diff --git a/src/voice_engine/main/source/channel.cc b/src/voice_engine/main/source/channel.cc
index fb24494beb..cd9313714c 100644
--- a/src/voice_engine/main/source/channel.cc
+++ b/src/voice_engine/main/source/channel.cc
@@ -85,7 +85,7 @@ Channel::InFrameType(WebRtc_Word16 frameType)
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::InFrameType(frameType=%d)", frameType);
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
     // 1 indicates speech
     _sendFrameType = (frameType == 1) ? 1 : 0;
     return 0;
@@ -101,7 +101,7 @@ Channel::IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool end)
 
     if (digitDtmf != 999)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         if (_telephoneEventDetectionPtr)
         {
             _telephoneEventDetectionPtr->OnReceivedTelephoneEventInband(
@@ -119,7 +119,7 @@ Channel::OnRxVadDetected(const int vadDecision)
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::OnRxVadDetected(vadDecision=%d)", vadDecision);
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
     if (_rxVadObserverPtr)
     {
         _rxVadObserverPtr->OnRxVad(_channelId, vadDecision);
@@ -174,7 +174,7 @@ Channel::SendPacket(int channel, const void *data, int len)
     // SRTP or External encryption
     if (_encrypting)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         if (_encryptionPtr)
         {
@@ -224,7 +224,7 @@ Channel::SendPacket(int channel, const void *data, int len)
 
     // Packet transmission using external transport transport
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         int n = _transportPtr->SendPacket(channel,
                                           bufferToSendPtr,
@@ -251,7 +251,7 @@ Channel::SendRTCPPacket(int channel, const void *data, int len)
                  "Channel::SendRTCPPacket(channel=%d, len=%d)", channel, len);
 
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         if (_transportPtr == NULL)
         {
             WEBRTC_TRACE(kTraceError, kTraceVoice,
@@ -276,7 +276,7 @@ Channel::SendRTCPPacket(int channel, const void *data, int len)
     // SRTP or External encryption
     if (_encrypting)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         if (_encryptionPtr)
         {
@@ -327,7 +327,7 @@ Channel::SendRTCPPacket(int channel, const void *data, int len)
 
     // Packet transmission using external transport transport
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         int n = _transportPtr->SendRTCPPacket(channel,
                                               bufferToSendPtr,
@@ -371,7 +371,7 @@ Channel::IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
     // SRTP or External decryption
     if (_decrypting)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         if (_encryptionPtr)
         {
@@ -451,7 +451,7 @@ Channel::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
     // SRTP or External decryption
     if (_decrypting)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         if (_encryptionPtr)
         {
@@ -515,7 +515,7 @@ Channel::OnReceivedTelephoneEvent(const WebRtc_Word32 id,
 #ifdef WEBRTC_DTMF_DETECTION
     if (_outOfBandTelephoneEventDetecion)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         if (_telephoneEventDetectionPtr)
         {
@@ -567,7 +567,7 @@ Channel::OnIncomingSSRCChanged(const WebRtc_Word32 id,
 
     if (_rtpObserver)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         if (_rtpObserverPtr)
         {
@@ -590,7 +590,7 @@ void Channel::OnIncomingCSRCChanged(const WebRtc_Word32 id,
 
     if (_rtpObserver)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         if (_rtpObserverPtr)
         {
@@ -616,7 +616,7 @@ Channel::OnApplicationDataReceived(const WebRtc_Word32 id,
 
     if (_rtcpObserver)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         if (_rtcpObserverPtr)
         {
@@ -677,7 +677,7 @@ Channel::OnPacketTimeout(const WebRtc_Word32 id)
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnPacketTimeout(id=%d)", id);
 
-    CriticalSectionScoped cs(*_callbackCritSectPtr);
+    CriticalSectionScoped cs(_callbackCritSectPtr);
     if (_voiceEngineObserverPtr)
     {
         if (_receiving || _externalTransport)
@@ -711,7 +711,7 @@ Channel::OnReceivedPacket(const WebRtc_Word32 id,
     // Notify only for the case when we have restarted an RTP session.
     if (_rtpPacketTimedOut && (kPacketRtp == packetType))
     {
-        CriticalSectionScoped cs(*_callbackCritSectPtr);
+        CriticalSectionScoped cs(_callbackCritSectPtr);
         if (_voiceEngineObserverPtr)
         {
             WebRtc_Word32 channel = VoEChannelId(id);
@@ -772,7 +772,7 @@ Channel::OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
     // Send callback to the registered observer
     if (_connectionObserver)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         if (_connectionObserverPtr)
         {
             _connectionObserverPtr->OnPeriodicDeadOrAlive(channel, isAlive);
@@ -898,7 +898,7 @@ WebRtc_Word32 Channel::GetAudioFrame(const WebRtc_Word32 id,
     // External media
     if (_outputExternalMedia)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         const bool isStereo = (audioFrame._audioChannel == 2);
         if (_outputExternalMediaCallbackPtr)
         {
@@ -914,7 +914,7 @@ WebRtc_Word32 Channel::GetAudioFrame(const WebRtc_Word32 id,
 
     // Record playout if enabled
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
 
         if (_outputFileRecording && _outputFileRecorderPtr)
         {
@@ -968,7 +968,7 @@ Channel::NeededFrequency(const WebRtc_Word32 id)
     // limit the spectrum anyway.
     if (_outputFilePlaying)
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
         if (_outputFilePlayerPtr && _outputFilePlaying)
         {
             if(_outputFilePlayerPtr->Frequency()>highestNeeded)
@@ -1032,7 +1032,7 @@ Channel::PlayFileEnded(const WebRtc_Word32 id)
 
     if (id == _inputFilePlayerId)
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
 
         _inputFilePlaying = false;
         WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
@@ -1042,7 +1042,7 @@ Channel::PlayFileEnded(const WebRtc_Word32 id)
     }
     else if (id == _outputFilePlayerId)
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
 
         _outputFilePlaying = false;
         WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
@@ -1060,7 +1060,7 @@ Channel::RecordFileEnded(const WebRtc_Word32 id)
 
     assert(id == _outputFileRecorderId);
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     _outputFileRecording = false;
     WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
@@ -1073,7 +1073,6 @@ Channel::Channel(const WebRtc_Word32 channelId,
                  const WebRtc_UWord32 instanceId) :
     _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _transmitCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _instanceId(instanceId),
     _channelId(channelId),
     _rtpRtcpModule(*RtpRtcp::CreateRtpRtcp(VoEModuleId(
@@ -1220,7 +1219,7 @@ Channel::~Channel()
     StopPlayout();
 
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
         if (_inputFilePlayerPtr)
         {
             _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
@@ -1352,7 +1351,6 @@ Channel::~Channel()
     delete [] _encryptionRTCPBufferPtr;
     delete [] _decryptionRTCPBufferPtr;
     delete &_callbackCritSect;
-    delete &_transmitCritSect;
     delete &_fileCritSect;
 }
 
@@ -1530,7 +1528,7 @@ Channel::Init()
     {
         // A lock is needed here since users can call
         // RegisterExternalTransport() at the same time.
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         _transportPtr = &_socketTransportModule;
     }
 #endif
@@ -1689,7 +1687,7 @@ Channel::StartSend()
     {
         // A lock is needed because |_sending| can be accessed or modified by
         // another thread at the same time.
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         if (_sending)
         {
@@ -1703,7 +1701,7 @@ Channel::StartSend()
         _engineStatisticsPtr->SetLastError(
             VE_RTP_RTCP_MODULE_ERROR, kTraceError,
             "StartSend() RTP/RTCP failed to start sending");
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         _sending = false;
         return -1;
     }
@@ -1719,7 +1717,7 @@ Channel::StopSend()
     {
         // A lock is needed because |_sending| can be accessed or modified by
         // another thread at the same time.
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
 
         if (!_sending)
         {
@@ -2252,7 +2250,7 @@ Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RegisterVoiceEngineObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_voiceEngineObserverPtr)
     {
@@ -2270,7 +2268,7 @@ Channel::DeRegisterVoiceEngineObserver()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::DeRegisterVoiceEngineObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (!_voiceEngineObserverPtr)
     {
@@ -2769,7 +2767,7 @@ WebRtc_Word32 Channel::RegisterExternalTransport(Transport& transport)
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterExternalTransport()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
 #ifndef WEBRTC_EXTERNAL_TRANSPORT
     // Sanity checks for default (non external transport) to avoid conflict with
@@ -2807,7 +2805,7 @@ Channel::DeRegisterExternalTransport()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::DeRegisterExternalTransport()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (!_transportPtr)
     {
@@ -3252,7 +3250,7 @@ Channel::RegisterDeadOrAliveObserver(VoEConnectionObserver& observer)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RegisterDeadOrAliveObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_connectionObserverPtr)
     {
@@ -3272,7 +3270,7 @@ Channel::DeRegisterDeadOrAliveObserver()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::DeRegisterDeadOrAliveObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (!_connectionObserverPtr)
     {
@@ -3425,7 +3423,7 @@ int Channel::StartPlayingFileLocally(const char* fileName,
     }
 
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
 
         if (_outputFilePlayerPtr)
         {
@@ -3473,7 +3471,7 @@ int Channel::StartPlayingFileLocally(const char* fileName,
     // the file, _fileCritSect will be taken. This would result in a deadlock.
     if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0)
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
         _outputFilePlaying = false;
         _engineStatisticsPtr->SetLastError(
             VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
@@ -3517,7 +3515,7 @@ int Channel::StartPlayingFileLocally(InStream* stream,
     }
 
     {
-      CriticalSectionScoped cs(_fileCritSect);
+      CriticalSectionScoped cs(&_fileCritSect);
 
       // Destroy the old instance
       if (_outputFilePlayerPtr)
@@ -3563,7 +3561,7 @@ int Channel::StartPlayingFileLocally(InStream* stream,
     // StartPlayingFileLocally(const char* ...) for more details.
     if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0)
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
         _outputFilePlaying = false;
         _engineStatisticsPtr->SetLastError(
             VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
@@ -3591,7 +3589,7 @@ int Channel::StopPlayingFileLocally()
     }
 
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
 
         if (_outputFilePlayerPtr->StopPlayingFile() != 0)
         {
@@ -3633,7 +3631,7 @@ int Channel::ScaleLocalFilePlayout(const float scale)
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::ScaleLocalFilePlayout(scale=%5.3f)", scale);
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     if (!_outputFilePlaying)
     {
@@ -3661,7 +3659,7 @@ int Channel::GetLocalPlayoutPosition(int& positionMs)
 
     WebRtc_UWord32 position;
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     if (_outputFilePlayerPtr == NULL)
     {
@@ -3705,7 +3703,7 @@ int Channel::StartPlayingFileAsMicrophone(const char* fileName,
         return 0;
     }
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     // Destroy the old instance
     if (_inputFilePlayerPtr)
@@ -3780,7 +3778,7 @@ int Channel::StartPlayingFileAsMicrophone(InStream* stream,
         return 0;
     }
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     // Destroy the old instance
     if (_inputFilePlayerPtr)
@@ -3836,7 +3834,7 @@ int Channel::StopPlayingFileAsMicrophone()
         return 0;
     }
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
     if (_inputFilePlayerPtr->StopPlayingFile() != 0)
     {
         _engineStatisticsPtr->SetLastError(
@@ -3865,7 +3863,7 @@ int Channel::ScaleFileAsMicrophonePlayout(const float scale)
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::ScaleFileAsMicrophonePlayout(scale=%5.3f)", scale);
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     if (!_inputFilePlaying)
     {
@@ -3927,7 +3925,7 @@ int Channel::StartRecordingPlayout(const char* fileName,
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     // Destroy the old instance
     if (_outputFileRecorderPtr)
@@ -4004,7 +4002,7 @@ int Channel::StartRecordingPlayout(OutStream* stream,
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     // Destroy the old instance
     if (_outputFileRecorderPtr)
@@ -4055,7 +4053,7 @@ int Channel::StopRecordingPlayout()
     }
 
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     if (_outputFileRecorderPtr->StopRecording() != 0)
     {
@@ -4171,7 +4169,7 @@ Channel::EnableSRTPSend(
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::EnableSRTPSend()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_encrypting)
     {
@@ -4241,7 +4239,7 @@ Channel::DisableSRTPSend()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::DisableSRTPSend()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (!_encrypting)
     {
@@ -4284,7 +4282,7 @@ Channel::EnableSRTPReceive(
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::EnableSRTPReceive()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_decrypting)
     {
@@ -4355,7 +4353,7 @@ Channel::DisableSRTPReceive()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::DisableSRTPReceive()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (!_decrypting)
     {
@@ -4391,7 +4389,7 @@ Channel::RegisterExternalEncryption(Encryption& encryption)
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::RegisterExternalEncryption()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_encryptionPtr)
     {
@@ -4415,7 +4413,7 @@ Channel::DeRegisterExternalEncryption()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                "Channel::DeRegisterExternalEncryption()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (!_encryptionPtr)
     {
@@ -4540,7 +4538,7 @@ Channel::RegisterTelephoneEventDetection(
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RegisterTelephoneEventDetection()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_telephoneEventDetectionPtr)
     {
@@ -4604,7 +4602,7 @@ Channel::DeRegisterTelephoneEventDetection()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::DeRegisterTelephoneEventDetection()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (!_telephoneEventDetectionPtr)
     {
@@ -4639,7 +4637,7 @@ Channel::GetTelephoneEventDetectionStatus(
                  "Channel::GetTelephoneEventDetectionStatus()");
 
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         enabled = (_telephoneEventDetectionPtr != NULL);
     }
 
@@ -4696,7 +4694,7 @@ Channel::RegisterRxVadObserver(VoERxVadCallback &observer)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RegisterRxVadObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_rxVadObserverPtr)
     {
@@ -4715,7 +4713,7 @@ Channel::DeRegisterRxVadObserver()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::DeRegisterRxVadObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (!_rxVadObserverPtr)
     {
@@ -4986,7 +4984,7 @@ Channel::RegisterRTPObserver(VoERTPObserver& observer)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::RegisterRTPObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_rtpObserverPtr)
     {
@@ -5007,7 +5005,7 @@ Channel::DeRegisterRTPObserver()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::DeRegisterRTPObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (!_rtpObserverPtr)
     {
@@ -5028,7 +5026,7 @@ Channel::RegisterRTCPObserver(VoERTCPObserver& observer)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RegisterRTCPObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_rtcpObserverPtr)
     {
@@ -5049,7 +5047,7 @@ Channel::DeRegisterRTCPObserver()
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::DeRegisterRTCPObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (!_rtcpObserverPtr)
     {
@@ -5877,7 +5875,7 @@ Channel::PrepareEncodeAndSend(int mixingFrequency)
 
     if (_inputExternalMedia)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         const bool isStereo = (_audioFrame._audioChannel == 2);
         if (_inputExternalMediaCallbackPtr)
         {
@@ -5973,7 +5971,7 @@ int Channel::RegisterExternalMediaProcessing(
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::RegisterExternalMediaProcessing()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (kPlaybackPerChannel == type)
     {
@@ -6009,7 +6007,7 @@ int Channel::DeRegisterExternalMediaProcessing(ProcessingTypes type)
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::DeRegisterExternalMediaProcessing()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (kPlaybackPerChannel == type)
     {
@@ -6223,7 +6221,7 @@ Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
     WebRtc_UWord32 fileSamples(0);
 
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
 
         if (_inputFilePlayerPtr == NULL)
         {
@@ -6287,7 +6285,7 @@ Channel::MixAudioWithFile(AudioFrame& audioFrame,
     WebRtc_UWord32 fileSamples(0);
 
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
 
         if (_outputFilePlayerPtr == NULL)
         {
diff --git a/src/voice_engine/main/source/channel.h b/src/voice_engine/main/source/channel.h
index e26413e610..7ceca4af6b 100644
--- a/src/voice_engine/main/source/channel.h
+++ b/src/voice_engine/main/source/channel.h
@@ -476,7 +476,7 @@ public:
         // A lock is needed because |_sending| is accessed by both
         // TransmitMixer::PrepareDemux() and StartSend()/StopSend(), which
         // are called by different threads.
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         return _sending;
     }
     bool Receiving() const
@@ -534,7 +534,6 @@ private:
 private:
     CriticalSectionWrapper& _fileCritSect;
     CriticalSectionWrapper& _callbackCritSect;
-    CriticalSectionWrapper& _transmitCritSect;
     WebRtc_UWord32 _instanceId;
     WebRtc_Word32 _channelId;
 
diff --git a/src/voice_engine/main/source/channel_manager_base.cc b/src/voice_engine/main/source/channel_manager_base.cc
index ea9938f078..572720c9c8 100644
--- a/src/voice_engine/main/source/channel_manager_base.cc
+++ b/src/voice_engine/main/source/channel_manager_base.cc
@@ -46,7 +46,7 @@ ChannelManagerBase::~ChannelManagerBase()
 
 bool ChannelManagerBase::GetFreeItemId(WebRtc_Word32& itemId)
 {
-    CriticalSectionScoped cs(*_itemsCritSectPtr);
+    CriticalSectionScoped cs(_itemsCritSectPtr);
     WebRtc_Word32 i(0);
     while (i < KMaxNumberOfItems)
     {
@@ -100,7 +100,7 @@ bool ChannelManagerBase::CreateItem(WebRtc_Word32& itemId)
 
 void ChannelManagerBase::InsertItem(WebRtc_Word32 itemId, void* item)
 {
-    CriticalSectionScoped cs(*_itemsCritSectPtr);
+    CriticalSectionScoped cs(_itemsCritSectPtr);
     assert(!_items.Find(itemId));
     _items.Insert(itemId, item);
 }
@@ -108,7 +108,7 @@ void ChannelManagerBase::InsertItem(WebRtc_Word32 itemId, void* item)
 void*
 ChannelManagerBase::RemoveItem(WebRtc_Word32 itemId)
 {
-    CriticalSectionScoped cs(*_itemsCritSectPtr);
+    CriticalSectionScoped cs(_itemsCritSectPtr);
     WriteLockScoped wlock(*_itemsRWLockPtr);
     MapItem* it = _items.Find(itemId);
     if (!it)
@@ -124,7 +124,7 @@ ChannelManagerBase::RemoveItem(WebRtc_Word32 itemId)
 
 void ChannelManagerBase::DestroyAllItems()
 {
-    CriticalSectionScoped cs(*_itemsCritSectPtr);
+    CriticalSectionScoped cs(_itemsCritSectPtr);
     MapItem* it = _items.First();
     while (it)
     {
@@ -148,7 +148,7 @@ WebRtc_Word32 ChannelManagerBase::MaxNumOfItems() const
 void*
 ChannelManagerBase::GetItem(WebRtc_Word32 itemId) const
 {
-    CriticalSectionScoped cs(*_itemsCritSectPtr);
+    CriticalSectionScoped cs(_itemsCritSectPtr);
     MapItem* it = _items.Find(itemId);
     if (!it)
     {
@@ -161,7 +161,7 @@ ChannelManagerBase::GetItem(WebRtc_Word32 itemId) const
 void*
 ChannelManagerBase::GetFirstItem(void*& iterator) const
 {
-    CriticalSectionScoped cs(*_itemsCritSectPtr);
+    CriticalSectionScoped cs(_itemsCritSectPtr);
     MapItem* it = _items.First();
     iterator = (void*) it;
     if (!it)
@@ -174,7 +174,7 @@ ChannelManagerBase::GetFirstItem(void*& iterator) const
 void*
 ChannelManagerBase::GetNextItem(void*& iterator) const
 {
-    CriticalSectionScoped cs(*_itemsCritSectPtr);
+    CriticalSectionScoped cs(_itemsCritSectPtr);
     MapItem* it = (MapItem*) iterator;
     if (!it)
     {
@@ -210,7 +210,7 @@ void ChannelManagerBase::GetItemIds(WebRtc_Word32* channelsArray,
 
 void ChannelManagerBase::GetChannels(MapWrapper& channels) const
 {
-    CriticalSectionScoped cs(*_itemsCritSectPtr);
+    CriticalSectionScoped cs(_itemsCritSectPtr);
     if (_items.Size() == 0)
     {
         return;
diff --git a/src/voice_engine/main/source/dtmf_inband.cc b/src/voice_engine/main/source/dtmf_inband.cc
index 473af1038b..689bc543d7 100644
--- a/src/voice_engine/main/source/dtmf_inband.cc
+++ b/src/voice_engine/main/source/dtmf_inband.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -129,7 +129,7 @@ DtmfInband::AddTone(const WebRtc_UWord8 eventCode,
                     WebRtc_Word32 lengthMs,
                     WebRtc_Word32 attenuationDb)
 {
-    CriticalSectionScoped lock(_critSect);
+    CriticalSectionScoped lock(&_critSect);
 
     if (attenuationDb > 36 || eventCode > 15)
     {
@@ -158,7 +158,7 @@ DtmfInband::AddTone(const WebRtc_UWord8 eventCode,
 int
 DtmfInband::ResetTone()
 {
-    CriticalSectionScoped lock(_critSect);
+    CriticalSectionScoped lock(&_critSect);
 
     ReInit();
 
@@ -173,7 +173,7 @@ int
 DtmfInband::StartTone(const WebRtc_UWord8 eventCode,
                       WebRtc_Word32 attenuationDb)
 {
-    CriticalSectionScoped lock(_critSect);
+    CriticalSectionScoped lock(&_critSect);
 
     if (attenuationDb > 36 || eventCode > 15)
     {
@@ -199,7 +199,7 @@ DtmfInband::StartTone(const WebRtc_UWord8 eventCode,
 int
 DtmfInband::StopTone()
 {
-    CriticalSectionScoped lock(_critSect);
+    CriticalSectionScoped lock(&_critSect);
 
     if (!_playing)
     {
@@ -221,7 +221,7 @@ DtmfInband::ReInit()
 bool 
 DtmfInband::IsAddingTone()
 {
-    CriticalSectionScoped lock(_critSect);
+    CriticalSectionScoped lock(&_critSect);
     return (_remainingSamples > 0 || _playing);
 }
 
@@ -229,7 +229,7 @@ int
 DtmfInband::Get10msTone(WebRtc_Word16 output[320],
                         WebRtc_UWord16& outputSizeInSamples)
 {
-    CriticalSectionScoped lock(_critSect);
+    CriticalSectionScoped lock(&_critSect);
     if (DtmfFix_generate(output,
                          _eventCode,
                          _attenuationDb,
diff --git a/src/voice_engine/main/source/dtmf_inband_queue.cc b/src/voice_engine/main/source/dtmf_inband_queue.cc
index 080ef3ec9e..b81d8273cf 100644
--- a/src/voice_engine/main/source/dtmf_inband_queue.cc
+++ b/src/voice_engine/main/source/dtmf_inband_queue.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -33,7 +33,7 @@ DtmfInbandQueue::AddDtmf(WebRtc_UWord8 key,
                          WebRtc_UWord16 len,
                          WebRtc_UWord8 level)
 {
-    CriticalSectionScoped lock(_DtmfCritsect);
+    CriticalSectionScoped lock(&_DtmfCritsect);
 
     if (_nextEmptyIndex >= kDtmfInbandMax)
     {
@@ -52,7 +52,7 @@ DtmfInbandQueue::AddDtmf(WebRtc_UWord8 key,
 WebRtc_Word8
 DtmfInbandQueue::NextDtmf(WebRtc_UWord16* len, WebRtc_UWord8* level)
 {
-    CriticalSectionScoped lock(_DtmfCritsect);
+    CriticalSectionScoped lock(&_DtmfCritsect);
 
     if(!PendingDtmf())
     {
diff --git a/src/voice_engine/main/source/monitor_module.cc b/src/voice_engine/main/source/monitor_module.cc
index 4cb98a1f38..07b17fb1a4 100644
--- a/src/voice_engine/main/source/monitor_module.cc
+++ b/src/voice_engine/main/source/monitor_module.cc
@@ -30,7 +30,7 @@ MonitorModule::~MonitorModule()
 WebRtc_Word32 
 MonitorModule::RegisterObserver(MonitorObserver& observer)
 {
-    CriticalSectionScoped lock(_callbackCritSect);
+    CriticalSectionScoped lock(&_callbackCritSect);
     if (_observerPtr)
     {
         return -1;
@@ -42,7 +42,7 @@ MonitorModule::RegisterObserver(MonitorObserver& observer)
 WebRtc_Word32 
 MonitorModule::DeRegisterObserver()
 {
-    CriticalSectionScoped lock(_callbackCritSect);
+    CriticalSectionScoped lock(&_callbackCritSect);
     if (!_observerPtr)
     {
         return 0;
@@ -80,7 +80,7 @@ MonitorModule::Process()
     _lastProcessTime = GET_TIME_IN_MS();
     if (_observerPtr)
     {
-        CriticalSectionScoped lock(_callbackCritSect);
+        CriticalSectionScoped lock(&_callbackCritSect);
         _observerPtr->OnPeriodicProcess();
     }
     return 0;
diff --git a/src/voice_engine/main/source/output_mixer.cc b/src/voice_engine/main/source/output_mixer.cc
index 9dfe0ad428..2764f78eae 100644
--- a/src/voice_engine/main/source/output_mixer.cc
+++ b/src/voice_engine/main/source/output_mixer.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -94,7 +94,7 @@ void OutputMixer::RecordFileEnded(const WebRtc_Word32 id)
                  "OutputMixer::RecordFileEnded(id=%d)", id);
     assert(id == _instanceId);
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
     _outputFileRecording = false;
     WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::RecordFileEnded() =>"
@@ -165,7 +165,7 @@ OutputMixer::~OutputMixer()
         DeRegisterExternalMediaProcessing();
     }
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
         if (_outputFileRecorderPtr)
         {
             _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
@@ -207,7 +207,7 @@ int OutputMixer::RegisterExternalMediaProcessing(
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
                "OutputMixer::RegisterExternalMediaProcessing()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
     _externalMediaCallbackPtr = &proccess_object;
     _externalMedia = true;
 
@@ -219,7 +219,7 @@ int OutputMixer::DeRegisterExternalMediaProcessing()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
                  "OutputMixer::DeRegisterExternalMediaProcessing()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
     _externalMedia = false;
     _externalMediaCallbackPtr = NULL;
 
@@ -365,7 +365,7 @@ int OutputMixer::StartRecordingPlayout(const char* fileName,
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
     
     // Destroy the old instance
     if (_outputFileRecorderPtr)
@@ -445,7 +445,7 @@ int OutputMixer::StartRecordingPlayout(OutStream* stream,
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     // Destroy the old instance
     if (_outputFileRecorderPtr)
@@ -496,7 +496,7 @@ int OutputMixer::StopRecordingPlayout()
         return -1;
     }
 
-    CriticalSectionScoped cs(_fileCritSect);
+    CriticalSectionScoped cs(&_fileCritSect);
 
     if (_outputFileRecorderPtr->StopRecording() != 0)
     {
@@ -526,7 +526,7 @@ OutputMixer::GetMixedAudio(const WebRtc_Word32 desiredFreqHz,
 
     // --- Record playout if enabled
     {
-        CriticalSectionScoped cs(_fileCritSect);
+        CriticalSectionScoped cs(&_fileCritSect);
         if (_outputFileRecording)
         {
             assert(audioFrame._audioChannel == 1);
@@ -632,7 +632,7 @@ OutputMixer::DoOperationsOnCombinedSignal()
 
     if (_externalMedia)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         const bool isStereo = (_audioFrame._audioChannel == 2);
         if (_externalMediaCallbackPtr)
         {
diff --git a/src/voice_engine/main/source/ref_count.cc b/src/voice_engine/main/source/ref_count.cc
index f1ed0be996..9723bc3ca7 100644
--- a/src/voice_engine/main/source/ref_count.cc
+++ b/src/voice_engine/main/source/ref_count.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -29,7 +29,7 @@ RefCount::~RefCount()
 RefCount&
 RefCount::operator++(int)
 {
-    CriticalSectionScoped lock(_crit);
+    CriticalSectionScoped lock(&_crit);
     _count++;
     return *this;
 }
@@ -37,7 +37,7 @@ RefCount::operator++(int)
 RefCount&
 RefCount::operator--(int)
 {
-    CriticalSectionScoped lock(_crit);
+    CriticalSectionScoped lock(&_crit);
     _count--;
     return *this;
 }
@@ -45,7 +45,7 @@ RefCount::operator--(int)
 void 
 RefCount::Reset()
 {
-    CriticalSectionScoped lock(_crit);
+    CriticalSectionScoped lock(&_crit);
     _count = 0;
 }
 
diff --git a/src/voice_engine/main/source/statistics.cc b/src/voice_engine/main/source/statistics.cc
index a5340300c4..4f1bc7915f 100644
--- a/src/voice_engine/main/source/statistics.cc
+++ b/src/voice_engine/main/source/statistics.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -56,7 +56,7 @@ bool Statistics::Initialized() const
 
 WebRtc_Word32 Statistics::SetLastError(const WebRtc_Word32 error) const
 {
-    CriticalSectionScoped cs(*_critPtr);
+    CriticalSectionScoped cs(_critPtr);
     _lastError = error;
     return 0;
 }
@@ -64,7 +64,7 @@ WebRtc_Word32 Statistics::SetLastError(const WebRtc_Word32 error) const
 WebRtc_Word32 Statistics::SetLastError(const WebRtc_Word32 error,
                                        const TraceLevel level) const
 {
-    CriticalSectionScoped cs(*_critPtr);
+    CriticalSectionScoped cs(_critPtr);
     _lastError = error;
     WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1),
                  "error code is set to %d",
@@ -76,7 +76,7 @@ WebRtc_Word32 Statistics::SetLastError(
     const WebRtc_Word32 error,
     const TraceLevel level, const char* msg) const
 {
-    CriticalSectionScoped cs(*_critPtr);
+    CriticalSectionScoped cs(_critPtr);
     char traceMessage[KTraceMaxMessageSize];
     assert(strlen(msg) < KTraceMaxMessageSize);
     _lastError = error;
@@ -88,7 +88,7 @@ WebRtc_Word32 Statistics::SetLastError(
 
 WebRtc_Word32 Statistics::LastError() const
 {
-    CriticalSectionScoped cs(*_critPtr);
+    CriticalSectionScoped cs(_critPtr);
     WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
                "LastError() => %d", _lastError);
     return _lastError;
diff --git a/src/voice_engine/main/source/transmit_mixer.cc b/src/voice_engine/main/source/transmit_mixer.cc
index 5c30ab1417..864c170787 100644
--- a/src/voice_engine/main/source/transmit_mixer.cc
+++ b/src/voice_engine/main/source/transmit_mixer.cc
@@ -36,7 +36,7 @@ TransmitMixer::OnPeriodicProcess()
 #if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION)
     if (_typingNoiseWarning > 0)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         if (_voiceEngineObserverPtr)
         {
             WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
@@ -51,7 +51,7 @@ TransmitMixer::OnPeriodicProcess()
 
     if (_saturationWarning > 0)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         if (_voiceEngineObserverPtr)
         {
             WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
@@ -64,7 +64,7 @@ TransmitMixer::OnPeriodicProcess()
 
     if (_noiseWarning > 0)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         if (_voiceEngineObserverPtr)
         {
             WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
@@ -104,7 +104,7 @@ void TransmitMixer::PlayFileEnded(const WebRtc_Word32 id)
 
     assert(id == _filePlayerId);
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     _filePlaying = false;
     WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
@@ -120,14 +120,14 @@ TransmitMixer::RecordFileEnded(const WebRtc_Word32 id)
 
     if (id == _fileRecorderId)
     {
-        CriticalSectionScoped cs(_critSect);
+        CriticalSectionScoped cs(&_critSect);
         _fileRecording = false;
         WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
                      "TransmitMixer::RecordFileEnded() => fileRecorder module"
                      "is shutdown");
     } else if (id == _fileCallRecorderId)
     {
-        CriticalSectionScoped cs(_critSect);
+        CriticalSectionScoped cs(&_critSect);
         _fileCallRecording = false;
         WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
                      "TransmitMixer::RecordFileEnded() => fileCallRecorder"
@@ -216,7 +216,7 @@ TransmitMixer::~TransmitMixer()
         DeRegisterExternalMediaProcessing();
     }
     {
-        CriticalSectionScoped cs(_critSect);
+        CriticalSectionScoped cs(&_critSect);
         if (_fileRecorderPtr)
         {
             _fileRecorderPtr->RegisterModuleFileCallback(NULL);
@@ -273,7 +273,7 @@ TransmitMixer::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::RegisterVoiceEngineObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_voiceEngineObserverPtr)
     {
@@ -394,7 +394,7 @@ TransmitMixer::PrepareDemux(const WebRtc_Word8* audioSamples,
 
     if (_externalMedia)
     {
-        CriticalSectionScoped cs(_callbackCritSect);
+        CriticalSectionScoped cs(&_callbackCritSect);
         const bool isStereo = (_audioFrame._audioChannel == 2);
         if (_externalMediaCallbackPtr)
         {
@@ -514,7 +514,7 @@ int TransmitMixer::StartPlayingFileAsMicrophone(const char* fileName,
         return 0;
     }
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     // Destroy the old instance
     if (_filePlayerPtr)
@@ -591,7 +591,7 @@ int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream,
         return 0;
     }
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     // Destroy the old instance
     if (_filePlayerPtr)
@@ -651,7 +651,7 @@ int TransmitMixer::StopPlayingFileAsMicrophone()
         return 0;
     }
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     if (_filePlayerPtr->StopPlayingFile() != 0)
     {
@@ -682,7 +682,7 @@ int TransmitMixer::ScaleFileAsMicrophonePlayout(const float scale)
                  "TransmitMixer::ScaleFileAsMicrophonePlayout(scale=%5.3f)",
                  scale);
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     if (!_filePlaying)
     {
@@ -744,7 +744,7 @@ int TransmitMixer::StartRecordingMicrophone(const char* fileName,
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     // Destroy the old instance
     if (_fileRecorderPtr)
@@ -822,7 +822,7 @@ int TransmitMixer::StartRecordingMicrophone(OutStream* stream,
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     // Destroy the old instance
     if (_fileRecorderPtr)
@@ -874,7 +874,7 @@ int TransmitMixer::StopRecordingMicrophone()
         return -1;
     }
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     if (_fileRecorderPtr->StopRecording() != 0)
     {
@@ -929,7 +929,7 @@ int TransmitMixer::StartRecordingCall(const char* fileName,
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     // Destroy the old instance
     if (_fileCallRecorderPtr)
@@ -1007,7 +1007,7 @@ int TransmitMixer::StartRecordingCall(OutStream* stream,
         format = kFileFormatCompressedFile;
     }
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     // Destroy the old instance
     if (_fileCallRecorderPtr)
@@ -1058,7 +1058,7 @@ int TransmitMixer::StopRecordingCall()
         return -1;
     }
 
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
 
     if (_fileCallRecorderPtr->StopRecording() != 0)
     {
@@ -1088,7 +1088,7 @@ int TransmitMixer::RegisterExternalMediaProcessing(
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::RegisterExternalMediaProcessing()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
     _externalMediaCallbackPtr = &proccess_object;
     _externalMedia = true;
 
@@ -1100,7 +1100,7 @@ int TransmitMixer::DeRegisterExternalMediaProcessing()
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
                  "TransmitMixer::DeRegisterExternalMediaProcessing()");
 
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
     _externalMedia = false;
     _externalMediaCallbackPtr = NULL;
 
@@ -1195,7 +1195,7 @@ TransmitMixer::GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
 WebRtc_Word32 TransmitMixer::RecordAudioToFile(
     const WebRtc_UWord32 mixingFrequency)
 {
-    CriticalSectionScoped cs(_critSect);
+    CriticalSectionScoped cs(&_critSect);
     if (_fileRecorderPtr == NULL)
     {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
@@ -1223,7 +1223,7 @@ WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
     WebRtc_UWord32 fileSamples(0);
 
     {
-        CriticalSectionScoped cs(_critSect);
+        CriticalSectionScoped cs(&_critSect);
         if (_filePlayerPtr == NULL)
         {
             WEBRTC_TRACE(kTraceWarning, kTraceVoice,
diff --git a/src/voice_engine/main/source/voe_base_impl.cc b/src/voice_engine/main/source/voe_base_impl.cc
index cdc0ee3fe1..934b50280f 100644
--- a/src/voice_engine/main/source/voe_base_impl.cc
+++ b/src/voice_engine/main/source/voe_base_impl.cc
@@ -84,7 +84,7 @@ int VoEBaseImpl::Release()
 
 void VoEBaseImpl::OnErrorIsReported(const ErrorCode error)
 {
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
     if (_voiceEngineObserver)
     {
         if (_voiceEngineObserverPtr)
@@ -112,7 +112,7 @@ void VoEBaseImpl::OnErrorIsReported(const ErrorCode error)
 
 void VoEBaseImpl::OnWarningIsReported(const WarningCode warning)
 {
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
     if (_voiceEngineObserver)
     {
         if (_voiceEngineObserverPtr)
@@ -289,7 +289,7 @@ int VoEBaseImpl::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "RegisterVoiceEngineObserver(observer=0x%d)", &observer);
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
     if (_voiceEngineObserverPtr)
     {
         _engineStatistics.SetLastError(VE_INVALID_OPERATION, kTraceError,
@@ -319,7 +319,7 @@ int VoEBaseImpl::DeRegisterVoiceEngineObserver()
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "DeRegisterVoiceEngineObserver()");
-    CriticalSectionScoped cs(_callbackCritSect);
+    CriticalSectionScoped cs(&_callbackCritSect);
     if (!_voiceEngineObserverPtr)
     {
         _engineStatistics.SetLastError(VE_INVALID_OPERATION, kTraceError,
@@ -347,7 +347,7 @@ int VoEBaseImpl::Init(AudioDeviceModule* external_adm)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), 
         "Init(external_adm=0x%p)", external_adm);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
 
     if (_engineStatistics.Initialized())
     {
@@ -657,7 +657,7 @@ int VoEBaseImpl::Terminate()
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "Terminate()");
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
     return TerminateInternal();
 }
 
@@ -675,7 +675,7 @@ int VoEBaseImpl::CreateChannel()
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "CreateChannel()");
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
 
     if (!_engineStatistics.Initialized())
     {
@@ -741,7 +741,7 @@ int VoEBaseImpl::DeleteChannel(int channel)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "DeleteChannel(channel=%d)", channel);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
 
     if (!_engineStatistics.Initialized())
     {
@@ -796,7 +796,7 @@ int VoEBaseImpl::SetLocalReceiver(int channel, int port, int RTCPport,
     //  SetSendDestination and StartSend without having called SetLocalReceiver
     // first. The sockets are then created at the first packet transmission.
 
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
     if (ipAddr == NULL && multiCastAddr == NULL)
     {
         WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
@@ -924,7 +924,7 @@ int VoEBaseImpl::SetSendDestination(int channel, int port, const char* ipaddr,
                  "SetSendDestination(channel=%d, port=%d, ipaddr=%s,"
                  "sourcePort=%d, RTCPport=%d)",
                  channel, port, ipaddr, sourcePort, RTCPport);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
 #ifndef WEBRTC_EXTERNAL_TRANSPORT
     if (!_engineStatistics.Initialized())
     {
@@ -1050,7 +1050,7 @@ int VoEBaseImpl::StartReceive(int channel)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "StartReceive(channel=%d)", channel);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
     if (!_engineStatistics.Initialized())
     {
         _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
@@ -1072,7 +1072,7 @@ int VoEBaseImpl::StopReceive(int channel)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "StopListen(channel=%d)", channel);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
     if (!_engineStatistics.Initialized())
     {
         _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
@@ -1094,7 +1094,7 @@ int VoEBaseImpl::StartPlayout(int channel)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "StartPlayout(channel=%d)", channel);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
     if (!_engineStatistics.Initialized())
     {
         _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
@@ -1128,7 +1128,7 @@ int VoEBaseImpl::StopPlayout(int channel)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "StopPlayout(channel=%d)", channel);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
     if (!_engineStatistics.Initialized())
     {
         _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
@@ -1156,7 +1156,7 @@ int VoEBaseImpl::StartSend(int channel)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "StartSend(channel=%d)", channel);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
     if (!_engineStatistics.Initialized())
     {
         _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
@@ -1198,7 +1198,7 @@ int VoEBaseImpl::StopSend(int channel)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "StopSend(channel=%d)", channel);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
     if (!_engineStatistics.Initialized())
     {
         _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
diff --git a/src/voice_engine/main/source/voe_dtmf_impl.cc b/src/voice_engine/main/source/voe_dtmf_impl.cc
index 67f4c4613f..a554da8980 100644
--- a/src/voice_engine/main/source/voe_dtmf_impl.cc
+++ b/src/voice_engine/main/source/voe_dtmf_impl.cc
@@ -1,5 +1,5 @@
 /*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *  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
@@ -395,7 +395,7 @@ int VoEDtmfImpl::SetDtmfFeedbackStatus(bool enable, bool directFeedback)
                  "SetDtmfFeedbackStatus(enable=%d, directFeeback=%d)",
                  (int)enable, (int)directFeedback);
 
-    CriticalSectionScoped sc(*_apiCritPtr);
+    CriticalSectionScoped sc(_apiCritPtr);
 
     _dtmfFeedback = enable;
     _dtmfDirectFeedback = directFeedback;
@@ -408,7 +408,7 @@ int VoEDtmfImpl::GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback)
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
                  "GetDtmfFeedbackStatus()");
 
-    CriticalSectionScoped sc(*_apiCritPtr);
+    CriticalSectionScoped sc(_apiCritPtr);
 
     enabled = _dtmfFeedback;
     directFeedback = _dtmfDirectFeedback;
diff --git a/src/voice_engine/main/source/voe_hardware_impl.cc b/src/voice_engine/main/source/voe_hardware_impl.cc
index 0ce440d7c8..3a86c6c264 100644
--- a/src/voice_engine/main/source/voe_hardware_impl.cc
+++ b/src/voice_engine/main/source/voe_hardware_impl.cc
@@ -341,7 +341,7 @@ int VoEHardwareImpl::SetRecordingDevice(int index,
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "SetRecordingDevice(index=%d, recordingChannel=%d)",
                  index, (int) recordingChannel);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
     ANDROID_NOT_SUPPORTED(_engineStatistics);
     IPHONE_NOT_SUPPORTED();
 
@@ -478,7 +478,7 @@ int VoEHardwareImpl::SetPlayoutDevice(int index)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
                  "SetPlayoutDevice(index=%d)", index);
-    CriticalSectionScoped cs(*_apiCritPtr);
+    CriticalSectionScoped cs(_apiCritPtr);
     ANDROID_NOT_SUPPORTED(_engineStatistics);
     IPHONE_NOT_SUPPORTED();