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sdrangel/sdrbase/audio/audiocompressorsnd.h

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///////////////////////////////////////////////////////////////////////////////////
// Copyright (C) 2019 Edouard Griffiths, F4EXB <f4exb06@gmail.com> //
// Copyright (C) 2020 Kacper Michajłow <kasper93@gmail.com> //
// //
// Audio compressor based on sndfilter by Sean Connelly (@voidqk) //
// https://github.com/voidqk/sndfilter //
// //
// Sample by sample interface to facilitate integration in SDRangel modulators. //
// Uses mono samples (just floats) //
// //
// 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 as version 3 of the License, or //
// (at your option) any later version. //
// //
// This program is distributed in the hope that it will be useful, //
// but WITHOUT ANY WARRANTY; without even the implied warranty of //
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
// GNU General Public License V3 for more details. //
// //
// You should have received a copy of the GNU General Public License //
// along with this program. If not, see <http://www.gnu.org/licenses/>. //
///////////////////////////////////////////////////////////////////////////////////
#ifndef SDRBASE_AUDIO_AUDIOCOMPRESSORSND_H_
#define SDRBASE_AUDIO_AUDIOCOMPRESSORSND_H_
#include <cmath>
// maximum number of samples in the delay buffer
#define AUDIOCOMPRESSORSND_SF_COMPRESSOR_MAXDELAY 1024
// samples per update; the compressor works by dividing the input chunks into even smaller sizes,
// and performs heavier calculations after each mini-chunk to adjust the final envelope
#define AUDIOCOMPRESSORSND_SF_COMPRESSOR_SPU 32
// not sure what this does exactly, but it is part of the release curve
#define AUDIOCOMPRESSORSND_SF_COMPRESSOR_SPACINGDB 5.0f
// the "chunk" size as defined in original sndfilter library
#define AUDIOCOMPRESSORSND_SF_COMPRESSOR_CHUNKSIZE 128
#include "export.h"
class SDRBASE_API AudioCompressorSnd
{
public:
AudioCompressorSnd();
~AudioCompressorSnd();
void initDefault(int rate)
{
m_rate = rate;
m_pregain = 0.000f;
m_threshold = -24.000f;
m_knee = 30.000f;
m_ratio = 12.000f;
m_attack = 0.003f;
m_release = 0.250f;
m_predelay = 0.006f;
m_releasezone1 = 0.090f;
m_releasezone2 = 0.160f;
m_releasezone3 = 0.420f;
m_releasezone4 = 0.980f;
m_postgain = 0.000f;
m_wet = 1.000f;
initState();
}
void initSimple(
int rate, // input sample rate (samples per second)
float pregain, // dB, amount to boost the signal before applying compression [0 to 100]
float threshold, // dB, level where compression kicks in [-100 to 0]
float knee, // dB, width of the knee [0 to 40]
float ratio, // unitless, amount to inversely scale the output when applying comp [1 to 20]
float attack, // seconds, length of the attack phase [0 to 1]
float release // seconds, length of the release phase [0 to 1]
)
{
m_rate = rate;
m_pregain = pregain;
m_threshold = threshold;
m_knee = knee;
m_ratio = ratio;
m_attack = attack;
m_release = release;
m_predelay = 0.006f;
m_releasezone1 = 0.090f;
m_releasezone2 = 0.160f;
m_releasezone3 = 0.420f;
m_releasezone4 = 0.980f;
m_postgain = 0.000f;
m_wet = 1.000f;
initState();
}
void initState();
float compress(float sample);
float m_rate;
float m_pregain;
float m_threshold;
float m_knee;
float m_ratio;
float m_attack;
float m_release;
float m_predelay;
float m_releasezone1;
float m_releasezone2;
float m_releasezone3;
float m_releasezone4;
float m_postgain;
float m_wet;
private:
static inline float db2lin(float db){ // dB to linear
return powf(10.0f, 0.05f * db);
}
static inline float lin2db(float lin){ // linear to dB
return 20.0f * log10f(lin);
}
// for more information on the knee curve, check out the compressor-curve.html demo + source code
// included in this repo
static inline float kneecurve(float x, float k, float linearthreshold){
return linearthreshold + (1.0f - expf(-k * (x - linearthreshold))) / k;
}
static inline float kneeslope(float x, float k, float linearthreshold){
return k * x / ((k * linearthreshold + 1.0f) * expf(k * (x - linearthreshold)) - 1);
}
static inline float compcurve(float x, float k, float slope, float linearthreshold,
float linearthresholdknee, float threshold, float knee, float kneedboffset){
if (x < linearthreshold)
return x;
if (knee <= 0.0f) // no knee in curve
return db2lin(threshold + slope * (lin2db(x) - threshold));
if (x < linearthresholdknee)
return kneecurve(x, k, linearthreshold);
return db2lin(kneedboffset + slope * (lin2db(x) - threshold - knee));
}
// for more information on the adaptive release curve, check out adaptive-release-curve.html demo +
// source code included in this repo
static inline float adaptivereleasecurve(float x, float a, float b, float c, float d){
// a*x^3 + b*x^2 + c*x + d
float x2 = x * x;
return a * x2 * x + b * x2 + c * x + d;
}
static inline float clampf(float v, float min, float max){
return v < min ? min : (v > max ? max : v);
}
static inline float absf(float v){
return v < 0.0f ? -v : v;
}
static inline float fixf(float v, float def){
// fix NaN and infinity values that sneak in... not sure why this is needed, but it is
if (std::isnan(v) || std::isinf(v))
return def;
return v;
}
struct CompressorState
{ // sf_compressor_state_st
// user can read the metergain state variable after processing a chunk to see how much dB the
// compressor would have liked to compress the sample; the meter values aren't used to shape the
// sound in any way, only used for output if desired
float metergain;
// everything else shouldn't really be mucked with unless you read the algorithm and feel
// comfortable
float meterrelease;
float threshold;
float knee;
float linearpregain;
float linearthreshold;
float slope;
float attacksamplesinv;
float satreleasesamplesinv;
float wet;
float dry;
float k;
float kneedboffset;
float linearthresholdknee;
float mastergain;
float a; // adaptive release polynomial coefficients
float b;
float c;
float d;
float detectoravg;
float compgain;
float maxcompdiffdb;
int delaybufsize;
int delaywritepos;
int delayreadpos;
float delaybuf[AUDIOCOMPRESSORSND_SF_COMPRESSOR_MAXDELAY]; // predelay buffer
// populate the compressor state with advanced parameters
void sf_advancecomp(
// these parameters are the same as the simple version above:
int rate, float pregain, float threshold, float knee, float ratio, float attack, float release,
// these are the advanced parameters:
float predelay, // seconds, length of the predelay buffer [0 to 1]
float releasezone1, // release zones should be increasing between 0 and 1, and are a fraction
float releasezone2, // of the release time depending on the input dB -- these parameters define
float releasezone3, // the adaptive release curve, which is discussed in further detail in the
float releasezone4, // demo: adaptive-release-curve.html
float postgain, // dB, amount of gain to apply after compression [0 to 100]
float wet // amount to apply the effect [0 completely dry to 1 completely wet]
);
};
static void sf_compressor_process(CompressorState *state, int size, float *input, float *output);
CompressorState m_compressorState;
float m_storageBuffer[AUDIOCOMPRESSORSND_SF_COMPRESSOR_CHUNKSIZE];
float m_processedBuffer[AUDIOCOMPRESSORSND_SF_COMPRESSOR_CHUNKSIZE];
int m_sampleIndex;
};
#endif // SDRBASE_AUDIO_AUDIOCOMPRESSORSND_H_
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