Move reverb code out of utils

Jamulus.pro

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

src/client.h

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

src/plugins/audioreverb.cpp

@@ -0,0 +1,207 @@
+/******************************************************************************\
+* Audio Reverberation                                                          *
+\******************************************************************************/
+/*
+    The following code is based on "JCRev: John Chowning's reverberator class"
+    by Perry R. Cook and Gary P. Scavone, 1995 - 2004
+    which is in "The Synthesis ToolKit in C++ (STK)"
+    http://ccrma.stanford.edu/software/stk
+
+    Original description:
+    This class is derived from the CLM JCRev function, which is based on the use
+    of networks of simple allpass and comb delay filters. This class implements
+    three series allpass units, followed by four parallel comb filters, and two
+    decorrelation delay lines in parallel at the output.
+*/
+
+#include "audioreverb.h"
+
+void CAudioReverb::Init ( const EAudChanConf eNAudioChannelConf, const int iNStereoBlockSizeSam, const int iSampleRate, const float fT60 )
+{
+    // store parameters
+    eAudioChannelConf   = eNAudioChannelConf;
+    iStereoBlockSizeSam = iNStereoBlockSizeSam;
+
+    // delay lengths for 44100 Hz sample rate
+    int         lengths[9] = { 1116, 1356, 1422, 1617, 225, 341, 441, 211, 179 };
+    const float scaler     = static_cast<float> ( iSampleRate ) / 44100.0f;
+
+    if ( scaler != 1.0f )
+    {
+        for ( int i = 0; i < 9; i++ )
+        {
+            int delay = static_cast<int> ( floorf ( scaler * lengths[i] ) );
+
+            if ( ( delay & 1 ) == 0 )
+            {
+                delay++;
+            }
+
+            while ( !isPrime ( delay ) )
+            {
+                delay += 2;
+            }
+
+            lengths[i] = delay;
+        }
+    }
+
+    for ( int i = 0; i < 3; i++ )
+    {
+        allpassDelays[i].Init ( lengths[i + 4] );
+    }
+
+    for ( int i = 0; i < 4; i++ )
+    {
+        combDelays[i].Init ( lengths[i] );
+        combFilters[i].setPole ( 0.2f );
+    }
+
+    setT60 ( fT60, iSampleRate );
+    outLeftDelay.Init ( lengths[7] );
+    outRightDelay.Init ( lengths[8] );
+    allpassCoefficient = 0.7f;
+    Clear();
+}
+
+bool CAudioReverb::isPrime ( const int number )
+{
+    /*
+        Returns true if argument value is prime. Taken from "class Effect" in
+        "STK abstract effects parent class".
+    */
+    if ( number == 2 )
+    {
+        return true;
+    }
+
+    if ( number & 1 )
+    {
+        for ( int i = 3; i < static_cast<int> ( sqrtf ( static_cast<float> ( number ) ) ) + 1; i += 2 )
+        {
+            if ( ( number % i ) == 0 )
+            {
+                return false;
+            }
+        }
+
+        return true; // prime
+    }
+    else
+    {
+        return false; // even
+    }
+}
+
+void CAudioReverb::Clear()
+{
+    // reset and clear all internal state
+    allpassDelays[0].Reset ( 0 );
+    allpassDelays[1].Reset ( 0 );
+    allpassDelays[2].Reset ( 0 );
+    combDelays[0].Reset ( 0 );
+    combDelays[1].Reset ( 0 );
+    combDelays[2].Reset ( 0 );
+    combDelays[3].Reset ( 0 );
+    combFilters[0].Reset();
+    combFilters[1].Reset();
+    combFilters[2].Reset();
+    combFilters[3].Reset();
+    outRightDelay.Reset ( 0 );
+    outLeftDelay.Reset ( 0 );
+}
+
+void CAudioReverb::setT60 ( const float fT60, const int iSampleRate )
+{
+    // set the reverberation T60 decay time
+    for ( int i = 0; i < 4; i++ )
+    {
+        combCoefficient[i] = powf ( 10.0f, static_cast<float> ( -3.0f * combDelays[i].Size() / ( fT60 * iSampleRate ) ) );
+    }
+}
+
+void CAudioReverb::COnePole::setPole ( const float fPole )
+{
+    // calculate IIR filter coefficients based on the pole value
+    fA = -fPole;
+    fB = 1.0f - fPole;
+}
+
+float CAudioReverb::COnePole::Calc ( const float fIn )
+{
+    // calculate IIR filter
+    fLastSample = fB * fIn - fA * fLastSample;
+
+    return fLastSample;
+}
+
+void CAudioReverb::Process ( CVector<int16_t>& vecsStereoInOut, const bool bReverbOnLeftChan, const float fAttenuation )
+{
+    float fMixedInput, temp, temp0, temp1, temp2;
+
+    for ( int i = 0; i < iStereoBlockSizeSam; i += 2 )
+    {
+        // we sum up the stereo input channels (in case mono input is used, a zero
+        // shall be input for the right channel)
+        if ( eAudioChannelConf == CC_STEREO )
+        {
+            fMixedInput = 0.5f * ( vecsStereoInOut[i] + vecsStereoInOut[i + 1] );
+        }
+        else
+        {
+            if ( bReverbOnLeftChan )
+            {
+                fMixedInput = vecsStereoInOut[i];
+            }
+            else
+            {
+                fMixedInput = vecsStereoInOut[i + 1];
+            }
+        }
+
+        temp  = allpassDelays[0].Get();
+        temp0 = allpassCoefficient * temp;
+        temp0 += fMixedInput;
+        allpassDelays[0].Add ( temp0 );
+        temp0 = -( allpassCoefficient * temp0 ) + temp;
+
+        temp  = allpassDelays[1].Get();
+        temp1 = allpassCoefficient * temp;
+        temp1 += temp0;
+        allpassDelays[1].Add ( temp1 );
+        temp1 = -( allpassCoefficient * temp1 ) + temp;
+
+        temp  = allpassDelays[2].Get();
+        temp2 = allpassCoefficient * temp;
+        temp2 += temp1;
+        allpassDelays[2].Add ( temp2 );
+        temp2 = -( allpassCoefficient * temp2 ) + temp;
+
+        const float temp3 = temp2 + combFilters[0].Calc ( combCoefficient[0] * combDelays[0].Get() );
+        const float temp4 = temp2 + combFilters[1].Calc ( combCoefficient[1] * combDelays[1].Get() );
+        const float temp5 = temp2 + combFilters[2].Calc ( combCoefficient[2] * combDelays[2].Get() );
+        const float temp6 = temp2 + combFilters[3].Calc ( combCoefficient[3] * combDelays[3].Get() );
+
+        combDelays[0].Add ( temp3 );
+        combDelays[1].Add ( temp4 );
+        combDelays[2].Add ( temp5 );
+        combDelays[3].Add ( temp6 );
+
+        const float filtout = temp3 + temp4 + temp5 + temp6;
+
+        outLeftDelay.Add ( filtout );
+        outRightDelay.Add ( filtout );
+
+        // inplace apply the attenuated reverb signal (for stereo always apply
+        // reverberation effect on both channels)
+        if ( ( eAudioChannelConf == CC_STEREO ) || bReverbOnLeftChan )
+        {
+            vecsStereoInOut[i] = Float2Short ( ( 1.0f - fAttenuation ) * vecsStereoInOut[i] + 0.5f * fAttenuation * outLeftDelay.Get() );
+        }
+
+        if ( ( eAudioChannelConf == CC_STEREO ) || !bReverbOnLeftChan )
+        {
+            vecsStereoInOut[i + 1] = Float2Short ( ( 1.0f - fAttenuation ) * vecsStereoInOut[i + 1] + 0.5f * fAttenuation * outRightDelay.Get() );
+        }
+    }
+}

src/plugins/audioreverb.h

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