL),
-    _outputIsOnHold(false),
     _externalPlayout(false),
     _externalMixing(false),
     _inputIsOnHold(false),
-    _playing(false),
-    _sending(false),
-    _receiving(false),
     _mixFileWithMicrophone(false),
     _rtpObserver(false),
     _rtcpObserver(false),
@@ -924,7 +915,6 @@ Channel::Channel(int32_t channelId,
     _previousTimestamp(0),
     _recPacketDelayMs(20),
     _RxVadDetection(false),
-    _rxApmIsEnabled(false),
     _rxAgcIsEnabled(false),
     _rxNsIsEnabled(false),
     restored_packet_in_use_(false)
@@ -960,7 +950,7 @@ Channel::~Channel()
     {
         DeRegisterExternalMediaProcessing(kPlaybackPerChannel);
     }
-    if (_inputExternalMedia)
+    if (channel_state_.Get().input_external_media)
     {
         DeRegisterExternalMediaProcessing(kRecordingPerChannel);
     }
@@ -1033,6 +1023,8 @@ Channel::Init()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::Init()");
 
+    channel_state_.Reset();
+
     // --- Initial sanity
 
     if ((_engineStatisticsPtr == NULL) ||
@@ -1231,7 +1223,7 @@ Channel::StartPlayout()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StartPlayout()");
-    if (_playing)
+    if (channel_state_.Get().playing)
     {
         return 0;
     }
@@ -1247,8 +1239,7 @@ Channel::StartPlayout()
         }
     }
 
-    _playing = true;
+    channel_state_.SetPlaying(true);
-
     if (RegisterFilePlayingToMixer() != 0)
         return -1;
 
@@ -1260,7 +1251,7 @@ Channel::StopPlayout()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StopPlayout()");
-    if (!_playing)
+    if (!channel_state_.Get().playing)
     {
         return 0;
     }
@@ -1276,7 +1267,7 @@ Channel::StopPlayout()
         }
     }
 
-    _playing = false;
+    channel_state_.SetPlaying(false);
     _outputAudioLevel.Clear();
 
     return 0;
@@ -1288,21 +1279,15 @@ Channel::StartSend()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StartSend()");
     // Resume the previous sequence number which was reset by StopSend().
-    // This needs to be done before |_sending| is set to true.
+    // This needs to be done before |sending| is set to true.
     if (send_sequence_number_)
       SetInitSequenceNumber(send_sequence_number_);
 
-    {
+    if (channel_state_.Get().sending)
-        // A lock is needed because |_sending| can be accessed or modified by
-        // another thread at the same time.
-        CriticalSectionScoped cs(&_callbackCritSect);
-
-        if (_sending)
     {
       return 0;
     }
-        _sending = true;
+    channel_state_.SetSending(true);
-    }
 
     if (_rtpRtcpModule->SetSendingStatus(true) != 0)
     {
@@ -1310,7 +1295,7 @@ Channel::StartSend()
             VE_RTP_RTCP_MODULE_ERROR, kTraceError,
             "StartSend() RTP/RTCP failed to start sending");
         CriticalSectionScoped cs(&_callbackCritSect);
-        _sending = false;
+        channel_state_.SetSending(false);
         return -1;
     }
 
@@ -1322,17 +1307,11 @@ Channel::StopSend()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StopSend()");
-    {
+    if (!channel_state_.Get().sending)
-        // A lock is needed because |_sending| can be accessed or modified by
-        // another thread at the same time.
-        CriticalSectionScoped cs(&_callbackCritSect);
-
-        if (!_sending)
     {
       return 0;
     }
-        _sending = false;
+    channel_state_.SetSending(false);
-    }
 
     // Store the sequence number to be able to pick up the same sequence for
     // the next StartSend(). This is needed for restarting device, otherwise
@@ -1360,11 +1339,11 @@ Channel::StartReceiving()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StartReceiving()");
-    if (_receiving)
+    if (channel_state_.Get().receiving)
     {
         return 0;
     }
-    _receiving = true;
+    channel_state_.SetReceiving(true);
     _numberOfDiscardedPackets = 0;
     return 0;
 }
@@ -1374,7 +1353,7 @@ Channel::StopReceiving()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StopReceiving()");
-    if (!_receiving)
+    if (!channel_state_.Get().receiving)
     {
         return 0;
     }
@@ -1382,7 +1361,7 @@ Channel::StopReceiving()
     // Recover DTMF detection status.
     telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
     RegisterReceiveCodecsToRTPModule();
-    _receiving = false;
+    channel_state_.SetReceiving(false);
     return 0;
 }
 
@@ -1448,12 +1427,12 @@ Channel::SetOnHoldStatus(bool enable, OnHoldModes mode)
                  "Channel::SetOnHoldStatus()");
     if (mode == kHoldSendAndPlay)
     {
-        _outputIsOnHold = enable;
+        channel_state_.SetOutputIsOnHold(enable);
         _inputIsOnHold = enable;
     }
     else if (mode == kHoldPlayOnly)
     {
-        _outputIsOnHold = enable;
+        channel_state_.SetOutputIsOnHold(enable);
     }
     if (mode == kHoldSendOnly)
     {
@@ -1467,16 +1446,17 @@ Channel::GetOnHoldStatus(bool& enabled, OnHoldModes& mode)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::GetOnHoldStatus()");
-    enabled = (_outputIsOnHold || _inputIsOnHold);
+    bool output_is_on_hold = channel_state_.Get().output_is_on_hold;
-    if (_outputIsOnHold && _inputIsOnHold)
+    enabled = (output_is_on_hold || _inputIsOnHold);
+    if (output_is_on_hold && _inputIsOnHold)
     {
         mode = kHoldSendAndPlay;
     }
-    else if (_outputIsOnHold && !_inputIsOnHold)
+    else if (output_is_on_hold && !_inputIsOnHold)
     {
         mode = kHoldPlayOnly;
     }
-    else if (!_outputIsOnHold && _inputIsOnHold)
+    else if (!output_is_on_hold && _inputIsOnHold)
     {
         mode = kHoldSendOnly;
     }
@@ -1609,14 +1589,14 @@ Channel::SetRecPayloadType(const CodecInst& codec)
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SetRecPayloadType()");
 
-    if (_playing)
+    if (channel_state_.Get().playing)
     {
         _engineStatisticsPtr->SetLastError(
             VE_ALREADY_PLAYING, kTraceError,
             "SetRecPayloadType() unable to set PT while playing");
         return -1;
     }
-    if (_receiving)
+    if (channel_state_.Get().receiving)
     {
         _engineStatisticsPtr->SetLastError(
             VE_ALREADY_LISTENING, kTraceError,
@@ -1921,7 +1901,7 @@ Channel::SetISACMaxRate(int rateBps)
             return -1;
         }
     }
-    if (_sending)
+    if (channel_state_.Get().sending)
     {
         _engineStatisticsPtr->SetLastError(
             VE_SENDING, kTraceError,
@@ -1984,7 +1964,7 @@ Channel::SetISACMaxPayloadSize(int sizeBytes)
             return -1;
         }
     }
-    if (_sending)
+    if (channel_state_.Get().sending)
     {
         _engineStatisticsPtr->SetLastError(
             VE_SENDING, kTraceError,
@@ -2187,7 +2167,7 @@ int Channel::StartPlayingFileLocally(const char* fileName,
                  "stopPosition=%d)", fileName, loop, format, volumeScaling,
                  startPosition, stopPosition);
 
-    if (_outputFilePlaying)
+    if (channel_state_.Get().output_file_playing)
     {
         _engineStatisticsPtr->SetLastError(
             VE_ALREADY_PLAYING, kTraceError,
@@ -2236,7 +2216,7 @@ int Channel::StartPlayingFileLocally(const char* fileName,
             return -1;
         }
         _outputFilePlayerPtr->RegisterModuleFileCallback(this);
-        _outputFilePlaying = true;
+        channel_state_.SetOutputFilePlaying(true);
     }
 
     if (RegisterFilePlayingToMixer() != 0)
@@ -2266,7 +2246,7 @@ int Channel::StartPlayingFileLocally(InStream* stream,
     }
 
 
-    if (_outputFilePlaying)
+    if (channel_state_.Get().output_file_playing)
     {
         _engineStatisticsPtr->SetLastError(
             VE_ALREADY_PLAYING, kTraceError,
@@ -2314,7 +2294,7 @@ int Channel::StartPlayingFileLocally(InStream* stream,
           return -1;
       }
       _outputFilePlayerPtr->RegisterModuleFileCallback(this);
-      _outputFilePlaying = true;
+      channel_state_.SetOutputFilePlaying(true);
     }
 
     if (RegisterFilePlayingToMixer() != 0)
@@ -2328,7 +2308,7 @@ int Channel::StopPlayingFileLocally()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StopPlayingFileLocally()");
 
-    if (!_outputFilePlaying)
+    if (!channel_state_.Get().output_file_playing)
     {
         _engineStatisticsPtr->SetLastError(
             VE_INVALID_OPERATION, kTraceWarning,
@@ -2349,7 +2329,7 @@ int Channel::StopPlayingFileLocally()
         _outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
         FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
         _outputFilePlayerPtr = NULL;
-        _outputFilePlaying = false;
+        channel_state_.SetOutputFilePlaying(false);
     }
     // _fileCritSect cannot be taken while calling
     // SetAnonymousMixibilityStatus. Refer to comments in
@@ -2371,7 +2351,7 @@ int Channel::IsPlayingFileLocally() const
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::IsPlayingFileLocally()");
 
-    return (int32_t)_outputFilePlaying;
+    return channel_state_.Get().output_file_playing;
 }
 
 int Channel::RegisterFilePlayingToMixer()
@@ -2379,7 +2359,8 @@ int Channel::RegisterFilePlayingToMixer()
     // Return success for not registering for file playing to mixer if:
     // 1. playing file before playout is started on that channel.
     // 2. starting playout without file playing on that channel.
-    if (!_playing || !_outputFilePlaying)
+    if (!channel_state_.Get().playing ||
+        !channel_state_.Get().output_file_playing)
     {
         return 0;
     }
@@ -2390,8 +2371,8 @@ int Channel::RegisterFilePlayingToMixer()
     // the file, _fileCritSect will be taken. This would result in a deadlock.
     if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0)
     {
+        channel_state_.SetOutputFilePlaying(false);
         CriticalSectionScoped cs(&_fileCritSect);
-        _outputFilePlaying = false;
         _engineStatisticsPtr->SetLastError(
             VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
             "StartPlayingFile() failed to add participant as file to mixer");
@@ -2411,7 +2392,7 @@ int Channel::ScaleLocalFilePlayout(float scale)
 
     CriticalSectionScoped cs(&_fileCritSect);
 
-    if (!_outputFilePlaying)
+    if (!channel_state_.Get().output_file_playing)
     {
         _engineStatisticsPtr->SetLastError(
             VE_INVALID_OPERATION, kTraceError,
@@ -2473,7 +2454,9 @@ int Channel::StartPlayingFileAsMicrophone(const char* fileName,
                  "stopPosition=%d)", fileName, loop, format, volumeScaling,
                  startPosition, stopPosition);
 
-    if (_inputFilePlaying)
+    CriticalSectionScoped cs(&_fileCritSect);
+
+    if (channel_state_.Get().input_file_playing)
     {
         _engineStatisticsPtr->SetLastError(
             VE_ALREADY_PLAYING, kTraceWarning,
@@ -2481,8 +2464,6 @@ int Channel::StartPlayingFileAsMicrophone(const char* fileName,
         return 0;
     }
 
-    CriticalSectionScoped cs(&_fileCritSect);
-
     // Destroy the old instance
     if (_inputFilePlayerPtr)
     {
@@ -2523,7 +2504,7 @@ int Channel::StartPlayingFileAsMicrophone(const char* fileName,
         return -1;
     }
     _inputFilePlayerPtr->RegisterModuleFileCallback(this);
-    _inputFilePlaying = true;
+    channel_state_.SetInputFilePlaying(true);
 
     return 0;
 }
@@ -2548,7 +2529,9 @@ int Channel::StartPlayingFileAsMicrophone(InStream* stream,
         return -1;
     }
 
-    if (_inputFilePlaying)
+    CriticalSectionScoped cs(&_fileCritSect);
+
+    if (channel_state_.Get().input_file_playing)
     {
         _engineStatisticsPtr->SetLastError(
             VE_ALREADY_PLAYING, kTraceWarning,
@@ -2556,8 +2539,6 @@ int Channel::StartPlayingFileAsMicrophone(InStream* stream,
         return 0;
     }
 
-    CriticalSectionScoped cs(&_fileCritSect);
-
     // Destroy the old instance
     if (_inputFilePlayerPtr)
     {
@@ -2594,7 +2575,7 @@ int Channel::StartPlayingFileAsMicrophone(InStream* stream,
     }
 
     _inputFilePlayerPtr->RegisterModuleFileCallback(this);
-    _inputFilePlaying = true;
+    channel_state_.SetInputFilePlaying(true);
 
     return 0;
 }
@@ -2604,7 +2585,9 @@ int Channel::StopPlayingFileAsMicrophone()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::StopPlayingFileAsMicrophone()");
 
-    if (!_inputFilePlaying)
+    CriticalSectionScoped cs(&_fileCritSect);
+
+    if (!channel_state_.Get().input_file_playing)
     {
         _engineStatisticsPtr->SetLastError(
             VE_INVALID_OPERATION, kTraceWarning,
@@ -2612,7 +2595,6 @@ int Channel::StopPlayingFileAsMicrophone()
         return 0;
     }
 
-    CriticalSectionScoped cs(&_fileCritSect);
     if (_inputFilePlayerPtr->StopPlayingFile() != 0)
     {
         _engineStatisticsPtr->SetLastError(
@@ -2623,7 +2605,7 @@ int Channel::StopPlayingFileAsMicrophone()
     _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
     FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
     _inputFilePlayerPtr = NULL;
-    _inputFilePlaying = false;
+    channel_state_.SetInputFilePlaying(false);
 
     return 0;
 }
@@ -2632,8 +2614,7 @@ int Channel::IsPlayingFileAsMicrophone() const
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::IsPlayingFileAsMicrophone()");
-
+    return channel_state_.Get().input_file_playing;
-    return _inputFilePlaying;
 }
 
 int Channel::ScaleFileAsMicrophonePlayout(float scale)
@@ -2643,7 +2624,7 @@ int Channel::ScaleFileAsMicrophonePlayout(float scale)
 
     CriticalSectionScoped cs(&_fileCritSect);
 
-    if (!_inputFilePlaying)
+    if (!channel_state_.Get().input_file_playing)
     {
         _engineStatisticsPtr->SetLastError(
             VE_INVALID_OPERATION, kTraceError,
@@ -2852,6 +2833,7 @@ int Channel::StopRecordingPlayout()
 void
 Channel::SetMixWithMicStatus(bool mix)
 {
+    CriticalSectionScoped cs(&_fileCritSect);
     _mixFileWithMicrophone=mix;
 }
 
@@ -3155,7 +3137,7 @@ Channel::SetRxAgcStatus(bool enable, AgcModes mode)
     }
 
     _rxAgcIsEnabled = enable;
-    _rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true));
+    channel_state_.SetRxApmIsEnabled(_rxAgcIsEnabled || _rxNsIsEnabled);
 
     return 0;
 }
@@ -3304,7 +3286,7 @@ Channel::SetRxNsStatus(bool enable, NsModes mode)
     }
 
     _rxNsIsEnabled = enable;
-    _rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true));
+    channel_state_.SetRxApmIsEnabled(_rxAgcIsEnabled || _rxNsIsEnabled);
 
     return 0;
 }
@@ -3435,7 +3417,7 @@ Channel::SetLocalSSRC(unsigned int ssrc)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::SetLocalSSRC()");
-    if (_sending)
+    if (channel_state_.Get().sending)
     {
         _engineStatisticsPtr->SetLastError(
             VE_ALREADY_SENDING, kTraceError,
@@ -3722,7 +3704,7 @@ Channel::SendApplicationDefinedRTCPPacket(unsigned char subType,
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::SendApplicationDefinedRTCPPacket()");
-    if (!_sending)
+    if (!channel_state_.Get().sending)
     {
         _engineStatisticsPtr->SetLastError(
             VE_NOT_SENDING, kTraceError,
@@ -4202,7 +4184,7 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)
         return -1;
     }
 
-    if (_inputFilePlaying)
+    if (channel_state_.Get().input_file_playing)
     {
         MixOrReplaceAudioWithFile(mixingFrequency);
     }
@@ -4212,7 +4194,7 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)
         AudioFrameOperations::Mute(_audioFrame);
     }
 
-    if (_inputExternalMedia)
+    if (channel_state_.Get().input_external_media)
     {
         CriticalSectionScoped cs(&_callbackCritSect);
         const bool isStereo = (_audioFrame.num_channels_ == 2);
@@ -4311,7 +4293,7 @@ int Channel::RegisterExternalMediaProcessing(
             return -1;
         }
         _inputExternalMediaCallbackPtr = &processObject;
-        _inputExternalMedia = true;
+        channel_state_.SetInputExternalMedia(true);
     }
     return 0;
 }
@@ -4346,7 +4328,7 @@ int Channel::DeRegisterExternalMediaProcessing(ProcessingTypes type)
                 "input external media already disabled");
             return 0;
         }
-        _inputExternalMedia = false;
+        channel_state_.SetInputExternalMedia(false);
         _inputExternalMediaCallbackPtr = NULL;
     }
 
@@ -4357,7 +4339,7 @@ int Channel::SetExternalMixing(bool enabled) {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SetExternalMixing(enabled=%d)", enabled);
 
-    if (_playing)
+    if (channel_state_.Get().playing)
     {
         _engineStatisticsPtr->SetLastError(
             VE_INVALID_OPERATION, kTraceError,
@@ -4583,7 +4565,7 @@ Channel::SetInitTimestamp(unsigned int timestamp)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::SetInitTimestamp()");
-    if (_sending)
+    if (channel_state_.Get().sending)
     {
         _engineStatisticsPtr->SetLastError(
             VE_SENDING, kTraceError, "SetInitTimestamp() already sending");
@@ -4604,7 +4586,7 @@ Channel::SetInitSequenceNumber(short sequenceNumber)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SetInitSequenceNumber()");
-    if (_sending)
+    if (channel_state_.Get().sending)
     {
         _engineStatisticsPtr->SetLastError(
             VE_SENDING, kTraceError,

webrtc/voice_engine/channel.h

@@ -63,6 +63,91 @@ class StatisticsProxy;
 class TransmitMixer;
 class OutputMixer;
 
+// Helper class to simplify locking scheme for members that are accessed from
+// multiple threads.
+// Example: a member can be set on thread T1 and read by an internal audio
+// thread T2. Accessing the member via this class ensures that we are
+// safe and also avoid TSan v2 warnings.
+class ChannelState {
+ public:
+    struct State {
+        State() : rx_apm_is_enabled(false),
+                  input_external_media(false),
+                  output_is_on_hold(false),
+                  output_file_playing(false),
+                  input_file_playing(false),
+                  playing(false),
+                  sending(false),
+                  receiving(false) {}
+
+        bool rx_apm_is_enabled;
+        bool input_external_media;
+        bool output_is_on_hold;
+        bool output_file_playing;
+        bool input_file_playing;
+        bool playing;
+        bool sending;
+        bool receiving;
+    };
+
+    ChannelState() : lock_(CriticalSectionWrapper::CreateCriticalSection()) {
+    }
+    virtual ~ChannelState() {}
+
+    void Reset() {
+        CriticalSectionScoped lock(lock_.get());
+        state_ = State();
+    }
+
+    State Get() const {
+        CriticalSectionScoped lock(lock_.get());
+        return state_;
+    }
+
+    void SetRxApmIsEnabled(bool enable) {
+        CriticalSectionScoped lock(lock_.get());
+        state_.rx_apm_is_enabled = enable;
+    }
+
+    void SetInputExternalMedia(bool enable) {
+        CriticalSectionScoped lock(lock_.get());
+        state_.input_external_media = enable;
+    }
+
+    void SetOutputIsOnHold(bool enable) {
+        CriticalSectionScoped lock(lock_.get());
+        state_.output_is_on_hold = enable;
+    }
+
+    void SetOutputFilePlaying(bool enable) {
+        CriticalSectionScoped lock(lock_.get());
+        state_.output_file_playing = enable;
+    }
+
+    void SetInputFilePlaying(bool enable) {
+        CriticalSectionScoped lock(lock_.get());
+        state_.input_file_playing = enable;
+    }
+
+    void SetPlaying(bool enable) {
+        CriticalSectionScoped lock(lock_.get());
+        state_.playing = enable;
+    }
+
+    void SetSending(bool enable) {
+        CriticalSectionScoped lock(lock_.get());
+        state_.sending = enable;
+    }
+
+    void SetReceiving(bool enable) {
+        CriticalSectionScoped lock(lock_.get());
+        state_.receiving = enable;
+    }
+
+private:
+    scoped_ptr<CriticalSectionWrapper> lock_;
+    State state_;
+};
 
 class Channel:
     public RtpData,
@@ -371,32 +456,25 @@ public:
     }
     bool Playing() const
     {
-        return _playing;
+        return channel_state_.Get().playing;
     }
     bool Sending() const
     {
-        // A lock is needed because |_sending| is accessed by both
+        return channel_state_.Get().sending;
-        // TransmitMixer::PrepareDemux() and StartSend()/StopSend(), which
-        // are called by different threads.
-        CriticalSectionScoped cs(&_callbackCritSect);
-        return _sending;
     }
     bool Receiving() const
     {
-        return _receiving;
+        return channel_state_.Get().receiving;
     }
     bool ExternalTransport() const
     {
+        CriticalSectionScoped cs(&_callbackCritSect);
         return _externalTransport;
     }
     bool ExternalMixing() const
     {
         return _externalMixing;
     }
-    bool OutputIsOnHold() const
-    {
-        return _outputIsOnHold;
-    }
     bool InputIsOnHold() const
     {
         return _inputIsOnHold;
@@ -448,6 +526,8 @@ private:
     uint32_t _instanceId;
     int32_t _channelId;
 
+    ChannelState channel_state_;
+
     scoped_ptr<RtpHeaderParser> rtp_header_parser_;
     scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
     scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
@@ -471,12 +551,9 @@ private:
     int _inputFilePlayerId;
     int _outputFilePlayerId;
     int _outputFileRecorderId;
-    bool _inputFilePlaying;
-    bool _outputFilePlaying;
     bool _outputFileRecording;
     DtmfInbandQueue _inbandDtmfQueue;
     DtmfInband _inbandDtmfGenerator;
-    bool _inputExternalMedia;
     bool _outputExternalMedia;
     VoEMediaProcess* _inputExternalMediaCallbackPtr;
     VoEMediaProcess* _outputExternalMediaCallbackPtr;
@@ -509,13 +586,9 @@ private:
     VoERTPObserver* _rtpObserverPtr;
     VoERTCPObserver* _rtcpObserverPtr;
     // VoEBase
-    bool _outputIsOnHold;
     bool _externalPlayout;
     bool _externalMixing;
     bool _inputIsOnHold;
-    bool _playing;
-    bool _sending;
-    bool _receiving;
     bool _mixFileWithMicrophone;
     bool _rtpObserver;
     bool _rtcpObserver;
@@ -548,7 +621,6 @@ private:
     uint16_t _recPacketDelayMs;
     // VoEAudioProcessing
     bool _RxVadDetection;
-    bool _rxApmIsEnabled;
     bool _rxAgcIsEnabled;
     bool _rxNsIsEnabled;
     bool restored_packet_in_use_;

webrtc/voice_engine/dtmf_inband_queue.cc

@@ -74,12 +74,14 @@ DtmfInbandQueue::NextDtmf(uint16_t* len, uint8_t* level)
 bool
 DtmfInbandQueue::PendingDtmf()
 {
-    return(_nextEmptyIndex>0);        
+    CriticalSectionScoped lock(&_DtmfCritsect);
+    return _nextEmptyIndex > 0;
 }
 
 void
 DtmfInbandQueue::ResetDtmf()
 {
+    CriticalSectionScoped lock(&_DtmfCritsect);
     _nextEmptyIndex = 0;
 }
 

webrtc/voice_engine/output_mixer.cc

@@ -569,11 +569,11 @@ OutputMixer::DoOperationsOnCombinedSignal()
     APMAnalyzeReverseStream();
 
     // --- External media processing
-
-    if (_externalMedia)
     {
         CriticalSectionScoped cs(&_callbackCritSect);
-        const bool isStereo = (_audioFrame.num_channels_ == 2);
+        if (_externalMedia)
+        {
+            const bool is_stereo = (_audioFrame.num_channels_ == 2);
             if (_externalMediaCallbackPtr)
             {
                 _externalMediaCallbackPtr->Process(
@@ -582,7 +582,8 @@ OutputMixer::DoOperationsOnCombinedSignal()
                     (int16_t*)_audioFrame.data_,
                     _audioFrame.samples_per_channel_,
                     _audioFrame.sample_rate_hz_,
-                isStereo);
+                    is_stereo);
+            }
         }
     }
 

webrtc/voice_engine/transmit_mixer.cc

@@ -397,7 +397,12 @@ TransmitMixer::PrepareDemux(const void* audioSamples,
     }
 
     // --- Record to file
-    if (_fileRecording)
+    bool file_recording = false;
+    {
+        CriticalSectionScoped cs(&_critSect);
+        file_recording =  _fileRecording;
+    }
+    if (file_recording)
     {
         RecordAudioToFile(_audioFrame.sample_rate_hz_);
     }
@@ -728,6 +733,8 @@ int TransmitMixer::StartRecordingMicrophone(const char* fileName,
                  "TransmitMixer::StartRecordingMicrophone(fileName=%s)",
                  fileName);
 
+    CriticalSectionScoped cs(&_critSect);
+
     if (_fileRecording)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
@@ -761,8 +768,6 @@ int TransmitMixer::StartRecordingMicrophone(const char* fileName,
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(&_critSect);
-
     // Destroy the old instance
     if (_fileRecorderPtr)
     {
@@ -807,6 +812,8 @@ int TransmitMixer::StartRecordingMicrophone(OutStream* stream,
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                "TransmitMixer::StartRecordingMicrophone()");
 
+    CriticalSectionScoped cs(&_critSect);
+
     if (_fileRecording)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
@@ -839,8 +846,6 @@ int TransmitMixer::StartRecordingMicrophone(OutStream* stream,
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(&_critSect);
-
     // Destroy the old instance
     if (_fileRecorderPtr)
     {
@@ -884,6 +889,8 @@ int TransmitMixer::StopRecordingMicrophone()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::StopRecordingMicrophone()");
 
+    CriticalSectionScoped cs(&_critSect);
+
     if (!_fileRecording)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
@@ -891,8 +898,6 @@ int TransmitMixer::StopRecordingMicrophone()
         return 0;
     }
 
-    CriticalSectionScoped cs(&_critSect);
-
     if (_fileRecorderPtr->StopRecording() != 0)
     {
         _engineStatisticsPtr->SetLastError(
@@ -1170,7 +1175,7 @@ bool TransmitMixer::IsRecordingCall()
 
 bool TransmitMixer::IsRecordingMic()
 {
-
+    CriticalSectionScoped cs(&_critSect);
     return _fileRecording;
 }
 

webrtc/voice_engine/voice_engine_impl.cc

@@ -70,6 +70,12 @@ int VoiceEngineImpl::Release() {
                  "VoiceEngineImpl self deleting (voiceEngine=0x%p)",
                  this);
 
+    // Clear any pointers before starting destruction. Otherwise worker-
+    // threads will still have pointers to a partially destructed object.
+    // Example: AudioDeviceBuffer::RequestPlayoutData() can access a
+    // partially deconstructed |_ptrCbAudioTransport| during destruction
+    // if we don't call Terminate here.
+    Terminate();
     delete this;
   }