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ann0seeann0see
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Merge pull request #3403 from ann0see/refactoring/moveReverb
2 parents 7258a55 + 8f3a955 commit d723bfc

6 files changed

Lines changed: 269 additions & 247 deletions

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‎Jamulus.pro‎

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Original file line numberDiff line numberDiff line change
@@ -378,7 +378,8 @@ FORMS_GUI = src/aboutdlgbase.ui \
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src/connectdlgbase.ui
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}
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HEADERS += src/buffer.h \
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HEADERS += src/plugins/audioreverb.h \
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src/buffer.h \
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src/channel.h \
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src/global.h \
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src/protocol.h \
@@ -486,7 +487,8 @@ HEADERS_OPUS_X86 = libs/opus/celt/x86/celt_lpc_sse.h \
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libs/opus/celt/x86/x86cpu.h \
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$$files(libs/opus/silk/x86/*.h)
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SOURCES += src/buffer.cpp \
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SOURCES += src/plugins/audioreverb.cpp \
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src/buffer.cpp \
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src/channel.cpp \
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src/main.cpp \
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src/protocol.cpp \

‎src/client.h‎

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@@ -38,6 +38,7 @@
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#include "socket.h"
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#include "channel.h"
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#include "util.h"
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#include "plugins/audioreverb.h"
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#include "buffer.h"
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#include "signalhandler.h"
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‎src/plugins/audioreverb.cpp‎

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/******************************************************************************\
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* Audio Reverberation *
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\******************************************************************************/
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/*
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The following code is based on "JCRev: John Chowning's reverberator class"
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by Perry R. Cook and Gary P. Scavone, 1995 - 2004
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which is in "The Synthesis ToolKit in C++ (STK)"
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http://ccrma.stanford.edu/software/stk
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Original description:
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This class is derived from the CLM JCRev function, which is based on the use
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of networks of simple allpass and comb delay filters. This class implements
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three series allpass units, followed by four parallel comb filters, and two
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decorrelation delay lines in parallel at the output.
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*/
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#include "audioreverb.h"
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void CAudioReverb::Init ( const EAudChanConf eNAudioChannelConf, const int iNStereoBlockSizeSam, const int iSampleRate, const float fT60 )
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{
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// store parameters
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eAudioChannelConf = eNAudioChannelConf;
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iStereoBlockSizeSam = iNStereoBlockSizeSam;
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// delay lengths for 44100 Hz sample rate
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int lengths[9] = { 1116, 1356, 1422, 1617, 225, 341, 441, 211, 179 };
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const float scaler = static_cast<float> ( iSampleRate ) / 44100.0f;
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if ( scaler != 1.0f )
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{
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for ( int i = 0; i < 9; i++ )
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{
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int delay = static_cast<int> ( floorf ( scaler * lengths[i] ) );
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if ( ( delay & 1 ) == 0 )
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{
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delay++;
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}
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while ( !isPrime ( delay ) )
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{
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delay += 2;
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}
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lengths[i] = delay;
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}
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}
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for ( int i = 0; i < 3; i++ )
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{
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allpassDelays[i].Init ( lengths[i + 4] );
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}
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for ( int i = 0; i < 4; i++ )
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{
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combDelays[i].Init ( lengths[i] );
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combFilters[i].setPole ( 0.2f );
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}
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setT60 ( fT60, iSampleRate );
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outLeftDelay.Init ( lengths[7] );
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outRightDelay.Init ( lengths[8] );
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allpassCoefficient = 0.7f;
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Clear();
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}
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bool CAudioReverb::isPrime ( const int number )
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{
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/*
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Returns true if argument value is prime. Taken from "class Effect" in
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"STK abstract effects parent class".
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*/
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if ( number == 2 )
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{
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return true;
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}
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if ( number & 1 )
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{
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for ( int i = 3; i < static_cast<int> ( sqrtf ( static_cast<float> ( number ) ) ) + 1; i += 2 )
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{
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if ( ( number % i ) == 0 )
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{
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return false;
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}
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}
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return true; // prime
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}
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else
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{
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return false; // even
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}
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}
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void CAudioReverb::Clear()
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{
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// reset and clear all internal state
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allpassDelays[0].Reset ( 0 );
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allpassDelays[1].Reset ( 0 );
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allpassDelays[2].Reset ( 0 );
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combDelays[0].Reset ( 0 );
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combDelays[1].Reset ( 0 );
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combDelays[2].Reset ( 0 );
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combDelays[3].Reset ( 0 );
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combFilters[0].Reset();
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combFilters[1].Reset();
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combFilters[2].Reset();
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combFilters[3].Reset();
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outRightDelay.Reset ( 0 );
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outLeftDelay.Reset ( 0 );
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}
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void CAudioReverb::setT60 ( const float fT60, const int iSampleRate )
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{
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// set the reverberation T60 decay time
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for ( int i = 0; i < 4; i++ )
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{
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combCoefficient[i] = powf ( 10.0f, static_cast<float> ( -3.0f * combDelays[i].Size() / ( fT60 * iSampleRate ) ) );
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}
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}
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void CAudioReverb::COnePole::setPole ( const float fPole )
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{
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// calculate IIR filter coefficients based on the pole value
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fA = -fPole;
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fB = 1.0f - fPole;
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}
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float CAudioReverb::COnePole::Calc ( const float fIn )
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{
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// calculate IIR filter
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fLastSample = fB * fIn - fA * fLastSample;
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return fLastSample;
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}
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void CAudioReverb::Process ( CVector<int16_t>& vecsStereoInOut, const bool bReverbOnLeftChan, const float fAttenuation )
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{
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float fMixedInput, temp, temp0, temp1, temp2;
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for ( int i = 0; i < iStereoBlockSizeSam; i += 2 )
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{
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// we sum up the stereo input channels (in case mono input is used, a zero
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// shall be input for the right channel)
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if ( eAudioChannelConf == CC_STEREO )
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{
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fMixedInput = 0.5f * ( vecsStereoInOut[i] + vecsStereoInOut[i + 1] );
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}
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else
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{
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if ( bReverbOnLeftChan )
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{
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fMixedInput = vecsStereoInOut[i];
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}
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else
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{
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fMixedInput = vecsStereoInOut[i + 1];
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}
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}
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temp = allpassDelays[0].Get();
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temp0 = allpassCoefficient * temp;
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temp0 += fMixedInput;
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allpassDelays[0].Add ( temp0 );
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temp0 = -( allpassCoefficient * temp0 ) + temp;
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temp = allpassDelays[1].Get();
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temp1 = allpassCoefficient * temp;
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temp1 += temp0;
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allpassDelays[1].Add ( temp1 );
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temp1 = -( allpassCoefficient * temp1 ) + temp;
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temp = allpassDelays[2].Get();
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temp2 = allpassCoefficient * temp;
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temp2 += temp1;
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allpassDelays[2].Add ( temp2 );
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temp2 = -( allpassCoefficient * temp2 ) + temp;
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const float temp3 = temp2 + combFilters[0].Calc ( combCoefficient[0] * combDelays[0].Get() );
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const float temp4 = temp2 + combFilters[1].Calc ( combCoefficient[1] * combDelays[1].Get() );
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const float temp5 = temp2 + combFilters[2].Calc ( combCoefficient[2] * combDelays[2].Get() );
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const float temp6 = temp2 + combFilters[3].Calc ( combCoefficient[3] * combDelays[3].Get() );
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combDelays[0].Add ( temp3 );
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combDelays[1].Add ( temp4 );
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combDelays[2].Add ( temp5 );
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combDelays[3].Add ( temp6 );
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const float filtout = temp3 + temp4 + temp5 + temp6;
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outLeftDelay.Add ( filtout );
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outRightDelay.Add ( filtout );
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// inplace apply the attenuated reverb signal (for stereo always apply
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// reverberation effect on both channels)
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if ( ( eAudioChannelConf == CC_STEREO ) || bReverbOnLeftChan )
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{
199+
vecsStereoInOut[i] = Float2Short ( ( 1.0f - fAttenuation ) * vecsStereoInOut[i] + 0.5f * fAttenuation * outLeftDelay.Get() );
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}
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if ( ( eAudioChannelConf == CC_STEREO ) || !bReverbOnLeftChan )
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{
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vecsStereoInOut[i + 1] = Float2Short ( ( 1.0f - fAttenuation ) * vecsStereoInOut[i + 1] + 0.5f * fAttenuation * outRightDelay.Get() );
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}
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}
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}

‎src/plugins/audioreverb.h‎

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Original file line numberDiff line numberDiff line change
@@ -0,0 +1,57 @@
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/******************************************************************************\
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* Audio Reverberation *
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\******************************************************************************/
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/*
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The following code is based on "JCRev: John Chowning's reverberator class"
6+
by Perry R. Cook and Gary P. Scavone, 1995 - 2004
7+
which is in "The Synthesis ToolKit in C++ (STK)"
8+
http://ccrma.stanford.edu/software/stk
9+
10+
Original description:
11+
This class is derived from the CLM JCRev function, which is based on the use
12+
of networks of simple allpass and comb delay filters. This class implements
13+
three series allpass units, followed by four parallel comb filters, and two
14+
decorrelation delay lines in parallel at the output.
15+
*/
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17+
#pragma once
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#include "util.h"
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class CAudioReverb
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{
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public:
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CAudioReverb() {}
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void Init ( const EAudChanConf eNAudioChannelConf, const int iNStereoBlockSizeSam, const int iSampleRate, const float fT60 = 1.1f );
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void Clear();
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void Process ( CVector<int16_t>& vecsStereoInOut, const bool bReverbOnLeftChan, const float fAttenuation );
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protected:
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void setT60 ( const float fT60, const int iSampleRate );
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bool isPrime ( const int number );
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class COnePole
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{
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public:
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COnePole() : fA ( 0 ), fB ( 0 ) { Reset(); }
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void setPole ( const float fPole );
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float Calc ( const float fIn );
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void Reset() { fLastSample = 0; }
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protected:
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float fA;
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float fB;
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float fLastSample;
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};
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EAudChanConf eAudioChannelConf;
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int iStereoBlockSizeSam;
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CFIFO<float> allpassDelays[3];
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CFIFO<float> combDelays[4];
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COnePole combFilters[4];
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CFIFO<float> outLeftDelay;
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CFIFO<float> outRightDelay;
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float allpassCoefficient;
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float combCoefficient[4];
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};

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