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Implement simple audio compressor for all input and output. #672

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37 changes: 32 additions & 5 deletions src/client.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -24,6 +24,7 @@

#include "client.h"

#include "compressor.h"

/* Implementation *************************************************************/
CClient::CClient ( const quint16 iPortNumber,
Expand Down Expand Up @@ -54,6 +55,8 @@ CClient::CClient ( const quint16 iPortNumber,
iSndCrdFrameSizeFactor ( FRAME_SIZE_FACTOR_DEFAULT ),
bSndCrdConversionBufferRequired ( false ),
iSndCardMonoBlockSizeSamConvBuff ( 0 ),
fAudioInputPeak ( 0 ),
fAudioOutputPeak ( 0 ),
bFraSiFactPrefSupported ( false ),
bFraSiFactDefSupported ( false ),
bFraSiFactSafeSupported ( false ),
Expand Down Expand Up @@ -1024,13 +1027,30 @@ void CClient::ProcessAudioDataIntern ( CVector<float>& vecfStereoSndCrd )

for ( i = 0, j = 0; i < iMonoBlockSizeSam; i++, j += 2 )
{
// clip samples for stereo pan mode
vecfStereoSndCrd[i] = ClipFloat (
fGainL * vecfStereoSndCrd[j] + fGainR * vecfStereoSndCrd[j + 1] );
vecfStereoSndCrd[i] =
fGainL * vecfStereoSndCrd[j] + fGainR * vecfStereoSndCrd[j + 1];
}
}
}

// apply local audio input compressor
if ( eAudioChannelConf == CC_STEREO )
{
for ( i = 0; i < iMonoBlockSizeSam; i++ )
{
stereoCompressor(SYSTEM_SAMPLE_RATE_HZ, fAudioInputPeak,
vecfStereoSndCrd[2 * i], vecfStereoSndCrd[2 * i + 1]);
}
}
else
{
for ( i = 0; i < iMonoBlockSizeSam; i++ )
{
monoCompressor(SYSTEM_SAMPLE_RATE_HZ, fAudioInputPeak,
vecfStereoSndCrd[i]);
}
}

// Support for mono-in/stereo-out mode: Per definition this mode works in
// full stereo mode at the transmission level. The only thing which is done
// is to mix both sound card inputs together and then put this signal on
Expand Down Expand Up @@ -1122,8 +1142,8 @@ void CClient::ProcessAudioDataIntern ( CVector<float>& vecfStereoSndCrd )
{
for ( i = 0; i < iStereoBlockSizeSam; i++ )
{
vecfStereoSndCrd[i] = ClipFloat (
vecfStereoSndCrd[i] + vecfStereoSndCrdMuteStream[i] * fMuteOutStreamGain );
vecfStereoSndCrd[i] =
vecfStereoSndCrd[i] + vecfStereoSndCrdMuteStream[i] * fMuteOutStreamGain;
}
}

Expand All @@ -1147,6 +1167,13 @@ void CClient::ProcessAudioDataIntern ( CVector<float>& vecfStereoSndCrd )
vecfStereoSndCrd.Reset ( 0 );
}

// apply local audio output compressor
for ( i = 0; i < iMonoBlockSizeSam; i++ )
{
stereoCompressor(SYSTEM_SAMPLE_RATE_HZ, fAudioOutputPeak,
vecfStereoSndCrd[2 * i], vecfStereoSndCrd[2 * i + 1]);
}

// update socket buffer size
Channel.UpdateSocketBufferSize();

Expand Down
2 changes: 2 additions & 0 deletions src/client.h
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Original file line number Diff line number Diff line change
Expand Up @@ -344,6 +344,8 @@ class CClient : public QObject
CVector<float> vecDataConvBuf;
CVector<float> vecfStereoSndCrdMuteStream;
CVector<float> vecZeros;
float fAudioInputPeak;
float fAudioOutputPeak;

bool bFraSiFactPrefSupported;
bool bFraSiFactDefSupported;
Expand Down
91 changes: 91 additions & 0 deletions src/compressor.h
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Original file line number Diff line number Diff line change
@@ -0,0 +1,91 @@
/*
ZynAddSubFX - a software synthesizer

Compressor.h - simple audio compressor macros
Copyright (C) 2016 Hans Petter Selasky

This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
*/

#pragma once

static inline bool
floatIsValid(const float x)
{
const float __r = x * 0.0f;
return (__r == 0.0f || __r == -0.0f);
}

#define stereoCompressor(div,pv,l,r) do { \
/* \
* Don't max the output range to avoid \
* overflowing sample rate conversion and \
* equalizer filters in the DSP's output \
* path. Keep one 10th, 1dB, reserved. \
*/ \
static constexpr float __limit = \
1.0f - (1.0f / 10.0f); \
float __peak; \
\
/* sanity checks */ \
__peak = (pv); \
if (!floatIsValid(__peak)) \
__peak = 0.0; \
if (!floatIsValid(l)) \
(l) = 0.0; \
if (!floatIsValid(r)) \
(r) = 0.0; \
/* compute maximum */ \
if ((l) < -__peak) \
__peak = -(l); \
else if ((l) > __peak) \
__peak = (l); \
if ((r) < -__peak) \
__peak = -(r); \
else if ((r) > __peak) \
__peak = (r); \
/* compressor */ \
if (__peak > __limit) { \
(l) /= __peak; \
(r) /= __peak; \
(l) *= __limit; \
(r) *= __limit; \
__peak -= __peak / (div); \
} \
(pv) = __peak; \
} while (0)

#define monoCompressor(div,pv,l) do { \
/* \
* Don't max the output range to avoid \
* overflowing sample rate conversion and \
* equalizer filters in the DSP's output \
* path. Keep one 10th, 1dB, reserved. \
*/ \
static constexpr float __limit = \
1.0f - (1.0f / 10.0f); \
float __peak; \
\
/* sanity checks */ \
__peak = (pv); \
if (!floatIsValid(__peak)) \
__peak = 0.0; \
if (!floatIsValid(l)) \
(l) = 0.0; \
/* compute maximum */ \
if ((l) < -__peak) \
__peak = -(l); \
else if ((l) > __peak) \
__peak = (l); \
/* compressor */ \
if (__peak > __limit) { \
(l) /= __peak; \
(l) *= __limit; \
__peak -= __peak / (div); \
} \
(pv) = __peak; \
} while (0)

26 changes: 10 additions & 16 deletions src/server.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -24,6 +24,7 @@

#include "server.h"

#include "compressor.h"

// CHighPrecisionTimer implementation ******************************************
#ifdef _WIN32
Expand Down Expand Up @@ -339,8 +340,8 @@ CServer::CServer ( const int iNewMaxNumChan,
vecvecfGains.Init ( iMaxNumChannels );
vecvecfPannings.Init ( iMaxNumChannels );
vecvecfData.Init ( iMaxNumChannels );
vecvecfSendData.Init ( iMaxNumChannels );
vecvecfIntermediateProcBuf.Init ( iMaxNumChannels );
vecfOutputAudioPeak.Init ( iMaxNumChannels, 0 );
vecvecbyCodedData.Init ( iMaxNumChannels );
vecNumAudioChannels.Init ( iMaxNumChannels );
vecNumFrameSizeConvBlocks.Init ( iMaxNumChannels );
Expand All @@ -356,10 +357,6 @@ CServer::CServer ( const int iNewMaxNumChan,
// we always use stereo audio buffers (which is the worst case)
vecvecfData[i].Init ( 2 /* stereo */ * DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES /* worst case buffer size */ );

// (note that we only allocate iMaxNumChannels buffers for the send
// and coded data because of the OMP implementation)
vecvecfSendData[i].Init ( 2 /* stereo */ * DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES /* worst case buffer size */ );

// allocate worst case memory for intermediate processing buffers in float precision
vecvecfIntermediateProcBuf[i].Init ( 2 /* stereo */ * DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES /* worst case buffer size */ );

Expand Down Expand Up @@ -1120,7 +1117,7 @@ void CServer::MixEncodeTransmitData ( const int iChanCnt,
{
int i, j, k, iUnused;
CVector<float>& vecfIntermProcBuf = vecvecfIntermediateProcBuf[iChanCnt]; // use reference for faster access
CVector<float>& vecfSendData = vecvecfSendData[iChanCnt]; // use reference for faster access
float &fAudioPeak = vecfOutputAudioPeak[iChanCnt];

// get actual ID of current channel
const int iCurChanID = vecChanIDsCurConChan[iChanCnt];
Expand Down Expand Up @@ -1183,11 +1180,9 @@ void CServer::MixEncodeTransmitData ( const int iChanCnt,
// When adding multiple sound sources together
// the resulting signal level may exceed the maximum
// audio range which is from -1.0f to 1.0f inclusivly.
// Clip the intermediate sound buffer to be within
// the expected range
for ( i = 0; i < iServerFrameSizeSamples; i++ )
{
vecfSendData[i] = ClipFloat ( vecfIntermProcBuf[i] );
monoCompressor(SYSTEM_SAMPLE_RATE_HZ, fAudioPeak, vecfIntermProcBuf[i]);
}
}
else
Expand Down Expand Up @@ -1255,11 +1250,10 @@ void CServer::MixEncodeTransmitData ( const int iChanCnt,
// When adding multiple sound sources together
// the resulting signal level may exceed the maximum
// audio range which is from -1.0f to 1.0f inclusivly.
// Clip the intermediate sound buffer to be within
// the expected range
for ( i = 0; i < ( 2 * iServerFrameSizeSamples ); i++ )
for ( i = 0; i < iServerFrameSizeSamples; i++ )
{
vecfSendData[i] = ClipFloat ( vecfIntermProcBuf[i] );
stereoCompressor(SYSTEM_SAMPLE_RATE_HZ, fAudioPeak,
vecfIntermProcBuf[2 * i], vecfIntermProcBuf[2 * i + 1]);
}
}

Expand Down Expand Up @@ -1303,12 +1297,12 @@ void CServer::MixEncodeTransmitData ( const int iChanCnt,
// is false and the Get() function is not called at all. Therefore if the buffer is not needed
// we do not spend any time in the function but go directly inside the if condition.
if ( ( vecUseDoubleSysFraSizeConvBuf[iChanCnt] == 0 ) ||
DoubleFrameSizeConvBufOut[iCurChanID].Put ( vecfSendData, SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[iChanCnt] ) )
DoubleFrameSizeConvBufOut[iCurChanID].Put ( vecfIntermProcBuf, SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[iChanCnt] ) )
{
if ( vecUseDoubleSysFraSizeConvBuf[iChanCnt] != 0 )
{
// get the large frame from the conversion buffer
DoubleFrameSizeConvBufOut[iCurChanID].GetAll ( vecfSendData, DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[iChanCnt] );
DoubleFrameSizeConvBufOut[iCurChanID].GetAll ( vecfIntermProcBuf, DOUBLE_SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[iChanCnt] );
}

for ( int iB = 0; iB < vecNumFrameSizeConvBlocks[iChanCnt]; iB++ )
Expand All @@ -1321,7 +1315,7 @@ void CServer::MixEncodeTransmitData ( const int iChanCnt,
opus_custom_encoder_ctl ( pCurOpusEncoder, OPUS_SET_BITRATE ( CalcBitRateBitsPerSecFromCodedBytes ( iCeltNumCodedBytes, iClientFrameSizeSamples ) ) );

iUnused = opus_custom_encode_float ( pCurOpusEncoder,
&vecfSendData[iB * SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[iChanCnt]],
&vecfIntermProcBuf[iB * SYSTEM_FRAME_SIZE_SAMPLES * vecNumAudioChannels[iChanCnt]],
iClientFrameSizeSamples,
&vecvecbyCodedData[iChanCnt][0],
iCeltNumCodedBytes );
Expand Down
2 changes: 1 addition & 1 deletion src/server.h
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Original file line number Diff line number Diff line change
Expand Up @@ -361,8 +361,8 @@ class CServer :
CVector<int> vecNumFrameSizeConvBlocks;
CVector<int> vecUseDoubleSysFraSizeConvBuf;
CVector<EAudComprType> vecAudioComprType;
CVector<CVector<float> > vecvecfSendData;
CVector<CVector<float> > vecvecfIntermediateProcBuf;
CVector<float> vecfOutputAudioPeak;
CVector<CVector<uint8_t> > vecvecbyCodedData;

// Channel levels
Expand Down