diff --git a/src/modules/audio_coding/main/source/audio_coding_module.cc b/src/modules/audio_coding/main/source/audio_coding_module.cc index e4f5a5f7c..00dca2544 100644 --- a/src/modules/audio_coding/main/source/audio_coding_module.cc +++ b/src/modules/audio_coding/main/source/audio_coding_module.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 @@ -36,8 +36,6 @@ AudioCodingModule::Destroy( // Get number of supported codecs WebRtc_UWord8 AudioCodingModule::NumberOfCodecs() { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, -1, - "NumberOfCodecs()"); return static_cast(ACMCodecDB::kNumCodecs); } @@ -47,9 +45,6 @@ AudioCodingModule::Codec( const WebRtc_UWord8 listId, CodecInst& codec) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, -1, - "Codec(const WebRtc_UWord8 listId, CodecInst& codec)"); - // Get the codec settings for the codec with the given list ID return ACMCodecDB::Codec(listId, &codec); } @@ -61,9 +56,6 @@ AudioCodingModule::Codec( CodecInst& codec, const WebRtc_Word32 samplingFreqHz) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, -1, - "Codec(const WebRtc_Word8* payloadName, CodecInst& codec)"); - // Search through codec list for a matching name for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++) { @@ -97,8 +89,6 @@ AudioCodingModule::Codec( const WebRtc_Word8* payloadName, const WebRtc_Word32 samplingFreqHz) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, -1, - "Codec(const WebRtc_Word8* payloadName)"); CodecInst codec; // Search through codec list for a matching name @@ -130,8 +120,6 @@ AudioCodingModule::IsCodecValid( int mirrorID; char errMsg[500]; - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, -1, - "IsCodecValid(const CodecInst& codec)"); int codecNumber = ACMCodecDB::CodecNumber(&codec, &mirrorID, errMsg, 500); if(codecNumber < 0) diff --git a/src/modules/audio_coding/main/source/audio_coding_module_impl.cc b/src/modules/audio_coding/main/source/audio_coding_module_impl.cc index 6691fa3ce..57647f012 100644 --- a/src/modules/audio_coding/main/source/audio_coding_module_impl.cc +++ b/src/modules/audio_coding/main/source/audio_coding_module_impl.cc @@ -227,7 +227,6 @@ AudioCodingModuleImpl::~AudioCodingModuleImpl() } } - #ifdef ACM_QA_TEST if(_incomingPL != NULL) { @@ -252,8 +251,6 @@ WebRtc_Word32 AudioCodingModuleImpl::ChangeUniqueId( const WebRtc_Word32 id) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "ChangeUniqueId(new id:%d)", id); { CriticalSectionScoped lock(*_acmCritSect); _id = id; @@ -550,9 +547,6 @@ AudioCodingModuleImpl::Process() WebRtc_Word32 AudioCodingModuleImpl::InitializeSender() { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "InitializeSender()"); - CriticalSectionScoped lock(*_acmCritSect); _sendCodecRegistered = false; @@ -592,9 +586,6 @@ AudioCodingModuleImpl::InitializeSender() WebRtc_Word32 AudioCodingModuleImpl::ResetEncoder() { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "ResetEncoder()"); - CriticalSectionScoped lock(*_acmCritSect); if(!HaveValidEncoder("ResetEncoder")) { @@ -617,9 +608,6 @@ ACMGenericCodec* AudioCodingModuleImpl::CreateCodec( const CodecInst& codec) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "CreateCodec()"); - ACMGenericCodec* myCodec = NULL; myCodec = ACMCodecDB::CreateCodecInstance(&codec); @@ -644,9 +632,6 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterSendCodec( const CodecInst& sendCodec) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "Registering Send Codec"); - if((sendCodec.channels != 1) && (sendCodec.channels != 2)) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id, @@ -704,7 +689,6 @@ mono codecs are supported, i.e. channels=1.", sendCodec.channels); if(!STR_CASE_CMP(sendCodec.plname, "CN")) { // CNG is registered - switch(sendCodec.plfreq) { case 8000: @@ -998,9 +982,6 @@ AudioCodingModuleImpl::SendFrequency() const WebRtc_Word32 AudioCodingModuleImpl::SendBitrate() const { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SendBitrate()"); - CriticalSectionScoped lock(*_acmCritSect); if(!_sendCodecRegistered) @@ -1023,8 +1004,6 @@ WebRtc_Word32 AudioCodingModuleImpl::SetReceivedEstimatedBandwidth( const WebRtc_Word32 bw ) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SetReceivedEstimatedBandwidth()"); return _codecs[_currentSendCodecIdx]->SetEstimatedBandwidth(bw); } @@ -1034,8 +1013,6 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterTransportCallback( AudioPacketizationCallback* transport) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "RegisterTransportCallback()"); CriticalSectionScoped lock(*_callbackCritSect); _packetizationCallback = transport; return 0; @@ -1049,8 +1026,6 @@ AudioCodingModuleImpl::RegisterIncomingMessagesCallback( AudioCodingFeedback* /* incomingMessagesCallback */, const ACMCountries /* cpt */) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "RegisterIncomingMessagesCallback()"); return -1; #else AudioCodingFeedback* incomingMessagesCallback, @@ -1245,8 +1220,6 @@ match"); bool AudioCodingModuleImpl::FECStatus() const { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "FECStatus()"); CriticalSectionScoped lock(*_acmCritSect); return _fecEnabled; } @@ -1257,8 +1230,6 @@ AudioCodingModuleImpl::SetFECStatus( #ifdef WEBRTC_CODEC_RED const bool enableFEC) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SetFECStatus()"); CriticalSectionScoped lock(*_acmCritSect); if (_fecEnabled != enableFEC) @@ -1300,8 +1271,6 @@ AudioCodingModuleImpl::SetVAD( const bool enableVAD, const ACMVADMode vadMode) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SetVAD()"); CriticalSectionScoped lock(*_acmCritSect); // sanity check of the mode @@ -1352,8 +1321,6 @@ AudioCodingModuleImpl::VAD( bool& vadEnabled, ACMVADMode& vadMode) const { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "VAD()"); CriticalSectionScoped lock(*_acmCritSect); dtxEnabled = _dtxEnabled; @@ -1378,9 +1345,6 @@ AudioCodingModuleImpl::InitializeReceiver() WebRtc_Word32 AudioCodingModuleImpl::InitializeReceiverSafe() { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "InitializeReceiver()"); - // If the receiver is already initialized then we // also like to destruct decoders if any exist. After a call // to this function, we should have a clean start-up. @@ -1441,8 +1405,6 @@ AudioCodingModuleImpl::InitializeReceiverSafe() WebRtc_Word32 AudioCodingModuleImpl::ResetDecoder() { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "ResetDecoder()"); CriticalSectionScoped lock(*_acmCritSect); for(int codecCntr = 0; codecCntr < ACMCodecDB::kMaxNumCodecs; codecCntr++) @@ -1500,9 +1462,6 @@ AudioCodingModuleImpl::RegisterReceiveCodec( { CriticalSectionScoped lock(*_acmCritSect); - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "RegisterReceiveCodec()"); - if(receiveCodec.channels > 2) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id, @@ -1557,7 +1516,6 @@ AudioCodingModuleImpl::RegisterReceiveCodec( return -1; } - // If receive stereo, make sure we have two instances of NetEQ, one for each channel if(receiveCodec.channels == 2) { @@ -1631,9 +1589,6 @@ AudioCodingModuleImpl::RegisterRecCodecMSSafe( WebRtc_Word16 mirrorId, ACMNetEQ::JB jitterBuffer) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "RegisterReceiveCodecMSSafe()"); - ACMGenericCodec** codecArray; if(jitterBuffer == ACMNetEQ::masterJB) { @@ -1729,8 +1684,6 @@ WebRtc_Word32 AudioCodingModuleImpl::ReceiveCodec( CodecInst& currentReceiveCodec) const { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "ReceiveCodec()"); WebRtcACMCodecParams decoderParam; CriticalSectionScoped lock(*_acmCritSect); @@ -1765,7 +1718,6 @@ AudioCodingModuleImpl::IncomingPacket( const WebRtc_Word32 payloadLength, const WebRtcRTPHeader& rtpInfo) { - if (payloadLength < 0) { // Log error @@ -1877,8 +1829,6 @@ WebRtc_Word32 AudioCodingModuleImpl::SetMinimumPlayoutDelay( const WebRtc_Word32 timeMs) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SetMinimumPlayoutDelay()"); if((timeMs < 0) || (timeMs > 1000)) { WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id, @@ -1892,8 +1842,6 @@ AudioCodingModuleImpl::SetMinimumPlayoutDelay( bool AudioCodingModuleImpl::DtmfPlayoutStatus() const { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "DtmfPlayoutStatus()"); #ifndef WEBRTC_CODEC_AVT return false; #else @@ -1914,8 +1862,6 @@ AudioCodingModuleImpl::SetDtmfPlayoutStatus( #else const bool enable) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SetDtmfPlayoutStatus()"); return _netEq.SetAVTPlayout(enable); #endif } @@ -1926,9 +1872,6 @@ AudioCodingModuleImpl::SetDtmfPlayoutStatus( WebRtc_Word32 AudioCodingModuleImpl::DecoderEstimatedBandwidth() const { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "DecoderEstimatedBandwidth()"); - CodecInst codecInst; WebRtc_Word16 codecID = -1; int plTypWB; @@ -1972,8 +1915,6 @@ WebRtc_Word32 AudioCodingModuleImpl::SetPlayoutMode( const AudioPlayoutMode mode) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SetPlayoutMode()"); if((mode != voice) && (mode != fax) && (mode != streaming)) @@ -1989,8 +1930,6 @@ AudioCodingModuleImpl::SetPlayoutMode( AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "PlayoutMode()"); return _netEq.PlayoutMode(); } @@ -2155,8 +2094,6 @@ AudioCodingModuleImpl::PlayoutData10Ms( return 0; } - - ///////////////////////////////////////// // (CNG) Comfort Noise Generation // Generate comfort noise when receiving DTX packets @@ -2166,8 +2103,6 @@ AudioCodingModuleImpl::PlayoutData10Ms( ACMVADMode AudioCodingModuleImpl::ReceiveVADMode() const { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "ReceiveVADMode()"); return _netEq.VADMode(); } @@ -2176,8 +2111,6 @@ WebRtc_Word16 AudioCodingModuleImpl::SetReceiveVADMode( const ACMVADMode mode) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SetReceiveVADMode()"); return _netEq.SetVADMode(mode); } @@ -2189,8 +2122,6 @@ WebRtc_Word32 AudioCodingModuleImpl::NetworkStatistics( ACMNetworkStatistics& statistics) const { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "NetworkStatistics()"); WebRtc_Word32 status; status = _netEq.NetworkStatistics(&statistics); return status; @@ -2423,8 +2354,6 @@ AudioCodingModuleImpl::DecoderListIDByPlName( WebRtc_Word32 AudioCodingModuleImpl::LastEncodedTimestamp(WebRtc_UWord32& timestamp) const { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "LastEncodedTimestamp()"); CriticalSectionScoped lock(*_acmCritSect); if(!HaveValidEncoder("LastEncodedTimestamp")) { @@ -2437,8 +2366,6 @@ AudioCodingModuleImpl::LastEncodedTimestamp(WebRtc_UWord32& timestamp) const WebRtc_Word32 AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(bool useWebRtcDTX) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "ReplaceInternalDTXWithWebRtc()"); CriticalSectionScoped lock(*_acmCritSect); if(!HaveValidEncoder("ReplaceInternalDTXWithWebRtc")) @@ -2466,8 +2393,6 @@ AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(bool useWebRtcDTX) WebRtc_Word32 AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(bool& usesWebRtcDTX) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "IsInternalDTXReplacedWithWebRtc()"); CriticalSectionScoped lock(*_acmCritSect); if(!HaveValidEncoder("IsInternalDTXReplacedWithWebRtc")) @@ -2486,8 +2411,6 @@ WebRtc_Word32 AudioCodingModuleImpl::SetISACMaxRate( const WebRtc_UWord32 maxRateBitPerSec) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SetISACMaxRate()"); CriticalSectionScoped lock(*_acmCritSect); if(!HaveValidEncoder("SetISACMaxRate")) @@ -2503,8 +2426,6 @@ WebRtc_Word32 AudioCodingModuleImpl::SetISACMaxPayloadSize( const WebRtc_UWord16 maxPayloadLenBytes) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SetISACPayloadSize()"); CriticalSectionScoped lock(*_acmCritSect); if(!HaveValidEncoder("SetISACMaxPayloadSize")) @@ -2521,8 +2442,6 @@ AudioCodingModuleImpl::ConfigISACBandwidthEstimator( const WebRtc_UWord16 initRateBitPerSec, const bool enforceFrameSize) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "ConfigISACBandwidthEstimator()"); CriticalSectionScoped lock(*_acmCritSect); if(!HaveValidEncoder("ConfigISACBandwidthEstimator")) @@ -2538,8 +2457,6 @@ WebRtc_Word32 AudioCodingModuleImpl::SetBackgroundNoiseMode( const ACMBackgroundNoiseMode mode) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "SetBackgroundNoiseMode()"); if((mode < On) || (mode > Off)) { @@ -2554,8 +2471,6 @@ WebRtc_Word32 AudioCodingModuleImpl::BackgroundNoiseMode( ACMBackgroundNoiseMode& mode) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "BackgroundNoiseMode()"); return _netEq.BackgroundNoiseMode(mode); } @@ -2568,10 +2483,6 @@ AudioCodingModuleImpl::PlayoutTimestamp( return _netEq.PlayoutTimestamp(timestamp); } - - - - bool AudioCodingModuleImpl::HaveValidEncoder( const WebRtc_Word8* callerName) const @@ -2604,8 +2515,6 @@ WebRtc_Word32 AudioCodingModuleImpl::UnregisterReceiveCodec( const WebRtc_Word16 payloadType) { - WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, _id, - "UnregisterReceiveCodec()"); CriticalSectionScoped lock(*_acmCritSect); WebRtc_Word16 codecID; @@ -2693,7 +2602,6 @@ AudioCodingModuleImpl::UnregisterReceiveCodecSafe( return 0; } - WebRtc_Word32 AudioCodingModuleImpl::REDPayloadISAC( const WebRtc_Word32 isacRate, @@ -2701,7 +2609,6 @@ AudioCodingModuleImpl::REDPayloadISAC( WebRtc_UWord8* payload, WebRtc_Word16* payloadLenByte) { - if(!HaveValidEncoder("EncodeData")) { return -1;