mirror of
https://github.com/f4exb/sdrangel.git
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230 lines
8.9 KiB
C++
230 lines
8.9 KiB
C++
///////////////////////////////////////////////////////////////////////////////////
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// Copyright (C) 2019 F4EXB //
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// written by Edouard Griffiths //
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// //
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// Audio compressor based on sndfilter by Sean Connelly (@voidqk) //
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// https://github.com/voidqk/sndfilter //
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// //
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// Sample by sample interface to facilitate integration in SDRangel modulators. //
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// Uses mono samples (just floats) //
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// //
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// This program is free software; you can redistribute it and/or modify //
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// it under the terms of the GNU General Public License as published by //
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// the Free Software Foundation as version 3 of the License, or //
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// (at your option) any later version. //
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// //
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// This program is distributed in the hope that it will be useful, //
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// but WITHOUT ANY WARRANTY; without even the implied warranty of //
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the //
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// GNU General Public License V3 for more details. //
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// //
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// You should have received a copy of the GNU General Public License //
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// along with this program. If not, see <http://www.gnu.org/licenses/>. //
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///////////////////////////////////////////////////////////////////////////////////
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#ifndef SDRBASE_AUDIO_AUDIOCOMPRESSORSND_H_
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#define SDRBASE_AUDIO_AUDIOCOMPRESSORSND_H_
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#define _USE_MATH_DEFINES
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#include <math.h>
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// maximum number of samples in the delay buffer
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#define AUDIOCOMPRESSORSND_SF_COMPRESSOR_MAXDELAY 1024
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// samples per update; the compressor works by dividing the input chunks into even smaller sizes,
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// and performs heavier calculations after each mini-chunk to adjust the final envelope
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#define AUDIOCOMPRESSORSND_SF_COMPRESSOR_SPU 32
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// not sure what this does exactly, but it is part of the release curve
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#define AUDIOCOMPRESSORSND_SF_COMPRESSOR_SPACINGDB 5.0f
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// the "chunk" size as defined in original sndfilter library
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#define AUDIOCOMPRESSORSND_SF_COMPRESSOR_CHUNKSIZE 128
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#include "export.h"
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class SDRBASE_API AudioCompressorSnd
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{
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public:
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AudioCompressorSnd();
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~AudioCompressorSnd();
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void initDefault(int rate)
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{
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m_rate = rate;
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m_pregain = 0.000f;
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m_threshold = -24.000f;
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m_knee = 30.000f;
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m_ratio = 12.000f;
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m_attack = 0.003f;
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m_release = 0.250f;
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m_predelay = 0.006f;
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m_releasezone1 = 0.090f;
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m_releasezone2 = 0.160f;
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m_releasezone3 = 0.420f;
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m_releasezone4 = 0.980f;
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m_postgain = 0.000f;
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m_wet = 1.000f;
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initState();
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}
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void initSimple(
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int rate, // input sample rate (samples per second)
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float pregain, // dB, amount to boost the signal before applying compression [0 to 100]
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float threshold, // dB, level where compression kicks in [-100 to 0]
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float knee, // dB, width of the knee [0 to 40]
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float ratio, // unitless, amount to inversely scale the output when applying comp [1 to 20]
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float attack, // seconds, length of the attack phase [0 to 1]
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float release // seconds, length of the release phase [0 to 1]
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)
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{
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m_rate = rate;
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m_pregain = pregain;
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m_threshold = threshold;
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m_knee = knee;
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m_ratio = ratio;
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m_attack = attack;
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m_release = release;
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m_predelay = 0.006f;
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m_releasezone1 = 0.090f;
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m_releasezone2 = 0.160f;
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m_releasezone3 = 0.420f;
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m_releasezone4 = 0.980f;
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m_postgain = 0.000f;
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m_wet = 1.000f;
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initState();
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}
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void initState();
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float compress(float sample);
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float m_rate;
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float m_pregain;
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float m_threshold;
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float m_knee;
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float m_ratio;
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float m_attack;
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float m_release;
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float m_predelay;
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float m_releasezone1;
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float m_releasezone2;
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float m_releasezone3;
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float m_releasezone4;
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float m_postgain;
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float m_wet;
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private:
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static inline float db2lin(float db){ // dB to linear
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return powf(10.0f, 0.05f * db);
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}
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static inline float lin2db(float lin){ // linear to dB
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return 20.0f * log10f(lin);
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}
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// for more information on the knee curve, check out the compressor-curve.html demo + source code
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// included in this repo
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static inline float kneecurve(float x, float k, float linearthreshold){
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return linearthreshold + (1.0f - expf(-k * (x - linearthreshold))) / k;
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}
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static inline float kneeslope(float x, float k, float linearthreshold){
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return k * x / ((k * linearthreshold + 1.0f) * expf(k * (x - linearthreshold)) - 1);
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}
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static inline float compcurve(float x, float k, float slope, float linearthreshold,
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float linearthresholdknee, float threshold, float knee, float kneedboffset){
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if (x < linearthreshold)
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return x;
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if (knee <= 0.0f) // no knee in curve
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return db2lin(threshold + slope * (lin2db(x) - threshold));
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if (x < linearthresholdknee)
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return kneecurve(x, k, linearthreshold);
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return db2lin(kneedboffset + slope * (lin2db(x) - threshold - knee));
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}
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// for more information on the adaptive release curve, check out adaptive-release-curve.html demo +
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// source code included in this repo
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static inline float adaptivereleasecurve(float x, float a, float b, float c, float d){
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// a*x^3 + b*x^2 + c*x + d
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float x2 = x * x;
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return a * x2 * x + b * x2 + c * x + d;
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}
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static inline float clampf(float v, float min, float max){
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return v < min ? min : (v > max ? max : v);
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}
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static inline float absf(float v){
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return v < 0.0f ? -v : v;
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}
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static inline float fixf(float v, float def){
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// fix NaN and infinity values that sneak in... not sure why this is needed, but it is
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if (isnan(v) || isinf(v))
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return def;
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return v;
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}
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struct CompressorState
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{ // sf_compressor_state_st
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// user can read the metergain state variable after processing a chunk to see how much dB the
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// compressor would have liked to compress the sample; the meter values aren't used to shape the
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// sound in any way, only used for output if desired
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float metergain;
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// everything else shouldn't really be mucked with unless you read the algorithm and feel
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// comfortable
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float meterrelease;
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float threshold;
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float knee;
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float linearpregain;
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float linearthreshold;
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float slope;
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float attacksamplesinv;
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float satreleasesamplesinv;
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float wet;
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float dry;
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float k;
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float kneedboffset;
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float linearthresholdknee;
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float mastergain;
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float a; // adaptive release polynomial coefficients
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float b;
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float c;
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float d;
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float detectoravg;
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float compgain;
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float maxcompdiffdb;
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int delaybufsize;
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int delaywritepos;
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int delayreadpos;
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float delaybuf[AUDIOCOMPRESSORSND_SF_COMPRESSOR_MAXDELAY]; // predelay buffer
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// populate the compressor state with advanced parameters
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void sf_advancecomp(
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// these parameters are the same as the simple version above:
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int rate, float pregain, float threshold, float knee, float ratio, float attack, float release,
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// these are the advanced parameters:
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float predelay, // seconds, length of the predelay buffer [0 to 1]
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float releasezone1, // release zones should be increasing between 0 and 1, and are a fraction
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float releasezone2, // of the release time depending on the input dB -- these parameters define
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float releasezone3, // the adaptive release curve, which is discussed in further detail in the
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float releasezone4, // demo: adaptive-release-curve.html
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float postgain, // dB, amount of gain to apply after compression [0 to 100]
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float wet // amount to apply the effect [0 completely dry to 1 completely wet]
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);
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};
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static void sf_compressor_process(CompressorState *state, int size, float *input, float *output);
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CompressorState m_compressorState;
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float m_storageBuffer[AUDIOCOMPRESSORSND_SF_COMPRESSOR_CHUNKSIZE];
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float m_processedBuffer[AUDIOCOMPRESSORSND_SF_COMPRESSOR_CHUNKSIZE];
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int m_sampleIndex;
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};
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#endif // SDRBASE_AUDIO_AUDIOCOMPRESSORSND_H_
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