VoiceEngine now uses pointer constructor of CriticalSectionScoped, instead of reference. BUG=184 TEST=Compiles on all platforms. Review URL: https://webrtc-codereview.appspot.com/436001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1853 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/src/voice_engine/main/source/channel.cc b/src/voice_engine/main/source/channel.cc index fb24494..cd93137 100644 --- a/src/voice_engine/main/source/channel.cc +++ b/src/voice_engine/main/source/channel.cc
@@ -85,7 +85,7 @@ 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 @@ if (digitDtmf != 999) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_telephoneEventDetectionPtr) { _telephoneEventDetectionPtr->OnReceivedTelephoneEventInband( @@ -119,7 +119,7 @@ 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 @@ // SRTP or External encryption if (_encrypting) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_encryptionPtr) { @@ -224,7 +224,7 @@ // Packet transmission using external transport transport { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); int n = _transportPtr->SendPacket(channel, bufferToSendPtr, @@ -251,7 +251,7 @@ "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 @@ // SRTP or External encryption if (_encrypting) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_encryptionPtr) { @@ -327,7 +327,7 @@ // Packet transmission using external transport transport { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); int n = _transportPtr->SendRTCPPacket(channel, bufferToSendPtr, @@ -371,7 +371,7 @@ // SRTP or External decryption if (_decrypting) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_encryptionPtr) { @@ -451,7 +451,7 @@ // SRTP or External decryption if (_decrypting) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_encryptionPtr) { @@ -515,7 +515,7 @@ #ifdef WEBRTC_DTMF_DETECTION if (_outOfBandTelephoneEventDetecion) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_telephoneEventDetectionPtr) { @@ -567,7 +567,7 @@ if (_rtpObserver) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_rtpObserverPtr) { @@ -590,7 +590,7 @@ if (_rtpObserver) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_rtpObserverPtr) { @@ -616,7 +616,7 @@ if (_rtcpObserver) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_rtcpObserverPtr) { @@ -677,7 +677,7 @@ 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 @@ // 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 @@ // Send callback to the registered observer if (_connectionObserver) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_connectionObserverPtr) { _connectionObserverPtr->OnPeriodicDeadOrAlive(channel, isAlive); @@ -898,7 +898,7 @@ // External media if (_outputExternalMedia) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); const bool isStereo = (audioFrame._audioChannel == 2); if (_outputExternalMediaCallbackPtr) { @@ -914,7 +914,7 @@ // Record playout if enabled { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_outputFileRecording && _outputFileRecorderPtr) { @@ -968,7 +968,7 @@ // limit the spectrum anyway. if (_outputFilePlaying) { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_outputFilePlayerPtr && _outputFilePlaying) { if(_outputFilePlayerPtr->Frequency()>highestNeeded) @@ -1032,7 +1032,7 @@ if (id == _inputFilePlayerId) { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); _inputFilePlaying = false; WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, @@ -1042,7 +1042,7 @@ } else if (id == _outputFilePlayerId) { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); _outputFilePlaying = false; WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, @@ -1060,7 +1060,7 @@ assert(id == _outputFileRecorderId); - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); _outputFileRecording = false; WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, @@ -1073,7 +1073,6 @@ 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 @@ StopPlayout(); { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_inputFilePlayerPtr) { _inputFilePlayerPtr->RegisterModuleFileCallback(NULL); @@ -1352,7 +1351,6 @@ delete [] _encryptionRTCPBufferPtr; delete [] _decryptionRTCPBufferPtr; delete &_callbackCritSect; - delete &_transmitCritSect; delete &_fileCritSect; } @@ -1530,7 +1528,7 @@ { // 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 @@ { // 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 @@ _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 @@ { // 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 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::RegisterVoiceEngineObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_voiceEngineObserverPtr) { @@ -2270,7 +2268,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::DeRegisterVoiceEngineObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (!_voiceEngineObserverPtr) { @@ -2769,7 +2767,7 @@ 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 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::DeRegisterExternalTransport()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (!_transportPtr) { @@ -3252,7 +3250,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::RegisterDeadOrAliveObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_connectionObserverPtr) { @@ -3272,7 +3270,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::DeRegisterDeadOrAliveObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (!_connectionObserverPtr) { @@ -3425,7 +3423,7 @@ } { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_outputFilePlayerPtr) { @@ -3473,7 +3471,7 @@ // 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 @@ } { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); // Destroy the old instance if (_outputFilePlayerPtr) @@ -3563,7 +3561,7 @@ // 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 @@ } { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_outputFilePlayerPtr->StopPlayingFile() != 0) { @@ -3633,7 +3631,7 @@ 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 @@ WebRtc_UWord32 position; - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_outputFilePlayerPtr == NULL) { @@ -3705,7 +3703,7 @@ return 0; } - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); // Destroy the old instance if (_inputFilePlayerPtr) @@ -3780,7 +3778,7 @@ return 0; } - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); // Destroy the old instance if (_inputFilePlayerPtr) @@ -3836,7 +3834,7 @@ return 0; } - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_inputFilePlayerPtr->StopPlayingFile() != 0) { _engineStatisticsPtr->SetLastError( @@ -3865,7 +3863,7 @@ 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 @@ format = kFileFormatCompressedFile; } - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); // Destroy the old instance if (_outputFileRecorderPtr) @@ -4004,7 +4002,7 @@ format = kFileFormatCompressedFile; } - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); // Destroy the old instance if (_outputFileRecorderPtr) @@ -4055,7 +4053,7 @@ } - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_outputFileRecorderPtr->StopRecording() != 0) { @@ -4171,7 +4169,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::EnableSRTPSend()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_encrypting) { @@ -4241,7 +4239,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::DisableSRTPSend()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (!_encrypting) { @@ -4284,7 +4282,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::EnableSRTPReceive()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_decrypting) { @@ -4355,7 +4353,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::DisableSRTPReceive()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (!_decrypting) { @@ -4391,7 +4389,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::RegisterExternalEncryption()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_encryptionPtr) { @@ -4415,7 +4413,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::DeRegisterExternalEncryption()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (!_encryptionPtr) { @@ -4540,7 +4538,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::RegisterTelephoneEventDetection()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_telephoneEventDetectionPtr) { @@ -4604,7 +4602,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), "Channel::DeRegisterTelephoneEventDetection()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (!_telephoneEventDetectionPtr) { @@ -4639,7 +4637,7 @@ "Channel::GetTelephoneEventDetectionStatus()"); { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); enabled = (_telephoneEventDetectionPtr != NULL); } @@ -4696,7 +4694,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::RegisterRxVadObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_rxVadObserverPtr) { @@ -4715,7 +4713,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::DeRegisterRxVadObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (!_rxVadObserverPtr) { @@ -4986,7 +4984,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), "Channel::RegisterRTPObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_rtpObserverPtr) { @@ -5007,7 +5005,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::DeRegisterRTPObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (!_rtpObserverPtr) { @@ -5028,7 +5026,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::RegisterRTCPObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_rtcpObserverPtr) { @@ -5049,7 +5047,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), "Channel::DeRegisterRTCPObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (!_rtcpObserverPtr) { @@ -5877,7 +5875,7 @@ if (_inputExternalMedia) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); const bool isStereo = (_audioFrame._audioChannel == 2); if (_inputExternalMediaCallbackPtr) { @@ -5973,7 +5971,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::RegisterExternalMediaProcessing()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (kPlaybackPerChannel == type) { @@ -6009,7 +6007,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), "Channel::DeRegisterExternalMediaProcessing()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (kPlaybackPerChannel == type) { @@ -6223,7 +6221,7 @@ WebRtc_UWord32 fileSamples(0); { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_inputFilePlayerPtr == NULL) { @@ -6287,7 +6285,7 @@ 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 e26413e..7ceca4a 100644 --- a/src/voice_engine/main/source/channel.h +++ b/src/voice_engine/main/source/channel.h
@@ -476,7 +476,7 @@ // 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: 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 ea9938f..572720c 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 @@ bool ChannelManagerBase::GetFreeItemId(WebRtc_Word32& itemId) { - CriticalSectionScoped cs(*_itemsCritSectPtr); + CriticalSectionScoped cs(_itemsCritSectPtr); WebRtc_Word32 i(0); while (i < KMaxNumberOfItems) { @@ -100,7 +100,7 @@ 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::RemoveItem(WebRtc_Word32 itemId) { - CriticalSectionScoped cs(*_itemsCritSectPtr); + CriticalSectionScoped cs(_itemsCritSectPtr); WriteLockScoped wlock(*_itemsRWLockPtr); MapItem* it = _items.Find(itemId); if (!it) @@ -124,7 +124,7 @@ void ChannelManagerBase::DestroyAllItems() { - CriticalSectionScoped cs(*_itemsCritSectPtr); + CriticalSectionScoped cs(_itemsCritSectPtr); MapItem* it = _items.First(); while (it) { @@ -148,7 +148,7 @@ void* ChannelManagerBase::GetItem(WebRtc_Word32 itemId) const { - CriticalSectionScoped cs(*_itemsCritSectPtr); + CriticalSectionScoped cs(_itemsCritSectPtr); MapItem* it = _items.Find(itemId); if (!it) { @@ -161,7 +161,7 @@ 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 @@ void* ChannelManagerBase::GetNextItem(void*& iterator) const { - CriticalSectionScoped cs(*_itemsCritSectPtr); + CriticalSectionScoped cs(_itemsCritSectPtr); MapItem* it = (MapItem*) iterator; if (!it) { @@ -210,7 +210,7 @@ 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 473af10..689bc54 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 @@ WebRtc_Word32 lengthMs, WebRtc_Word32 attenuationDb) { - CriticalSectionScoped lock(_critSect); + CriticalSectionScoped lock(&_critSect); if (attenuationDb > 36 || eventCode > 15) { @@ -158,7 +158,7 @@ int DtmfInband::ResetTone() { - CriticalSectionScoped lock(_critSect); + CriticalSectionScoped lock(&_critSect); ReInit(); @@ -173,7 +173,7 @@ DtmfInband::StartTone(const WebRtc_UWord8 eventCode, WebRtc_Word32 attenuationDb) { - CriticalSectionScoped lock(_critSect); + CriticalSectionScoped lock(&_critSect); if (attenuationDb > 36 || eventCode > 15) { @@ -199,7 +199,7 @@ int DtmfInband::StopTone() { - CriticalSectionScoped lock(_critSect); + CriticalSectionScoped lock(&_critSect); if (!_playing) { @@ -221,7 +221,7 @@ bool DtmfInband::IsAddingTone() { - CriticalSectionScoped lock(_critSect); + CriticalSectionScoped lock(&_critSect); return (_remainingSamples > 0 || _playing); } @@ -229,7 +229,7 @@ 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 080ef3e..b81d827 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 @@ WebRtc_UWord16 len, WebRtc_UWord8 level) { - CriticalSectionScoped lock(_DtmfCritsect); + CriticalSectionScoped lock(&_DtmfCritsect); if (_nextEmptyIndex >= kDtmfInbandMax) { @@ -52,7 +52,7 @@ 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 4cb98a1..07b17fb 100644 --- a/src/voice_engine/main/source/monitor_module.cc +++ b/src/voice_engine/main/source/monitor_module.cc
@@ -30,7 +30,7 @@ WebRtc_Word32 MonitorModule::RegisterObserver(MonitorObserver& observer) { - CriticalSectionScoped lock(_callbackCritSect); + CriticalSectionScoped lock(&_callbackCritSect); if (_observerPtr) { return -1; @@ -42,7 +42,7 @@ WebRtc_Word32 MonitorModule::DeRegisterObserver() { - CriticalSectionScoped lock(_callbackCritSect); + CriticalSectionScoped lock(&_callbackCritSect); if (!_observerPtr) { return 0; @@ -80,7 +80,7 @@ _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 9dfe0ad..2764f78 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 @@ "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 @@ DeRegisterExternalMediaProcessing(); } { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_outputFileRecorderPtr) { _outputFileRecorderPtr->RegisterModuleFileCallback(NULL); @@ -207,7 +207,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), "OutputMixer::RegisterExternalMediaProcessing()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); _externalMediaCallbackPtr = &proccess_object; _externalMedia = true; @@ -219,7 +219,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1), "OutputMixer::DeRegisterExternalMediaProcessing()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); _externalMedia = false; _externalMediaCallbackPtr = NULL; @@ -365,7 +365,7 @@ format = kFileFormatCompressedFile; } - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); // Destroy the old instance if (_outputFileRecorderPtr) @@ -445,7 +445,7 @@ format = kFileFormatCompressedFile; } - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); // Destroy the old instance if (_outputFileRecorderPtr) @@ -496,7 +496,7 @@ return -1; } - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_outputFileRecorderPtr->StopRecording() != 0) { @@ -526,7 +526,7 @@ // --- Record playout if enabled { - CriticalSectionScoped cs(_fileCritSect); + CriticalSectionScoped cs(&_fileCritSect); if (_outputFileRecording) { assert(audioFrame._audioChannel == 1); @@ -632,7 +632,7 @@ 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 f1ed0be..9723bc3 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::operator++(int) { - CriticalSectionScoped lock(_crit); + CriticalSectionScoped lock(&_crit); _count++; return *this; } @@ -37,7 +37,7 @@ RefCount& RefCount::operator--(int) { - CriticalSectionScoped lock(_crit); + CriticalSectionScoped lock(&_crit); _count--; return *this; } @@ -45,7 +45,7 @@ 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 a534030..4f1bc79 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 @@ 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 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 @@ 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::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 5c30ab1..864c170 100644 --- a/src/voice_engine/main/source/transmit_mixer.cc +++ b/src/voice_engine/main/source/transmit_mixer.cc
@@ -36,7 +36,7 @@ #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 @@ if (_saturationWarning > 0) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_voiceEngineObserverPtr) { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), @@ -64,7 +64,7 @@ if (_noiseWarning > 0) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_voiceEngineObserverPtr) { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), @@ -104,7 +104,7 @@ assert(id == _filePlayerId); - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); _filePlaying = false; WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1), @@ -120,14 +120,14 @@ 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 @@ DeRegisterExternalMediaProcessing(); } { - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); if (_fileRecorderPtr) { _fileRecorderPtr->RegisterModuleFileCallback(NULL); @@ -273,7 +273,7 @@ { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), "TransmitMixer::RegisterVoiceEngineObserver()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_voiceEngineObserverPtr) { @@ -394,7 +394,7 @@ if (_externalMedia) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); const bool isStereo = (_audioFrame._audioChannel == 2); if (_externalMediaCallbackPtr) { @@ -514,7 +514,7 @@ return 0; } - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); // Destroy the old instance if (_filePlayerPtr) @@ -591,7 +591,7 @@ return 0; } - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); // Destroy the old instance if (_filePlayerPtr) @@ -651,7 +651,7 @@ return 0; } - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); if (_filePlayerPtr->StopPlayingFile() != 0) { @@ -682,7 +682,7 @@ "TransmitMixer::ScaleFileAsMicrophonePlayout(scale=%5.3f)", scale); - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); if (!_filePlaying) { @@ -744,7 +744,7 @@ format = kFileFormatCompressedFile; } - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); // Destroy the old instance if (_fileRecorderPtr) @@ -822,7 +822,7 @@ format = kFileFormatCompressedFile; } - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); // Destroy the old instance if (_fileRecorderPtr) @@ -874,7 +874,7 @@ return -1; } - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); if (_fileRecorderPtr->StopRecording() != 0) { @@ -929,7 +929,7 @@ format = kFileFormatCompressedFile; } - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); // Destroy the old instance if (_fileCallRecorderPtr) @@ -1007,7 +1007,7 @@ format = kFileFormatCompressedFile; } - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); // Destroy the old instance if (_fileCallRecorderPtr) @@ -1058,7 +1058,7 @@ return -1; } - CriticalSectionScoped cs(_critSect); + CriticalSectionScoped cs(&_critSect); if (_fileCallRecorderPtr->StopRecording() != 0) { @@ -1088,7 +1088,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), "TransmitMixer::RegisterExternalMediaProcessing()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); _externalMediaCallbackPtr = &proccess_object; _externalMedia = true; @@ -1100,7 +1100,7 @@ WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1), "TransmitMixer::DeRegisterExternalMediaProcessing()"); - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); _externalMedia = false; _externalMediaCallbackPtr = NULL; @@ -1195,7 +1195,7 @@ 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_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 cdc0ee3..934b502 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 @@ void VoEBaseImpl::OnErrorIsReported(const ErrorCode error) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_voiceEngineObserver) { if (_voiceEngineObserverPtr) @@ -112,7 +112,7 @@ void VoEBaseImpl::OnWarningIsReported(const WarningCode warning) { - CriticalSectionScoped cs(_callbackCritSect); + CriticalSectionScoped cs(&_callbackCritSect); if (_voiceEngineObserver) { if (_voiceEngineObserverPtr) @@ -289,7 +289,7 @@ { 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 @@ { 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 @@ { 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 @@ { WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), "Terminate()"); - CriticalSectionScoped cs(*_apiCritPtr); + CriticalSectionScoped cs(_apiCritPtr); return TerminateInternal(); } @@ -675,7 +675,7 @@ { WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), "CreateChannel()"); - CriticalSectionScoped cs(*_apiCritPtr); + CriticalSectionScoped cs(_apiCritPtr); if (!_engineStatistics.Initialized()) { @@ -741,7 +741,7 @@ { WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), "DeleteChannel(channel=%d)", channel); - CriticalSectionScoped cs(*_apiCritPtr); + CriticalSectionScoped cs(_apiCritPtr); if (!_engineStatistics.Initialized()) { @@ -796,7 +796,7 @@ // 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 @@ "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 @@ { 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 @@ { 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 @@ { 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 @@ { 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 @@ { 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 @@ { 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 67f4c46..a554da8 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 @@ "SetDtmfFeedbackStatus(enable=%d, directFeeback=%d)", (int)enable, (int)directFeedback); - CriticalSectionScoped sc(*_apiCritPtr); + CriticalSectionScoped sc(_apiCritPtr); _dtmfFeedback = enable; _dtmfDirectFeedback = directFeedback; @@ -408,7 +408,7 @@ 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 0ce440d..3a86c6c 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 @@ 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 @@ { WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1), "SetPlayoutDevice(index=%d)", index); - CriticalSectionScoped cs(*_apiCritPtr); + CriticalSectionScoped cs(_apiCritPtr); ANDROID_NOT_SUPPORTED(_engineStatistics); IPHONE_NOT_SUPPORTED();