/usr/include/stk/FreeVerb.h is in libstk0-dev 4.5.0-3.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 | #ifndef STK_FREEVERB_H
#define STK_FREEVERB_H
#include "Effect.h"
#include "Delay.h"
#include "OnePole.h"
namespace stk {
/***********************************************************************/
/*! \class FreeVerb
\brief Jezar at Dreampoint's FreeVerb, implemented in STK.
Freeverb is a free and open-source Schroeder reverberator
originally implemented in C++. The parameters of the reverberation
model are exceptionally well tuned. FreeVerb uses 8
lowpass-feedback-comb-filters in parallel, followed by 4 Schroeder
allpass filters in series. The input signal can be either mono or
stereo, and the output signal is stereo. The delay lengths are
optimized for a sample rate of 44100 Hz.
Ported to STK by Gregory Burlet, 2012.
*/
/***********************************************************************/
class FreeVerb : public Effect
{
public:
//! FreeVerb Constructor
/*!
Initializes the effect with default parameters. Note that these defaults
are slightly different than those in the original implementation of
FreeVerb [Effect Mix: 0.75; Room Size: 0.75; Damping: 0.25; Width: 1.0;
Mode: freeze mode off].
*/
FreeVerb();
//! Destructor
~FreeVerb();
//! Set the effect mix [0 = mostly dry, 1 = mostly wet].
void setEffectMix( StkFloat mix );
//! Set the room size (comb filter feedback gain) parameter [0,1].
void setRoomSize( StkFloat value );
//! Get the room size (comb filter feedback gain) parameter.
StkFloat getRoomSize( void );
//! Set the damping parameter [0=low damping, 1=higher damping].
void setDamping( StkFloat value );
//! Get the damping parameter.
StkFloat getDamping( void );
//! Set the width (left-right mixing) parameter [0,1].
void setWidth( StkFloat value );
//! Get the width (left-right mixing) parameter.
StkFloat getWidth( void );
//! Set the mode [frozen = 1, unfrozen = 0].
void setMode( bool isFrozen );
//! Get the current freeze mode [frozen = 1, unfrozen = 0].
StkFloat getMode( void );
//! Clears delay lines, etc.
void clear( void );
//! Return the specified channel value of the last computed stereo frame.
/*!
Use the lastFrame() function to get both values of the last
computed stereo frame. The \c channel argument must be 0 or 1
(the first channel is specified by 0). However, range checking is
only performed if _STK_DEBUG_ is defined during compilation, in
which case an out-of-range value will trigger an StkError
exception.
*/
StkFloat lastOut( unsigned int channel = 0 );
//! Input one or two samples to the effect and return the specified \c channel value of the computed stereo frame.
/*!
Use the lastFrame() function to get both values of the computed
stereo output frame. The \c channel argument must be 0 or 1 (the
first channel is specified by 0). However, range checking is only
performed if _STK_DEBUG_ is defined during compilation, in which
case an out-of-range value will trigger an StkError exception.
*/
StkFloat tick( StkFloat inputL, StkFloat inputR = 0.0, unsigned int channel = 0 );
//! Take two channels of the StkFrames object as inputs to the effect and replace with stereo outputs.
/*!
The StkFrames argument reference is returned. The stereo
inputs are taken from (and written back to) the StkFrames argument
starting at the specified \c channel. Therefore, the \c channel
argument must be less than ( channels() - 1 ) of the StkFrames
argument (the first channel is specified by 0). However, range
checking is only performed if _STK_DEBUG_ is defined during
compilation, in which case an out-of-range value will trigger an
StkError exception.
*/
StkFrames& tick( StkFrames& frames, unsigned int channel = 0 );
//! Take one or two channels of the \c iFrames object as inputs to the effect and write stereo outputs to the \c oFrames object.
/*!
The \c iFrames object reference is returned. The \c iChannel
argument must be less than the number of channels in the \c
iFrames argument (the first channel is specified by 0). If more
than one channel of data exists in \c iFrames starting from \c
iChannel, stereo data is input to the effect. The \c oChannel
argument must be less than ( channels() - 1 ) of the \c oFrames
argument. However, range checking is only performed if
_STK_DEBUG_ is defined during compilation, in which case an
out-of-range value will trigger an StkError exception.
*/
StkFrames& tick( StkFrames& iFrames, StkFrames &oFrames, unsigned int iChannel = 0, unsigned int oChannel = 0 );
protected:
//! Update interdependent parameters.
void update( void );
// Clamp very small floats to zero, version from
// http://music.columbia.edu/pipermail/linux-audio-user/2004-July/013489.html .
// However, this is for 32-bit floats only.
//static inline StkFloat undenormalize( volatile StkFloat s ) {
// s += 9.8607615E-32f;
// return s - 9.8607615E-32f;
//}
static const int nCombs = 8;
static const int nAllpasses = 4;
static const int stereoSpread = 23;
static const StkFloat fixedGain;
static const StkFloat scaleWet;
static const StkFloat scaleDry;
static const StkFloat scaleDamp;
static const StkFloat scaleRoom;
static const StkFloat offsetRoom;
// Delay line lengths for 44100Hz sampling rate.
static int cDelayLengths[nCombs];
static int aDelayLengths[nAllpasses];
StkFloat g_; // allpass coefficient
StkFloat gain_;
StkFloat roomSizeMem_, roomSize_;
StkFloat dampMem_, damp_;
StkFloat wet1_, wet2_;
StkFloat dry_;
StkFloat width_;
bool frozenMode_;
// LBFC: Lowpass Feedback Comb Filters
Delay combDelayL_[nCombs];
Delay combDelayR_[nCombs];
OnePole combLPL_[nCombs];
OnePole combLPR_[nCombs];
// AP: Allpass Filters
Delay allPassDelayL_[nAllpasses];
Delay allPassDelayR_[nAllpasses];
};
inline StkFloat FreeVerb :: lastOut( unsigned int channel )
{
#if defined(_STK_DEBUG_)
if ( channel > 1 ) {
oStream_ << "FreeVerb::lastOut(): channel argument must be less than 2!";
handleError( StkError::FUNCTION_ARGUMENT );
}
#endif
return lastFrame_[channel];
}
inline StkFloat FreeVerb::tick( StkFloat inputL, StkFloat inputR, unsigned int channel )
{
#if defined(_STK_DEBUG_)
if ( channel > 1 ) {
oStream_ << "FreeVerb::tick(): channel argument must be less than 2!";
handleError(StkError::FUNCTION_ARGUMENT);
}
#endif
StkFloat fInput = (inputL + inputR) * gain_;
StkFloat outL = 0.0;
StkFloat outR = 0.0;
// Parallel LBCF filters
for ( int i = 0; i < nCombs; i++ ) {
// Left channel
//StkFloat yn = fInput + (roomSize_ * FreeVerb::undenormalize(combLPL_[i].tick(FreeVerb::undenormalize(combDelayL_[i].nextOut()))));
StkFloat yn = fInput + (roomSize_ * combLPL_[i].tick( combDelayL_[i].nextOut() ) );
combDelayL_[i].tick(yn);
outL += yn;
// Right channel
//yn = fInput + (roomSize_ * FreeVerb::undenormalize(combLPR_[i].tick(FreeVerb::undenormalize(combDelayR_[i].nextOut()))));
yn = fInput + (roomSize_ * combLPR_[i].tick( combDelayR_[i].nextOut() ) );
combDelayR_[i].tick(yn);
outR += yn;
}
// Series allpass filters
for ( int i = 0; i < nAllpasses; i++ ) {
// Left channel
//StkFloat vn_m = FreeVerb::undenormalize(allPassDelayL_[i].nextOut());
StkFloat vn_m = allPassDelayL_[i].nextOut();
StkFloat vn = outL + (g_ * vn_m);
allPassDelayL_[i].tick(vn);
// calculate output
outL = -vn + (1.0 + g_)*vn_m;
// Right channel
//vn_m = FreeVerb::undenormalize(allPassDelayR_[i].nextOut());
vn_m = allPassDelayR_[i].nextOut();
vn = outR + (g_ * vn_m);
allPassDelayR_[i].tick(vn);
// calculate output
outR = -vn + (1.0 + g_)*vn_m;
}
// Mix output
lastFrame_[0] = outL*wet1_ + outR*wet2_ + inputL*dry_;
lastFrame_[1] = outR*wet1_ + outL*wet2_ + inputR*dry_;
/*
// Hard limiter ... there's not much else we can do at this point
if ( lastFrame_[0] >= 1.0 ) {
lastFrame_[0] = 0.9999;
}
if ( lastFrame_[0] <= -1.0 ) {
lastFrame_[0] = -0.9999;
}
if ( lastFrame_[1] >= 1.0 ) {
lastFrame_[1] = 0.9999;
}
if ( lastFrame_[1] <= -1.0 ) {
lastFrame_[1] = -0.9999;
}
*/
return lastFrame_[channel];
}
}
#endif
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