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Remove the deprecated kTraceModuleCall trace from audio coding module.

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Review URL: https://webrtc-codereview.appspot.com/399002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1741 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
xians@webrtc.org
2012-02-22 08:35:03 +00:00
parent 20e9cf274d
commit 539ef94f20
2 changed files with 1 additions and 106 deletions
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14
src/modules/audio_coding/main/source/audio_coding_module.cc
View File

@@ -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 * 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 * 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 // Get number of supported codecs
WebRtc_UWord8 AudioCodingModule::NumberOfCodecs() WebRtc_UWord8 AudioCodingModule::NumberOfCodecs()
{ {
WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, -1,
"NumberOfCodecs()");
return static_cast<WebRtc_UWord8>(ACMCodecDB::kNumCodecs); return static_cast<WebRtc_UWord8>(ACMCodecDB::kNumCodecs);
} }
@@ -47,9 +45,6 @@ AudioCodingModule::Codec(
const WebRtc_UWord8 listId, const WebRtc_UWord8 listId,
CodecInst& codec) 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 // Get the codec settings for the codec with the given list ID
return ACMCodecDB::Codec(listId, &codec); return ACMCodecDB::Codec(listId, &codec);
} }
@@ -61,9 +56,6 @@ AudioCodingModule::Codec(
CodecInst& codec, CodecInst& codec,
const WebRtc_Word32 samplingFreqHz) 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 // Search through codec list for a matching name
for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++) for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++)
{ {
@@ -97,8 +89,6 @@ AudioCodingModule::Codec(
const WebRtc_Word8* payloadName, const WebRtc_Word8* payloadName,
const WebRtc_Word32 samplingFreqHz) const WebRtc_Word32 samplingFreqHz)
{ {
WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, -1,
"Codec(const WebRtc_Word8* payloadName)");
CodecInst codec; CodecInst codec;
// Search through codec list for a matching name // Search through codec list for a matching name
@@ -130,8 +120,6 @@ AudioCodingModule::IsCodecValid(
int mirrorID; int mirrorID;
char errMsg[500]; char errMsg[500];
WEBRTC_TRACE(webrtc::kTraceModuleCall, webrtc::kTraceAudioCoding, -1,
"IsCodecValid(const CodecInst& codec)");
int codecNumber = ACMCodecDB::CodecNumber(&codec, &mirrorID, errMsg, 500); int codecNumber = ACMCodecDB::CodecNumber(&codec, &mirrorID, errMsg, 500);
if(codecNumber < 0) if(codecNumber < 0)

93
src/modules/audio_coding/main/source/audio_coding_module_impl.cc
View File

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