Added a limiter after the dexed engine to avoid clipping.

pull/32/head
Holger Wirtz 5 years ago
parent f49205b964
commit 764f56fba5
  1. 8
      MicroDexed.ino
  2. 144
      effect_dynamics.cpp
  3. 194
      effect_dynamics.h
  4. 50
      fast_log.h

@ -36,12 +36,14 @@
#include "effect_modulated_delay.h"
#include "effect_stereo_mono.h"
#include "effect_mono_stereo.h"
#include "effect_dynamics.h"
#include "PluginFx.h"
#include "UI.hpp"
#include "source_microdexed.h"
// Audio engines
AudioSourceMicroDexed* MicroDexed[NUM_DEXED];
AudioEffectDynamics* dexed_dynamic[NUM_DEXED];
AudioAmplifier* dexed_level[NUM_DEXED];
#if defined(USE_FX)
AudioSynthWaveform* chorus_modulator[NUM_DEXED];
@ -145,6 +147,7 @@ AudioConnection * dynamicConnections[NUM_DEXED * 5];
void create_audio_engine_chain(uint8_t instance_id)
{
MicroDexed[instance_id] = new AudioSourceMicroDexed(SAMPLE_RATE);
dexed_dynamic[instance_id] = new AudioEffectDynamics;
dexed_level[instance_id] = new AudioAmplifier();
mono2stereo[instance_id] = new AudioEffectMonoStereo();
#if defined(USE_FX)
@ -160,7 +163,8 @@ void create_audio_engine_chain(uint8_t instance_id)
#endif
dynamicConnections[nDynamic++] = new AudioConnection(*MicroDexed[instance_id], 0, microdexed_peak_mixer, instance_id);
dynamicConnections[nDynamic++] = new AudioConnection(*MicroDexed[instance_id], 0, *dexed_level[instance_id], 0);
dynamicConnections[nDynamic++] = new AudioConnection(*MicroDexed[instance_id], 0, *dexed_dynamic[instance_id], 0);
dynamicConnections[nDynamic++] = new AudioConnection(*dexed_dynamic[instance_id], 0, *dexed_level[instance_id], 0);
#if defined(USE_FX)
dynamicConnections[nDynamic++] = new AudioConnection(*dexed_level[instance_id], 0, *chorus_mixer[instance_id], 0);
dynamicConnections[nDynamic++] = new AudioConnection(*dexed_level[instance_id], 0, *modchorus[instance_id], 0);
@ -447,6 +451,8 @@ void setup()
Serial.println(F("]"));
Serial.print(F("Polyphony: "));
Serial.println(configuration.dexed[instance_id].polyphony, DEC);
dexed_dynamic[instance_id]->limit();
}
Serial.print(F("AUDIO_BLOCK_SAMPLES="));

@ -0,0 +1,144 @@
/* Audio Library for Teensy 3.X
* Dynamics Processor (Gate, Compressor & Limiter)
* Copyright (c) 2017, Marc Paquette (marc@dacsystemes.com)
* Based on analyse_rms & mixer objects by Paul Stoffregen
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* 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, development funding 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.
*/
#include "effect_dynamics.h"
#include "fast_log.h"
#include "utility/dspinst.h"
#include "utility/sqrt_integer.h"
static float analyse_rms(int16_t *data) {
uint32_t *p = (uint32_t *)data;
const uint32_t *end = p + AUDIO_BLOCK_SAMPLES / 2;
int64_t sum = 0;
do {
uint32_t n1 = *p++;
uint32_t n2 = *p++;
uint32_t n3 = *p++;
uint32_t n4 = *p++;
sum = multiply_accumulate_16tx16t_add_16bx16b(sum, n1, n1);
sum = multiply_accumulate_16tx16t_add_16bx16b(sum, n2, n2);
sum = multiply_accumulate_16tx16t_add_16bx16b(sum, n3, n3);
sum = multiply_accumulate_16tx16t_add_16bx16b(sum, n4, n4);
} while (p < end);
int32_t meansq = sum / AUDIO_BLOCK_SAMPLES;
return sqrt_uint32(meansq) / 32767.0f;
}
static void applyGain(int16_t *data, int32_t mult1, int32_t mult2) {
uint32_t *p = (uint32_t *)data;
const uint32_t *end = p + AUDIO_BLOCK_SAMPLES / 2;
int32_t inc = (mult2 - mult1) / (AUDIO_BLOCK_SAMPLES / 2);
do {
uint32_t tmp32 = *p; // read 2 samples from *data
int32_t val1 = signed_multiply_32x16b(mult1, tmp32);
mult1 += inc;
int32_t val2 = signed_multiply_32x16t(mult1, tmp32);
mult1 += inc;
val1 = signed_saturate_rshift(val1, 16, 0);
val2 = signed_saturate_rshift(val2, 16, 0);
*p++ = pack_16b_16b(val2, val1);
} while (p < end);
}
void AudioEffectDynamics::update(void) {
audio_block_t *block;
block = receiveWritable(0);
if (!block) return;
if (!gateEnabled && !compEnabled && !limiterEnabled) {
//Transmit & release
transmit(block);
release(block);
return;
}
//Analyze received block
float rms = analyse_rms(block->data);
//Compute block RMS level in Db
float inputdb = MIN_DB;
if (rms > 0) inputdb = unitToDb(rms);
//Gate
if (gateEnabled) {
if (inputdb >= gateThresholdOpen) gatedb = (aGateAttack * gatedb) + (aOneMinusGateAttack * MAX_DB);
else if (inputdb < gateThresholdClose) gatedb = (aGateRelease * gatedb) + (aOneMinusGateRelease * MIN_DB);
}
else gatedb = MAX_DB;
//Compressor
if (compEnabled) {
float attdb = MAX_DB; //Below knee
if (inputdb >= aLowKnee) {
if(inputdb <= aHighKnee) {
//Knee transition
float knee = inputdb - aLowKnee;
attdb = aKneeRatio * knee * knee * aTwoKneeWidth;
}
else {
//Above knee
attdb = compThreshold + ((inputdb - compThreshold) * compRatio) - inputdb;
}
}
if (attdb <= compdb) compdb = (aCompAttack * compdb) + (aOneMinusCompAttack * attdb);
else compdb = (aCompRelease * compdb) + (aOneMinusCompRelease * attdb);
}
else compdb = MAX_DB;
//Brickwall Limiter
if (limiterEnabled) {
float outdb = inputdb + compdb + makeupdb;
if (outdb >= limitThreshold) limitdb = (aLimitAttack * limitdb) + (aOneMinusLimitAttack * (limitThreshold - outdb));
else limitdb *= aLimitRelease;
}
else limitdb = MAX_DB;
//Compute linear gain
float totalGain = gatedb + compdb + makeupdb + limitdb;
int32_t mult = dbToUnit(totalGain) * 65536.0f;
//Apply gain to block
applyGain(block->data, last_mult, mult);
last_mult = mult;
//Transmit & release
transmit(block);
release(block);
}

@ -0,0 +1,194 @@
/* Audio Library for Teensy 3.X
* Dynamics Processor (Gate, Compressor & Limiter)
* Copyright (c) 2018, Marc Paquette (marc@dacsystemes.com)
* Based on analyse_rms, effect_envelope & mixer objects by Paul Stoffregen
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* 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, development funding 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 effect_dynamics_h_
#define effect_dynamics_h_
#include "Arduino.h"
#include "AudioStream.h"
#define MIN_DB -110.0f
#define MAX_DB 0.0f
#define MIN_T 0.03f //Roughly 1 block
#define MAX_T 4.00f
#define RATIO_OFF 1.0f
#define RATIO_INFINITY 60.0f
class AudioEffectDynamics : public AudioStream
{
public:
AudioEffectDynamics(void) : AudioStream(1, inputQueueArray) {
gate();
compression();
limit();
autoMakeupGain();
gatedb = MIN_DB;
compdb = MIN_DB;
limitdb = MIN_DB;
}
//Sets the gate parameters.
//threshold is in dbFS
//attack & release are in seconds
void gate(float threshold = -50.0f, float attack = MIN_T, float release = 0.3f, float hysterisis = 6.0f) {
gateEnabled = threshold > MIN_DB;
gateThresholdOpen = constrain(threshold, MIN_DB, MAX_DB);
gateThresholdClose = gateThresholdOpen - constrain(hysterisis, 0.0f, 6.0f);
float gateAttackTime = constrain(attack, MIN_T, MAX_T);
float gateReleaseTime = constrain(release, MIN_T, MAX_T);
aGateAttack = timeToAlpha(gateAttackTime);
aOneMinusGateAttack = 1.0f - aGateAttack;
aGateRelease = timeToAlpha(gateReleaseTime);
aOneMinusGateRelease = 1.0f - aGateRelease;
}
//Sets the compression parameters.
//threshold & kneeWidth are in db(FS)
//attack and release are in seconds
//ratio is expressed as x:1 i.e. 1 for no compression, 60 for brickwall limiting
//Set kneeWidth to 0 for hard knee
void compression(float threshold = -40.0f, float attack = MIN_T, float release = 0.5f, float ratio = 35.0f, float kneeWidth = 6.0f) {
compEnabled = threshold < MAX_DB;
compThreshold = constrain(threshold, MIN_DB, MAX_DB);
float compAttackTime = constrain(attack, MIN_T, MAX_T);
float compReleaseTime = constrain(release, MIN_T, MAX_T);
compRatio = 1.0f / constrain(abs(ratio), RATIO_OFF, RATIO_INFINITY);
float compKneeWidth = constrain(abs(kneeWidth), 0.0f, 32.0f);
computeMakeupGain();
aCompAttack = timeToAlpha(compAttackTime);
aOneMinusCompAttack = 1.0f - aCompAttack;
aCompRelease = timeToAlpha(compReleaseTime);
aOneMinusCompRelease = 1.0f - aCompRelease;
aHalfKneeWidth = compKneeWidth / 2.0f;
aTwoKneeWidth = 1.0f / (compKneeWidth * 2.0f);
aKneeRatio = compRatio - 1.0f;
aLowKnee = compThreshold - aHalfKneeWidth;
aHighKnee = compThreshold + aHalfKneeWidth;
}
//Sets the hard limiter parameters
//threshold is in dbFS
//attack & release are in seconds
void limit(float threshold = -3.0f, float attack = MIN_T, float release = MIN_T) {
limiterEnabled = threshold < MAX_DB;
limitThreshold = constrain(threshold, MIN_DB, MAX_DB);
float limitAttackTime = constrain(attack, MIN_T, MAX_T);
float limitReleaseTime = constrain(release, MIN_T, MAX_T);
computeMakeupGain();
aLimitAttack = timeToAlpha(limitAttackTime);
aOneMinusLimitAttack = 1.0f - aLimitAttack;
aLimitRelease = timeToAlpha(limitReleaseTime);
}
//Enables automatic makeup gain setting
//headroom is in dbFS
void autoMakeupGain(float headroom = 6.0f) {
mgAutoEnabled = true;
mgHeadroom = constrain(headroom, 0.0f, 60.0f);
computeMakeupGain();
}
//Sets a fixed makeup gain value.
//gain is in dbFS
void makeupGain(float gain = 0.0f) {
mgAutoEnabled = false;
makeupdb = constrain(gain, -12.0f, 24.0f);
}
private:
audio_block_t *inputQueueArray[1];
bool gateEnabled = false;
float gateThresholdOpen;
float gateThresholdClose;
float gatedb;
bool compEnabled = false;
float compThreshold;
float compRatio;
float compdb;
bool limiterEnabled = false;
float limitThreshold;
float limitdb;
bool mgAutoEnabled;
float mgHeadroom;
float makeupdb;
float aGateAttack;
float aOneMinusGateAttack;
float aGateRelease;
float aOneMinusGateRelease;
float aHalfKneeWidth;
float aTwoKneeWidth;
float aKneeRatio;
float aLowKnee;
float aHighKnee;
float aCompAttack;
float aOneMinusCompAttack;
float aCompRelease;
float aOneMinusCompRelease;
float aLimitAttack;
float aOneMinusLimitAttack;
float aLimitRelease;
int32_t last_mult;
void computeMakeupGain() {
if (mgAutoEnabled) {
makeupdb = -compThreshold + (compThreshold * compRatio) + limitThreshold - mgHeadroom;
}
}
//Computes smoothing time constants for a 10% to 90% change
float timeToAlpha(float time) {
return expf(-0.9542f / (((float)AUDIO_SAMPLE_RATE_EXACT / (float)AUDIO_BLOCK_SAMPLES) * time));
}
virtual void update(void);
};
#endif

@ -0,0 +1,50 @@
/* ----------------------------------------------------------------------
* https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
* Fast approximation to the log2() function. It uses a two step
* process. First, it decomposes the floating-point number into
* a fractional component F and an exponent E. The fraction component
* is used in a polynomial approximation and then the exponent added
* to the result. A 3rd order polynomial is used and the result
* when computing db20() is accurate to 7.984884e-003 dB.
** ------------------------------------------------------------------- */
float log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
float log2f_approx(float X)
{
float *C = &log2f_approx_coeff[0];
float Y;
float F;
int E;
// This is the approximation to log2()
F = frexpf(fabsf(X), &E);
// Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
Y = *C++;
Y *= F;
Y += (*C++);
Y *= F;
Y += (*C++);
Y *= F;
Y += (*C++);
Y += E;
return(Y);
}
// https://codingforspeed.com/using-faster-exponential-approximation/
inline float expf_approx(float x) {
x = 1.0f + x / 1024;
x *= x; x *= x; x *= x; x *= x;
x *= x; x *= x; x *= x; x *= x;
x *= x; x *= x;
return x;
}
inline float unitToDb(float unit) {
return 6.02f * log2f_approx(unit);
}
inline float dbToUnit(float db) {
return expf_approx(db * 2.302585092994046f * 0.05f);
}
Loading…
Cancel
Save