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[DEV] rename internal menber of class

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This commit is contained in:
Edouard DUPIN 2015-04-03 21:48:21 +02:00
parent 942a9cf67b
commit 4d3296a9e1
16 changed files with 161 additions and 256 deletions
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16
audio/algo/chunkware/AttRelEnvelope.cpp
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@ -24,27 +24,27 @@
*/
#include "AttRelEnvelope.h"
#include <audio/algo/chunkware/AttRelEnvelope.h>
audio::algo::chunkware::AttRelEnvelope::AttRelEnvelope(double att_ms, double rel_ms, double sampleRate) :
att_(att_ms, sampleRate),
rel_(rel_ms, sampleRate) {
audio::algo::chunkware::AttRelEnvelope::AttRelEnvelope(double m_attackms, double m_releasems, double sampleRate) :
m_attack(m_attackms, sampleRate),
m_release(m_releasems, sampleRate) {
}
//-------------------------------------------------------------
void audio::algo::chunkware::AttRelEnvelope::setAttack(double ms) {
att_.setTc(ms);
m_attack.setTc(ms);
}
//-------------------------------------------------------------
void audio::algo::chunkware::AttRelEnvelope::setRelease(double ms) {
rel_.setTc(ms);
m_release.setTc(ms);
}
//-------------------------------------------------------------
void audio::algo::chunkware::AttRelEnvelope::setSampleRate(double sampleRate) {
att_.setSampleRate(sampleRate);
rel_.setSampleRate(sampleRate);
m_attack.setSampleRate(sampleRate);
m_release.setSampleRate(sampleRate);
}

20
audio/algo/chunkware/AttRelEnvelope.h
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@ -27,7 +27,7 @@
#ifndef __AUDIO_ALGO_CHUNKWARE_ATT_REL_ENVELOPE_H__
#define __AUDIO_ALGO_CHUNKWARE_ATT_REL_ENVELOPE_H__
#include "EnvelopeDetector.h"
#include <audio/algo/chunkware/EnvelopeDetector.h>
namespace audio {
namespace algo {
@ -37,24 +37,24 @@ namespace audio {
//-------------------------------------------------------------
class AttRelEnvelope {
public:
AttRelEnvelope(double att_ms = 10.0,
double rel_ms = 100.0,
AttRelEnvelope(double m_attackms = 10.0,
double m_releasems = 100.0,
double sampleRate = 44100.0);
virtual ~AttRelEnvelope() {}
// attack time constant
virtual void setAttack(double ms);
virtual double getAttack() const {
return att_.getTc();
return m_attack.getTc();
}
// release time constant
virtual void setRelease(double ms);
virtual double getRelease() const {
return rel_.getTc();
return m_release.getTc();
}
// sample rate dependencies
virtual void setSampleRate(double sampleRate);
virtual double getSampleRate() const {
return att_.getSampleRate();
return m_attack.getSampleRate();
}
// runtime function
void run(double in, double &state) {
@ -65,15 +65,15 @@ namespace audio {
*/
if (in > state) {
// attack
att_.run(in, state);
m_attack.run(in, state);
} else {
// release
rel_.run(in, state);
m_release.run(in, state);
}
}
private:
EnvelopeDetector att_;
EnvelopeDetector rel_;
EnvelopeDetector m_attack;
EnvelopeDetector m_release;
};
}
}

25
audio/algo/chunkware/Compressor.cpp
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@ -23,27 +23,28 @@
* DEALINGS IN THE SOFTWARE.
*/
#include "Compressor.h"
#include <audio/algo/chunkware/Compressor.h>
#include <audio/algo/chunkware/debug.h>
audio::algo::chunkware::Compresssor::Compresssor() :
AttRelEnvelope(10.0, 100.0),
threshdB_(0.0),
ratio_(1.0),
envdB_(DC_OFFSET) {
m_threshdB(0.0),
m_ratio(1.0),
m_overThresholdEnvelopeDB(DC_OFFSET) {
}
void audio::algo::chunkware::Compresssor::setThresh(double dB) {
threshdB_ = dB;
m_threshdB = dB;
}
void audio::algo::chunkware::Compresssor::setRatio(double ratio) {
assert(ratio > 0.0);
ratio_ = ratio;
AA_CHUNK_ASSERT(ratio > 0.0, "input function error");
m_ratio = ratio;
}
void audio::algo::chunkware::Compresssor::initRuntime() {
envdB_ = DC_OFFSET;
m_overThresholdEnvelopeDB = DC_OFFSET;
}
void audio::algo::chunkware::Compresssor::process(double &in1, double &in2) {
@ -63,13 +64,13 @@ void audio::algo::chunkware::Compresssor::process(double &in1, double &in2, doub
keyLinked += DC_OFFSET; // add DC offset to avoid log(0)
double keydB = lin2dB(keyLinked); // convert linear -> dB
// threshold
double overdB = keydB - threshdB_; // delta over threshold
double overdB = keydB - m_threshdB; // delta over threshold
if (overdB < 0.0)
overdB = 0.0;
// attack/release
overdB += DC_OFFSET; // add DC offset to avoid denormal
AttRelEnvelope::run(overdB, envdB_); // run attack/release envelope
overdB = envdB_ - DC_OFFSET; // subtract DC offset
AttRelEnvelope::run(overdB, m_overThresholdEnvelopeDB); // run attack/release envelope
overdB = m_overThresholdEnvelopeDB - DC_OFFSET; // subtract DC offset
/* REGARDING THE DC OFFSET: In this case, since the offset is added before
* the attack/release processes, the envelope will never fall below the offset,
* thereby avoiding denormals. However, to prevent the offset from causing
@ -77,7 +78,7 @@ void audio::algo::chunkware::Compresssor::process(double &in1, double &in2, doub
* a minimum value of 0dB.
*/
// transfer function
double gr = overdB * (ratio_ - 1.0); // gain reduction (dB)
double gr = overdB * (m_ratio - 1.0); // gain reduction (dB)
gr = dB2lin(gr); // convert dB -> linear
// output gain
in1 *= gr; // apply gain reduction to input

16
audio/algo/chunkware/Compressor.h
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@ -27,9 +27,9 @@
#ifndef __AUDIO_ALGO_CHUNKWARE_COMPRESSOR_H__
#define __AUDIO_ALGO_CHUNKWARE_COMPRESSOR_H__
#include "header.h"
#include "AttRelEnvelope.h"
#include "Gain.h"
#include <etk/types.h>
#include <audio/algo/chunkware/AttRelEnvelope.h>
#include <audio/algo/chunkware/Gain.h>
namespace audio {
namespace algo {
@ -42,10 +42,10 @@ namespace audio {
virtual void setThresh(double dB);
virtual void setRatio(double dB);
virtual double getThresh() const {
return threshdB_;
return m_threshdB;
}
virtual double getRatio() const {
return ratio_;
return m_ratio;
}
// runtime
// call before runtime (in resume())
@ -56,10 +56,10 @@ namespace audio {
void process(double &in1, double &in2, double keyLinked);
private:
// transfer function
double threshdB_;//!< threshold (dB)
double ratio_; //!< ratio (compression: < 1 ; expansion: > 1)
double m_threshdB;//!< threshold (dB)
double m_ratio; //!< ratio (compression: < 1 ; expansion: > 1)
// runtime variables
double envdB_; //!< over-threshold envelope (dB)
double m_overThresholdEnvelopeDB; //!< over-threshold envelope (dB)
};
}
}

16
audio/algo/chunkware/CompressorRms.cpp
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@ -24,27 +24,27 @@
*/
#include "CompressorRms.h"
#include <audio/algo/chunkware/CompressorRms.h>
audio::algo::chunkware::CompresssorRms::CompresssorRms() :
ave_(5.0),
aveOfSqrs_(DC_OFFSET) {
m_averager(5.0),
m_averageSuares(DC_OFFSET) {
}
void audio::algo::chunkware::CompresssorRms::setSampleRate(double sampleRate) {
audio::algo::chunkware::Compresssor::setSampleRate(sampleRate);
ave_.setSampleRate(sampleRate);
m_averager.setSampleRate(sampleRate);
}
void audio::algo::chunkware::CompresssorRms::setWindow(double ms) {
ave_.setTc(ms);
m_averager.setTc(ms);
}
void audio::algo::chunkware::CompresssorRms::initRuntime() {
audio::algo::chunkware::Compresssor::initRuntime();
aveOfSqrs_ = DC_OFFSET;
m_averageSuares = DC_OFFSET;
}
void audio::algo::chunkware::CompresssorRms::process(double &in1, double &in2) {
@ -53,8 +53,8 @@ void audio::algo::chunkware::CompresssorRms::process(double &in1, double &in2) {
double inSq2 = in2 * in2;
double sum = inSq1 + inSq2; // power summing
sum += DC_OFFSET; // DC offset, to prevent denormal
ave_.run(sum, aveOfSqrs_); // average of squares
double rms = sqrt(aveOfSqrs_); // rms (sort of ...)
m_averager.run(sum, m_averageSuares); // average of squares
double rms = sqrt(m_averageSuares); // rms (sort of ...)
/* REGARDING THE RMS AVERAGER: Ok, so this isn't a REAL RMS
* calculation. A true RMS is an FIR moving average. This
* approximation is a 1-pole IIR. Nonetheless, in practice,

8
audio/algo/chunkware/CompressorRms.h
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@ -27,7 +27,7 @@
#ifndef __AUDIO_ALGO_CHUNKWARE_COMPRESSOR_RMS_H__
#define __AUDIO_ALGO_CHUNKWARE_COMPRESSOR_RMS_H__
#include "Compressor.h"
#include <audio/algo/chunkware/Compressor.h>
namespace audio {
namespace algo {
@ -40,13 +40,13 @@ namespace audio {
virtual void setSampleRate(double sampleRate);
// RMS window
virtual void setWindow(double ms);
virtual double getWindow() const { return ave_.getTc(); }
virtual double getWindow() const { return m_averager.getTc(); }
// runtime process
virtual void initRuntime(); // call before runtime (in resume())
void process(double &in1, double &in2); // compressor runtime process
protected:
EnvelopeDetector ave_; //!< averager
double aveOfSqrs_; //!< average of squares
EnvelopeDetector m_averager; //!< averager
double m_averageSuares; //!< average of squares
};
}
}

21
audio/algo/chunkware/EnvelopeDetector.cpp
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@ -24,28 +24,29 @@
*/
#include "EnvelopeDetector.h"
#include <audio/algo/chunkware/debug.h>
#include <audio/algo/chunkware/EnvelopeDetector.h>
audio::algo::chunkware::EnvelopeDetector::EnvelopeDetector(double ms, double sampleRate) {
assert(sampleRate > 0.0);
assert(ms > 0.0);
sampleRate_ = sampleRate;
ms_ = ms;
AA_CHUNK_ASSERT(sampleRate > 0.0, "input function error");
AA_CHUNK_ASSERT(ms > 0.0, "input function error");
m_sampleRate = sampleRate;
m_timeMs = ms;
setCoef();
}
void audio::algo::chunkware::EnvelopeDetector::setTc(double ms) {
assert(ms > 0.0);
ms_ = ms;
AA_CHUNK_ASSERT(ms > 0.0, "input function error");
m_timeMs = ms;
setCoef();
}
void audio::algo::chunkware::EnvelopeDetector::setSampleRate(double sampleRate) {
assert(sampleRate > 0.0);
sampleRate_ = sampleRate;
AA_CHUNK_ASSERT(sampleRate > 0.0, "input function error");
m_sampleRate = sampleRate;
setCoef();
}
void audio::algo::chunkware::EnvelopeDetector::setCoef() {
coef_ = exp(-1000.0 / (ms_ * sampleRate_));
m_coefficient = exp(-1000.0 / (m_timeMs * m_sampleRate));
}

14
audio/algo/chunkware/EnvelopeDetector.h
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@ -27,7 +27,7 @@
#ifndef __AUDIO_ALGO_CHUNKWARE_ENVELOPE_DETECTOR_H__
#define __AUDIO_ALGO_CHUNKWARE_ENVELOPE_DETECTOR_H__
#include "header.h"
#include <etk/types.h>
namespace audio {
namespace algo {
@ -50,21 +50,21 @@ namespace audio {
// time constant
virtual void setTc(double ms);
virtual double getTc() const {
return ms_;
return m_timeMs;
}
// sample rate
virtual void setSampleRate(double sampleRate);
virtual double getSampleRate() const {
return sampleRate_;
return m_sampleRate;
}
// runtime function
void run(double in, double &state) {
state = in + coef_ * (state - in);
state = in + m_coefficient * (state - in);
}
protected:
double sampleRate_; //!< sample rate
double ms_; //!< time constant in ms
double coef_; //!< runtime coefficient
double m_sampleRate; //!< sample rate
double m_timeMs; //!< time constant in ms
double m_coefficient; //!< runtime coefficient
virtual void setCoef(); //!< coef calculation
};
}

2
audio/algo/chunkware/Gain.h
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@ -27,7 +27,7 @@
#ifndef __AUDIO_ALGO_CHUNKWARE_GAIN_H__
#define __AUDIO_ALGO_CHUNKWARE_GAIN_H__
#include "header.h"
#include <etk/types.h>
namespace audio {

59
audio/algo/chunkware/Gate.cpp
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@ -24,23 +24,23 @@
*/
#include "Gate.h"
#include <audio/algo/chunkware/Gate.h>
audio::algo::chunkware::Gate::Gate() :
AttRelEnvelope(1.0, 100.0),
threshdB_(0.0),
thresh_(1.0),
env_(DC_OFFSET) {
m_threshdB(0.0),
m_threshold(1.0),
m_overThresholdEnvelope(DC_OFFSET) {
}
void audio::algo::chunkware::Gate::setThresh(double dB) {
threshdB_ = dB;
thresh_ = dB2lin(dB);
m_threshdB = dB;
m_threshold = dB2lin(dB);
}
void audio::algo::chunkware::Gate::initRuntime() {
env_ = DC_OFFSET;
m_overThresholdEnvelope = DC_OFFSET;
}
@ -59,11 +59,11 @@ void audio::algo::chunkware::Gate::process(double &in1, double &in2, double keyL
keyLinked = fabs(keyLinked); // rectify (just in case)
// threshold
// key over threshold (0.0 or 1.0)
double over = double(keyLinked > thresh_);
double over = double(keyLinked > m_threshold);
// attack/release
over += DC_OFFSET; // add DC offset to avoid denormal
AttRelEnvelope::run(over, env_); // run attack/release
over = env_ - DC_OFFSET; // subtract DC offset
AttRelEnvelope::run(over, m_overThresholdEnvelope); // run attack/release
over = m_overThresholdEnvelope - DC_OFFSET; // subtract DC offset
/* REGARDING THE DC OFFSET: In this case, since the offset is added before
* the attack/release processes, the envelope will never fall below the offset,
* thereby avoiding denormals. However, to prevent the offset from causing
@ -74,42 +74,3 @@ void audio::algo::chunkware::Gate::process(double &in1, double &in2, double keyL
in1 *= over; // apply gain reduction to input
in2 *= over;
}
audio::algo::chunkware::GateRms::GateRms() :
ave_(5.0),
aveOfSqrs_(DC_OFFSET) {
}
void audio::algo::chunkware::GateRms::setSampleRate(double sampleRate) {
audio::algo::chunkware::Gate::setSampleRate(sampleRate);
ave_.setSampleRate(sampleRate);
}
void audio::algo::chunkware::GateRms::setWindow(double ms) {
ave_.setTc(ms);
}
void audio::algo::chunkware::GateRms::initRuntime() {
audio::algo::chunkware::Gate::initRuntime();
aveOfSqrs_ = DC_OFFSET;
}
void audio::algo::chunkware::GateRms::process(double &in1, double &in2) {
// create sidechain
double inSq1 = in1 * in1; // square input
double inSq2 = in2 * in2;
double sum = inSq1 + inSq2; // power summing
sum += DC_OFFSET; // DC offset, to prevent denormal
ave_.run(sum, aveOfSqrs_); // average of squares
double rms = sqrt(aveOfSqrs_); // rms (sort of ...)
/* REGARDING THE RMS AVERAGER: Ok, so this isn't a REAL RMS
* calculation. A true RMS is an FIR moving average. This
* approximation is a 1-pole IIR. Nonetheless, in practice,
* and in the interest of simplicity, this method will suffice,
* giving comparable results.
*/
Gate::process(in1, in2, rms); // rest of process
}

40
audio/algo/chunkware/Gate.h
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@ -27,16 +27,13 @@
#ifndef __AUDIO_ALGO_CHUNKWARE_GATE_H__
#define __AUDIO_ALGO_CHUNKWARE_GATE_H__
#include "header.h"
#include "AttRelEnvelope.h"
#include "Gain.h"
#include <etk/types.h>
#include <audio/algo/chunkware/AttRelEnvelope.h>
#include <audio/algo/chunkware/Gain.h>
namespace audio {
namespace algo {
namespace chunkware {
//-------------------------------------------------------------
// simple gate
//-------------------------------------------------------------
class Gate : public AttRelEnvelope {
public:
Gate();
@ -44,7 +41,7 @@ namespace audio {
// parameters
virtual void setThresh(double dB);
virtual double getThresh() const {
return threshdB_;
return m_threshdB;
}
// runtime
// call before runtime (in resume())
@ -55,33 +52,10 @@ namespace audio {
void process(double &in1, double &in2, double keyLinked);
private:
// transfer function
double threshdB_; //!< threshold (dB)
double thresh_; //!< threshold (linear)
double m_threshdB; //!< threshold (dB)
double m_threshold; //!< threshold (linear)
// runtime variables
double env_; //!< over-threshold envelope (linear)
};
//-------------------------------------------------------------
// simple gate with RMS detection
//-------------------------------------------------------------
class GateRms : public Gate {
public:
GateRms();
virtual ~GateRms() {}
// sample rate
virtual void setSampleRate(double sampleRate);
// RMS window
virtual void setWindow(double ms);
virtual double getWindow() const {
return ave_.getTc();
}
// runtime process
// call before runtime (in resume())
virtual void initRuntime();
// gate runtime process
void process(double &in1, double &in2);
private:
EnvelopeDetector ave_; //!< averager
double aveOfSqrs_; //!< average of squares
double m_overThresholdEnvelope; //!< over-threshold envelope (linear)
};
}
}

103
audio/algo/chunkware/Limiter.cpp
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@ -24,57 +24,58 @@
*/
#include "Limiter.h"
#include <audio/algo/chunkware/Limiter.h>
#include <audio/algo/chunkware/debug.h>
audio::algo::chunkware::Limiter::Limiter() :
threshdB_(0.0),
thresh_(1.0),
peakHold_(0),
peakTimer_(0),
maxPeak_(1.0),
att_(1.0),
rel_(10.0),
env_(1.0),
mask_(BUFFER_SIZE-1),
cur_(0) {
m_threshdB(0.0),
m_threshold(1.0),
m_peakHold(0),
m_peakTimer(0),
m_maxPeak(1.0),
m_attack(1.0),
m_release(10.0),
m_overThresholdEnvelope(1.0),
m_bufferMask(BUFFER_SIZE-1),
m_cursor(0) {
setAttack(1.0);
outBuffer_[ 0 ].resize(BUFFER_SIZE, 0.0);
outBuffer_[ 1 ].resize(BUFFER_SIZE, 0.0);
m_outputBuffer[ 0 ].resize(BUFFER_SIZE, 0.0);
m_outputBuffer[ 1 ].resize(BUFFER_SIZE, 0.0);
}
void audio::algo::chunkware::Limiter::setThresh(double dB) {
threshdB_ = dB;
thresh_ = dB2lin(dB);
m_threshdB = dB;
m_threshold = dB2lin(dB);
}
void audio::algo::chunkware::Limiter::setAttack(double ms) {
unsigned int samp = int(0.001 * ms * att_.getSampleRate());
assert(samp < BUFFER_SIZE);
peakHold_ = samp;
att_.setTc(ms);
unsigned int samp = int(0.001 * ms * m_attack.getSampleRate());
AA_CHUNK_ASSERT(samp < BUFFER_SIZE, "input function error");
m_peakHold = samp;
m_attack.setTc(ms);
}
void audio::algo::chunkware::Limiter::setRelease(double ms) {
rel_.setTc(ms);
m_release.setTc(ms);
}
void audio::algo::chunkware::Limiter::setSampleRate(double sampleRate) {
att_.setSampleRate(sampleRate);
rel_.setSampleRate(sampleRate);
m_attack.setSampleRate(sampleRate);
m_release.setSampleRate(sampleRate);
}
void audio::algo::chunkware::Limiter::initRuntime() {
peakTimer_ = 0;
maxPeak_ = thresh_;
env_ = thresh_;
cur_ = 0;
outBuffer_[ 0 ].assign(BUFFER_SIZE, 0.0);
outBuffer_[ 1 ].assign(BUFFER_SIZE, 0.0);
m_peakTimer = 0;
m_maxPeak = m_threshold;
m_overThresholdEnvelope = m_threshold;
m_cursor = 0;
m_outputBuffer[ 0 ].assign(BUFFER_SIZE, 0.0);
m_outputBuffer[ 1 ].assign(BUFFER_SIZE, 0.0);
}
void audio::algo::chunkware::Limiter::FastEnvelope::setCoef() {
// rises to 99% of in value over duration of time constant
coef_ = pow(0.01, (1000.0 / (ms_ * sampleRate_)));
m_coefficient = pow(0.01, (1000.0 / (m_timeMs * m_sampleRate)));
}
@ -85,15 +86,15 @@ void audio::algo::chunkware::Limiter::process(double &in1, double &in2) {
double keyLink = std::max(rect1, rect2); // link channels with greater of 2
// threshold
// we always want to feed the sidechain AT LEATS the threshold value
if (keyLink < thresh_)
keyLink = thresh_;
if (keyLink < m_threshold)
keyLink = m_threshold;
// test:
// a) whether peak timer has "expired"
// b) whether new peak is greater than previous max peak
if ((++peakTimer_ >= peakHold_) || (keyLink > maxPeak_)) {
if ((++m_peakTimer >= m_peakHold) || (keyLink > m_maxPeak)) {
// if either condition is met:
peakTimer_ = 0; // reset peak timer
maxPeak_ = keyLink; // assign new peak to max peak
m_peakTimer = 0; // reset peak timer
m_maxPeak = keyLink; // assign new peak to max peak
}
/* REGARDING THE MAX PEAK: This method assumes that the only important
* sample in a look-ahead buffer would be the highest peak. As such,
@ -112,12 +113,12 @@ void audio::algo::chunkware::Limiter::process(double &in1, double &in2) {
* envelope follower.
*/
// attack/release
if (maxPeak_ > env_) {
if (m_maxPeak > m_overThresholdEnvelope) {
// run attack phase
att_.run(maxPeak_, env_);
m_attack.run(m_maxPeak, m_overThresholdEnvelope);
} else {
// run release phase
rel_.run(maxPeak_, env_);
m_release.run(m_maxPeak, m_overThresholdEnvelope);
}
/* REGARDING THE ATTACK: This limiter achieves "look-ahead" detection
* by allowing the envelope follower to attack the max peak, which is
@ -130,18 +131,18 @@ void audio::algo::chunkware::Limiter::process(double &in1, double &in2) {
* threshold (which is assumed to be around 1.0 linear).
*/
// gain reduction
double gR = thresh_ / env_;
double gR = m_threshold / m_overThresholdEnvelope;
// unload current buffer index
// (cur_ - delay) & mask_ gets sample from [delay] samples ago
// mask_ variable wraps index
unsigned int delayIndex = (cur_ - peakHold_) & mask_;
double delay1 = outBuffer_[ 0 ][ delayIndex ];
double delay2 = outBuffer_[ 1 ][ delayIndex ];
// (m_cursor - delay) & m_bufferMask gets sample from [delay] samples ago
// m_bufferMask variable wraps index
unsigned int delayIndex = (m_cursor - m_peakHold) & m_bufferMask;
double delay1 = m_outputBuffer[ 0 ][ delayIndex ];
double delay2 = m_outputBuffer[ 1 ][ delayIndex ];
// load current buffer index and advance current index
// mask_ wraps cur_ index
outBuffer_[ 0 ][ cur_ ] = in1;
outBuffer_[ 1 ][ cur_ ] = in2;
++cur_ &= mask_;
// m_bufferMask wraps m_cursor index
m_outputBuffer[ 0 ][ m_cursor ] = in1;
m_outputBuffer[ 1 ][ m_cursor ] = in2;
++m_cursor &= m_bufferMask;
// output gain
in1 = delay1 * gR; // apply gain reduction to input
in2 = delay2 * gR;
@ -157,11 +158,11 @@ void audio::algo::chunkware::Limiter::process(double &in1, double &in2) {
*
* 2) Clip the output at the threshold, as such:
*
* if (in1 > thresh_) in1 = thresh_;
* else if (in1 < -thresh_) in1 = -thresh_;
* if (in1 > m_threshold) in1 = m_threshold;
* else if (in1 < -m_threshold) in1 = -m_threshold;
*
* if (in2 > thresh_) in2 = thresh_;
* else if (in2 < -thresh_) in2 = -thresh_;
* if (in2 > m_threshold) in2 = m_threshold;
* else if (in2 < -m_threshold) in2 = -m_threshold;
*
* (... or replace with your favorite branchless clipper ...)
*/

38
audio/algo/chunkware/Limiter.h
View File

@ -27,9 +27,9 @@
#ifndef __AUDIO_ALGO_CHUNKWARE_LIMITER_H__
#define __AUDIO_ALGO_CHUNKWARE_LIMITER_H__
#include "header.h"
#include "AttRelEnvelope.h"
#include "Gain.h"
#include <etk/types.h>
#include <audio/algo/chunkware/AttRelEnvelope.h>
#include <audio/algo/chunkware/Gain.h>
#include <vector>
namespace audio {
@ -47,22 +47,22 @@ namespace audio {
virtual void setAttack(double ms);
virtual void setRelease(double ms);
virtual double getThresh() const {
return threshdB_;
return m_threshdB;
}
virtual double getAttack() const {
return att_.getTc();
return m_attack.getTc();
}
virtual double getRelease() const {
return rel_.getTc();
return m_release.getTc();
}
// latency
virtual const unsigned int getLatency() const {
return peakHold_;
return m_peakHold;
}
// sample rate dependencies
virtual void setSampleRate(double sampleRate);
virtual double getSampleRate() {
return att_.getSampleRate();
return m_attack.getSampleRate();
}
// runtime
// call before runtime (in resume())
@ -84,23 +84,23 @@ namespace audio {
};
private:
// transfer function
double threshdB_; // threshold (dB)
double thresh_; // threshold (linear)
double m_threshdB; //!< threshold (dB)
double m_threshold; //!< threshold (linear)
// max peak
unsigned int peakHold_; // peak hold (samples)
unsigned int peakTimer_; // peak hold timer
double maxPeak_; // max peak
unsigned int m_peakHold; //!< peak hold (samples)
unsigned int m_peakTimer; //!< peak hold timer
double m_maxPeak; //!< max peak
// attack/release envelope
FastEnvelope att_; // attack
FastEnvelope rel_; // release
double env_; // over-threshold envelope (linear)
FastEnvelope m_attack; //!< attack
FastEnvelope m_release; //!< release
double m_overThresholdEnvelope; //!< over-threshold envelope (linear)
// buffer
// BUFFER_SIZE default can handle up to ~10ms at 96kHz
// change this if you require more
static const int BUFFER_SIZE = 1024; //!< buffer size (always a power of 2!)
unsigned int mask_; //!< buffer mask
unsigned int cur_; //!< cursor
std::vector< double > outBuffer_[ 2 ]; //!< output buffer
unsigned int m_bufferMask; //!< buffer mask
unsigned int m_cursor; //!< cursor
std::vector< double > m_outputBuffer[2]; //!< output buffer
};
}
}

2
audio/algo/chunkware/debug.h
View File

@ -56,7 +56,7 @@ namespace audio {
#define AA_CHUNK_ASSERT(cond,data) \
do { \
if (!(cond)) { \
APPL_CRITICAL(data); \
AA_CHUNK_CRITICAL(data); \
assert(!#cond); \
} \
} while (0)

34
audio/algo/chunkware/header.h
View File

@ -1,34 +0,0 @@
/**
* @author Bojan MARKOVIC
* @author Edouard DUPIN
* @copyright 2006, ChunkWare Music Software, OPEN-SOURCE
* @license BSD-1 (see license file)
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* * The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*/
#ifndef __SIMPLE_HEADER_H__
#define __SIMPLE_HEADER_H__
#include <algorithm>
#include <cassert>
#include <cmath>
#endif

1
lutin_audio_algo_chunkware.py
View File

@ -16,6 +16,7 @@ def create(target):
'audio/algo/chunkware/EnvelopeDetector.cpp',
'audio/algo/chunkware/AttRelEnvelope.cpp',
'audio/algo/chunkware/Gate.cpp',
'audio/algo/chunkware/GateRms.cpp',
'audio/algo/chunkware/Limiter.cpp'
])
myModule.add_module_depend(['etk'])