Change CriticalSectionScoped to use pointer constructor
BUG=issue183 TEST=audio_coding_module_test Review URL: https://webrtc-codereview.appspot.com/638005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2384 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
tina.legrand@webrtc.org
parent
276dc1872a
commit
fa7138f889
@ -61,7 +61,7 @@ _callbackCritSect(CriticalSectionWrapper::CreateCriticalSection())
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ACMNetEQ::~ACMNetEQ() {
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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RemoveNetEQSafe(0); // Master.
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RemoveSlavesSafe();
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}
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@ -81,7 +81,7 @@ ACMNetEQ::~ACMNetEQ() {
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WebRtc_Word32
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ACMNetEQ::Init()
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
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{
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@ -227,7 +227,7 @@ ACMNetEQ::AllocatePacketBuffer(
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// the following has to be int otherwise we will have compiler error
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// if not casted
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
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{
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if(AllocatePacketBufferByIdxSafe(usedCodecs, noOfCodecs, idx) < 0)
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@ -295,7 +295,7 @@ WebRtc_Word32
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ACMNetEQ::SetExtraDelay(
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const WebRtc_Word32 delayInMS)
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
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{
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@ -320,7 +320,7 @@ WebRtc_Word32
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ACMNetEQ::SetAVTPlayout(
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const bool enable)
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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if (_avtPlayout != enable)
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{
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for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
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@ -346,14 +346,14 @@ ACMNetEQ::SetAVTPlayout(
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bool
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ACMNetEQ::AVTPlayout() const
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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return _avtPlayout;
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}
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WebRtc_Word32
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ACMNetEQ::CurrentSampFreqHz() const
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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if(!_isInitialized[0])
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
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@ -368,7 +368,7 @@ WebRtc_Word32
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ACMNetEQ::SetPlayoutMode(
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const AudioPlayoutMode mode)
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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if(_playoutMode != mode)
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{
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for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
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@ -408,7 +408,7 @@ ACMNetEQ::SetPlayoutMode(
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AudioPlayoutMode
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ACMNetEQ::PlayoutMode() const
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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return _playoutMode;
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}
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@ -418,7 +418,7 @@ ACMNetEQ::NetworkStatistics(
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ACMNetworkStatistics* statistics) const
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{
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WebRtcNetEQ_NetworkStatistics stats;
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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if(!_isInitialized[0])
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
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@ -502,7 +502,7 @@ ACMNetEQ::RecIn(
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netEqRTPInfo.SSRC = rtpInfo.header.ssrc;
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netEqRTPInfo.markerBit = rtpInfo.header.markerBit;
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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// Down-cast the time to (32-6)-bit since we only care about
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// the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
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// we masked 6 most significant bits of 32-bit so we don't loose resolution
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@ -574,7 +574,7 @@ ACMNetEQ::RecOut(
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WebRtc_Word16 payloadLenSampleSlave;
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CriticalSectionScoped lockNetEq(*_netEqCritSect);
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CriticalSectionScoped lockNetEq(_netEqCritSect);
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if(!_receivedStereo)
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{
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@ -766,7 +766,7 @@ ACMNetEQ::AddCodec(
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"ACMNetEQ::AddCodec: error, codecDef is NULL");
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return -1;
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}
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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WebRtc_Word16 idx;
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if(toMaster)
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@ -869,7 +869,7 @@ ACMNetEQ::RTPPack(
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WebRtc_Word16
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ACMNetEQ::EnableVAD()
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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if (_vadStatus)
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{
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return 0;
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@ -899,7 +899,7 @@ ACMNetEQ::EnableVAD()
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ACMVADMode
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ACMNetEQ::VADMode() const
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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return _vadMode;
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}
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@ -908,7 +908,7 @@ WebRtc_Word16
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ACMNetEQ::SetVADMode(
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const ACMVADMode mode)
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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if((mode < VADNormal) || (mode > VADVeryAggr))
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
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@ -940,7 +940,7 @@ ACMNetEQ::SetVADMode(
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WebRtc_Word32
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ACMNetEQ::FlushBuffers()
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
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{
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if(!_isInitialized[idx])
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@ -1003,7 +1003,7 @@ ACMNetEQ::RemoveCodec(
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"RemoveCodec: NetEq error: could not Remove Codec, codec index out of range");
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return -1;
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}
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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if(!_isInitialized[0])
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
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@ -1033,7 +1033,7 @@ WebRtc_Word16
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ACMNetEQ::SetBackgroundNoiseMode(
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const ACMBackgroundNoiseMode mode)
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
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{
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if(!_isInitialized[idx])
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@ -1056,7 +1056,7 @@ ACMNetEQ::BackgroundNoiseMode(
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ACMBackgroundNoiseMode& mode)
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{
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WebRtcNetEQBGNMode myMode;
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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if(!_isInitialized[0])
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
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@ -1079,7 +1079,7 @@ void
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ACMNetEQ::SetUniqueId(
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WebRtc_Word32 id)
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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_id = id;
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}
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@ -1109,7 +1109,7 @@ WebRtc_Word32
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ACMNetEQ::PlayoutTimestamp(
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WebRtc_UWord32& timestamp)
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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if(WebRtcNetEQ_GetSpeechTimeStamp(_inst[0], ×tamp) < 0)
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{
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LogError("GetSpeechTimeStamp", 0);
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@ -1122,7 +1122,7 @@ ACMNetEQ::PlayoutTimestamp(
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}
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void ACMNetEQ::RemoveSlaves() {
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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RemoveSlavesSafe();
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}
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@ -1158,7 +1158,7 @@ ACMNetEQ::AddSlave(
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const WebRtcNetEQDecoder* usedCodecs,
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WebRtc_Word16 noOfCodecs)
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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const WebRtc_Word16 slaveIdx = 1;
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if(_numSlaves < 1)
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{
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@ -1263,14 +1263,14 @@ void
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ACMNetEQ::SetReceivedStereo(
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bool receivedStereo)
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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_receivedStereo = receivedStereo;
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}
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WebRtc_UWord8
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ACMNetEQ::NumSlaves()
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{
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CriticalSectionScoped lock(*_netEqCritSect);
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CriticalSectionScoped lock(_netEqCritSect);
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return _numSlaves;
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}
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@ -1,5 +1,5 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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@ -11,73 +11,62 @@
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#include <string.h>
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#include "acm_resampler.h"
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#include "critical_section_wrapper.h"
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#include "resampler.h"
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#include "signal_processing_library.h"
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#include "trace.h"
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namespace webrtc
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{
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namespace webrtc {
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ACMResampler::ACMResampler():
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_resamplerCritSect(*CriticalSectionWrapper::CreateCriticalSection())
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{
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ACMResampler::ACMResampler()
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: _resamplerCritSect(CriticalSectionWrapper::CreateCriticalSection()) {
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}
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ACMResampler::~ACMResampler()
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{
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delete &_resamplerCritSect;
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ACMResampler::~ACMResampler() {
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delete _resamplerCritSect;
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}
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WebRtc_Word16 ACMResampler::Resample10Msec(const WebRtc_Word16* inAudio,
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WebRtc_Word32 inFreqHz,
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WebRtc_Word16* outAudio,
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WebRtc_Word32 outFreqHz,
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WebRtc_UWord8 numAudioChannels) {
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CriticalSectionScoped cs(_resamplerCritSect);
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WebRtc_Word16
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ACMResampler::Resample10Msec(
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const WebRtc_Word16* inAudio,
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WebRtc_Word32 inFreqHz,
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WebRtc_Word16* outAudio,
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WebRtc_Word32 outFreqHz,
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WebRtc_UWord8 numAudioChannels)
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{
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if (inFreqHz == outFreqHz) {
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size_t length = static_cast<size_t>(inFreqHz * numAudioChannels / 100);
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memcpy(outAudio, inAudio, length * sizeof(WebRtc_Word16));
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return static_cast<WebRtc_Word16>(inFreqHz / 100);
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}
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CriticalSectionScoped cs(_resamplerCritSect);
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// |maxLen| is maximum number of samples for 10ms at 48kHz.
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int maxLen = 480 * numAudioChannels;
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int lengthIn = (WebRtc_Word16)(inFreqHz / 100) * numAudioChannels;
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int outLen;
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if(inFreqHz == outFreqHz)
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{
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size_t length = static_cast<size_t>(inFreqHz * numAudioChannels / 100);
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memcpy(outAudio, inAudio, length * sizeof(WebRtc_Word16));
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return static_cast<WebRtc_Word16>(inFreqHz / 100);
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}
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WebRtc_Word32 ret;
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ResamplerType type;
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type = (numAudioChannels == 1) ? kResamplerSynchronous :
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kResamplerSynchronousStereo;
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int maxLen = 480 * numAudioChannels; //max number of samples for 10ms at 48kHz
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int lengthIn = (WebRtc_Word16)(inFreqHz / 100) * numAudioChannels;
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int outLen;
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ret = _resampler.ResetIfNeeded(inFreqHz, outFreqHz, type);
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if (ret < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
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"Error in reset of resampler");
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return -1;
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}
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WebRtc_Word32 ret;
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ResamplerType type;
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type = (numAudioChannels == 1)? kResamplerSynchronous:kResamplerSynchronousStereo;
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ret = _resampler.Push(inAudio, lengthIn, outAudio, maxLen, outLen);
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if (ret < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
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"Error in resampler: resampler.Push");
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return -1;
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}
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ret = _resampler.ResetIfNeeded(inFreqHz,outFreqHz,type);
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if (ret < 0)
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
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"Error in reset of resampler");
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return -1;
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}
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ret = _resampler.Push(inAudio, lengthIn, outAudio, maxLen, outLen);
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if (ret < 0 )
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{
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
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"Error in resampler: resampler.Push");
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return -1;
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}
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WebRtc_Word16 outAudioLenSmpl = (WebRtc_Word16) outLen / numAudioChannels;
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return outAudioLenSmpl;
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WebRtc_Word16 outAudioLenSmpl = (WebRtc_Word16) outLen / numAudioChannels;
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return outAudioLenSmpl;
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}
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} // namespace webrtc
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} // namespace webrtc
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@ -1,5 +1,5 @@
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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@ -18,26 +18,23 @@ namespace webrtc {
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class CriticalSectionWrapper;
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class ACMResampler
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{
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public:
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ACMResampler();
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~ACMResampler();
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class ACMResampler {
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public:
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ACMResampler();
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~ACMResampler();
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WebRtc_Word16 Resample10Msec(
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const WebRtc_Word16* inAudio,
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const WebRtc_Word32 inFreqHz,
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WebRtc_Word16* outAudio,
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const WebRtc_Word32 outFreqHz,
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WebRtc_UWord8 numAudioChannels);
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WebRtc_Word16 Resample10Msec(const WebRtc_Word16* inAudio,
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const WebRtc_Word32 inFreqHz,
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WebRtc_Word16* outAudio,
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const WebRtc_Word32 outFreqHz,
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WebRtc_UWord8 numAudioChannels);
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private:
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//Use the Resampler class
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Resampler _resampler;
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CriticalSectionWrapper& _resamplerCritSect;
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private:
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// Use the Resampler class.
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Resampler _resampler;
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CriticalSectionWrapper* _resamplerCritSect;
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};
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} // namespace webrtc
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
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@ -137,7 +137,7 @@ AudioCodingModuleImpl::AudioCodingModuleImpl(const WebRtc_Word32 id)
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AudioCodingModuleImpl::~AudioCodingModuleImpl() {
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{
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CriticalSectionScoped lock(*_acmCritSect);
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CriticalSectionScoped lock(_acmCritSect);
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_currentSendCodecIdx = -1;
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for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++) {
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@ -211,7 +211,7 @@ AudioCodingModuleImpl::~AudioCodingModuleImpl() {
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WebRtc_Word32 AudioCodingModuleImpl::ChangeUniqueId(const WebRtc_Word32 id) {
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{
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CriticalSectionScoped lock(*_acmCritSect);
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CriticalSectionScoped lock(_acmCritSect);
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_id = id;
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#ifdef ACM_QA_TEST
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@ -246,7 +246,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ChangeUniqueId(const WebRtc_Word32 id) {
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// Returns the number of milliseconds until the module want a
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// worker thread to call Process.
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WebRtc_Word32 AudioCodingModuleImpl::TimeUntilNextProcess() {
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CriticalSectionScoped lock(*_acmCritSect);
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CriticalSectionScoped lock(_acmCritSect);
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if (!HaveValidEncoder("TimeUntilNextProcess")) {
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return -1;
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@ -271,7 +271,7 @@ WebRtc_Word32 AudioCodingModuleImpl::Process() {
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// Keep the scope of the ACM critical section limited.
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{
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CriticalSectionScoped lock(*_acmCritSect);
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CriticalSectionScoped lock(_acmCritSect);
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if (!HaveValidEncoder("Process")) {
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return -1;
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}
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@ -428,7 +428,7 @@ WebRtc_Word32 AudioCodingModuleImpl::Process() {
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}
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|
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if (has_data_to_send) {
|
||||
CriticalSectionScoped lock(*_callbackCritSect);
|
||||
CriticalSectionScoped lock(_callbackCritSect);
|
||||
#ifdef ACM_QA_TEST
|
||||
if(_outgoingPL != NULL) {
|
||||
if (fwrite(&rtp_timestamp, sizeof(WebRtc_UWord32), 1, _outgoingPL) != 1) {
|
||||
@ -476,7 +476,7 @@ WebRtc_Word32 AudioCodingModuleImpl::Process() {
|
||||
|
||||
// Initialize send codec.
|
||||
WebRtc_Word32 AudioCodingModuleImpl::InitializeSender() {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
// Start with invalid values.
|
||||
_sendCodecRegistered = false;
|
||||
@ -512,7 +512,7 @@ WebRtc_Word32 AudioCodingModuleImpl::InitializeSender() {
|
||||
}
|
||||
|
||||
WebRtc_Word32 AudioCodingModuleImpl::ResetEncoder() {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
if (!HaveValidEncoder("ResetEncoder")) {
|
||||
return -1;
|
||||
}
|
||||
@ -520,7 +520,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ResetEncoder() {
|
||||
}
|
||||
|
||||
void AudioCodingModuleImpl::UnregisterSendCodec() {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
_sendCodecRegistered = false;
|
||||
_currentSendCodecIdx = -1;
|
||||
return;
|
||||
@ -557,7 +557,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterSendCodec(
|
||||
int mirror_id;
|
||||
int codec_id = ACMCodecDB::CodecNumber(&send_codec, &mirror_id, error_message,
|
||||
sizeof(error_message));
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
// Check for reported errors from function CodecNumber().
|
||||
if (codec_id < 0) {
|
||||
@ -815,7 +815,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SendCodec(
|
||||
CodecInst& current_codec) const {
|
||||
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
|
||||
"SendCodec()");
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if (!_sendCodecRegistered) {
|
||||
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
|
||||
@ -835,7 +835,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SendCodec(
|
||||
WebRtc_Word32 AudioCodingModuleImpl::SendFrequency() const {
|
||||
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
|
||||
"SendFrequency()");
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if (!_sendCodecRegistered) {
|
||||
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
|
||||
@ -851,7 +851,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SendFrequency() const {
|
||||
// Adaptive rate codecs return their current encode target rate, while other
|
||||
// codecs return there longterm avarage or their fixed rate.
|
||||
WebRtc_Word32 AudioCodingModuleImpl::SendBitrate() const {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if (!_sendCodecRegistered) {
|
||||
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
|
||||
@ -877,7 +877,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(
|
||||
// the encoded buffers.
|
||||
WebRtc_Word32 AudioCodingModuleImpl::RegisterTransportCallback(
|
||||
AudioPacketizationCallback* transport) {
|
||||
CriticalSectionScoped lock(*_callbackCritSect);
|
||||
CriticalSectionScoped lock(_callbackCritSect);
|
||||
_packetizationCallback = transport;
|
||||
return 0;
|
||||
}
|
||||
@ -896,12 +896,12 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
|
||||
|
||||
// Enter the critical section for callback.
|
||||
{
|
||||
CriticalSectionScoped lock(*_callbackCritSect);
|
||||
CriticalSectionScoped lock(_callbackCritSect);
|
||||
_dtmfCallback = incoming_message;
|
||||
}
|
||||
// Enter the ACM critical section to set up the DTMF class.
|
||||
{
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
// Check if the call is to disable or enable the callback.
|
||||
if (incoming_message == NULL) {
|
||||
// Callback is disabled, delete DTMF-detector class.
|
||||
@ -932,7 +932,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
|
||||
// Check if we failed in setting up the DTMF-detector class.
|
||||
if ((status < 0)) {
|
||||
// We failed, we cannot have the callback.
|
||||
CriticalSectionScoped lock(*_callbackCritSect);
|
||||
CriticalSectionScoped lock(_callbackCritSect);
|
||||
_dtmfCallback = NULL;
|
||||
}
|
||||
|
||||
@ -944,7 +944,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
|
||||
WebRtc_Word32 AudioCodingModuleImpl::Add10MsData(
|
||||
const AudioFrame& audio_frame) {
|
||||
// Do we have a codec registered?
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
if (!HaveValidEncoder("Add10MsData")) {
|
||||
return -1;
|
||||
}
|
||||
@ -1058,7 +1058,7 @@ WebRtc_Word32 AudioCodingModuleImpl::Add10MsData(
|
||||
//
|
||||
|
||||
bool AudioCodingModuleImpl::FECStatus() const {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
return _fecEnabled;
|
||||
}
|
||||
|
||||
@ -1067,7 +1067,7 @@ WebRtc_Word32
|
||||
AudioCodingModuleImpl::SetFECStatus(
|
||||
#ifdef WEBRTC_CODEC_RED
|
||||
const bool enable_fec) {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if (_fecEnabled != enable_fec) {
|
||||
// Reset the RED buffer.
|
||||
@ -1104,7 +1104,7 @@ AudioCodingModuleImpl::SetFECStatus(
|
||||
WebRtc_Word32 AudioCodingModuleImpl::SetVAD(const bool enable_dtx,
|
||||
const bool enable_vad,
|
||||
const ACMVADMode mode) {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
// Sanity check of the mode.
|
||||
if ((mode != VADNormal) && (mode != VADLowBitrate)
|
||||
@ -1149,7 +1149,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SetVAD(const bool enable_dtx,
|
||||
// Get VAD/DTX settings.
|
||||
WebRtc_Word32 AudioCodingModuleImpl::VAD(bool& dtx_enabled, bool& vad_enabled,
|
||||
ACMVADMode& mode) const {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
dtx_enabled = _dtxEnabled;
|
||||
vad_enabled = _vadEnabled;
|
||||
@ -1163,7 +1163,7 @@ WebRtc_Word32 AudioCodingModuleImpl::VAD(bool& dtx_enabled, bool& vad_enabled,
|
||||
//
|
||||
|
||||
WebRtc_Word32 AudioCodingModuleImpl::InitializeReceiver() {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
return InitializeReceiverSafe();
|
||||
}
|
||||
|
||||
@ -1213,7 +1213,7 @@ WebRtc_Word32 AudioCodingModuleImpl::InitializeReceiverSafe() {
|
||||
|
||||
// Reset the decoder state.
|
||||
WebRtc_Word32 AudioCodingModuleImpl::ResetDecoder() {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) {
|
||||
if ((_codecs[id] != NULL) && (_registeredPlTypes[id] != -1)) {
|
||||
@ -1233,7 +1233,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ReceiveFrequency() const {
|
||||
"ReceiveFrequency()");
|
||||
WebRtcACMCodecParams codec_params;
|
||||
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
if (DecoderParamByPlType(_lastRecvAudioCodecPlType, codec_params) < 0) {
|
||||
return _netEq.CurrentSampFreqHz();
|
||||
} else {
|
||||
@ -1246,7 +1246,7 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutFrequency() const {
|
||||
WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
|
||||
"PlayoutFrequency()");
|
||||
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
return _netEq.CurrentSampFreqHz();
|
||||
}
|
||||
@ -1255,7 +1255,7 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutFrequency() const {
|
||||
// for codecs, CNG (NB, WB and SWB), DTMF, RED.
|
||||
WebRtc_Word32 AudioCodingModuleImpl::RegisterReceiveCodec(
|
||||
const CodecInst& receive_codec) {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if (receive_codec.channels > 2) {
|
||||
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
|
||||
@ -1456,7 +1456,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterRecCodecMSSafe(
|
||||
WebRtc_Word32 AudioCodingModuleImpl::ReceiveCodec(
|
||||
CodecInst& current_codec) const {
|
||||
WebRtcACMCodecParams decoderParam;
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
for (int id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) {
|
||||
if (_codecs[id] != NULL) {
|
||||
@ -1493,7 +1493,7 @@ WebRtc_Word32 AudioCodingModuleImpl::IncomingPacket(
|
||||
{
|
||||
// Store the payload Type. This will be used to retrieve "received codec"
|
||||
// and "received frequency."
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
#ifdef ACM_QA_TEST
|
||||
if(_incomingPL != NULL) {
|
||||
if (fwrite(&rtp_info.header.timestamp, sizeof(WebRtc_UWord32),
|
||||
@ -1763,7 +1763,7 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
|
||||
|
||||
// Limit the scope of ACM Critical section.
|
||||
{
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if ((receive_freq != desired_freq_hz) && (desired_freq_hz != -1)) {
|
||||
// Resample payloadData.
|
||||
@ -1844,7 +1844,7 @@ WebRtc_Word32 AudioCodingModuleImpl::PlayoutData10Ms(
|
||||
|
||||
if (tone_detected) {
|
||||
// We will deal with callback here, so enter callback critical section.
|
||||
CriticalSectionScoped lock(*_callbackCritSect);
|
||||
CriticalSectionScoped lock(_callbackCritSect);
|
||||
|
||||
if (_dtmfCallback != NULL) {
|
||||
if (tone != kACMToneEnd) {
|
||||
@ -1936,7 +1936,7 @@ WebRtc_Word32 AudioCodingModuleImpl::RegisterVADCallback(
|
||||
ACMVADCallback* vad_callback) {
|
||||
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
|
||||
"RegisterVADCallback()");
|
||||
CriticalSectionScoped lock(*_callbackCritSect);
|
||||
CriticalSectionScoped lock(_callbackCritSect);
|
||||
_vadCallback = vad_callback;
|
||||
return 0;
|
||||
}
|
||||
@ -2017,7 +2017,7 @@ WebRtc_Word32 AudioCodingModuleImpl::IncomingPayload(
|
||||
WebRtc_Word16 AudioCodingModuleImpl::DecoderParamByPlType(
|
||||
const WebRtc_UWord8 payload_type,
|
||||
WebRtcACMCodecParams& codec_params) const {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
for (WebRtc_Word16 id = 0; id < ACMCodecDB::kMaxNumCodecs;
|
||||
id++) {
|
||||
if (_codecs[id] != NULL) {
|
||||
@ -2041,7 +2041,7 @@ WebRtc_Word16 AudioCodingModuleImpl::DecoderParamByPlType(
|
||||
WebRtc_Word16 AudioCodingModuleImpl::DecoderListIDByPlName(
|
||||
const char* name, const WebRtc_UWord16 frequency) const {
|
||||
WebRtcACMCodecParams codec_params;
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
for (WebRtc_Word16 id = 0; id < ACMCodecDB::kMaxNumCodecs; id++) {
|
||||
if ((_codecs[id] != NULL)) {
|
||||
if (_codecs[id]->DecoderInitialized()) {
|
||||
@ -2071,7 +2071,7 @@ WebRtc_Word16 AudioCodingModuleImpl::DecoderListIDByPlName(
|
||||
|
||||
WebRtc_Word32 AudioCodingModuleImpl::LastEncodedTimestamp(
|
||||
WebRtc_UWord32& timestamp) const {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
if (!HaveValidEncoder("LastEncodedTimestamp")) {
|
||||
return -1;
|
||||
}
|
||||
@ -2081,7 +2081,7 @@ WebRtc_Word32 AudioCodingModuleImpl::LastEncodedTimestamp(
|
||||
|
||||
WebRtc_Word32 AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(
|
||||
bool use_webrtc_dtx) {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if (!HaveValidEncoder("ReplaceInternalDTXWithWebRtc")) {
|
||||
WEBRTC_TRACE(
|
||||
@ -2107,7 +2107,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(
|
||||
|
||||
WebRtc_Word32 AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
|
||||
bool& uses_webrtc_dtx) {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if (!HaveValidEncoder("IsInternalDTXReplacedWithWebRtc")) {
|
||||
return -1;
|
||||
@ -2121,7 +2121,7 @@ WebRtc_Word32 AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(
|
||||
|
||||
WebRtc_Word32 AudioCodingModuleImpl::SetISACMaxRate(
|
||||
const WebRtc_UWord32 max_bit_per_sec) {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if (!HaveValidEncoder("SetISACMaxRate")) {
|
||||
return -1;
|
||||
@ -2132,7 +2132,7 @@ WebRtc_Word32 AudioCodingModuleImpl::SetISACMaxRate(
|
||||
|
||||
WebRtc_Word32 AudioCodingModuleImpl::SetISACMaxPayloadSize(
|
||||
const WebRtc_UWord16 max_size_bytes) {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if (!HaveValidEncoder("SetISACMaxPayloadSize")) {
|
||||
return -1;
|
||||
@ -2146,7 +2146,7 @@ WebRtc_Word32 AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
|
||||
const WebRtc_UWord8 frame_size_ms,
|
||||
const WebRtc_UWord16 rate_bit_per_sec,
|
||||
const bool enforce_frame_size) {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
|
||||
if (!HaveValidEncoder("ConfigISACBandwidthEstimator")) {
|
||||
return -1;
|
||||
@ -2201,7 +2201,7 @@ bool AudioCodingModuleImpl::HaveValidEncoder(const char* caller_name) const {
|
||||
|
||||
WebRtc_Word32 AudioCodingModuleImpl::UnregisterReceiveCodec(
|
||||
const WebRtc_Word16 payload_type) {
|
||||
CriticalSectionScoped lock(*_acmCritSect);
|
||||
CriticalSectionScoped lock(_acmCritSect);
|
||||
int id;
|
||||
|
||||
// Search through the list of registered payload types.
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user