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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
* 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<WebRtc_UWord8>(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)

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

@@ -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;