/*
* AudioEffectCompWDR2_F32: Wide Dynamic Rnage Compressor #2
*
* Bob Larkin W7PUA 11 December 2020 *********** UNDER DEVELOPMENT SUBJECT TO CHANGE!!!!
* This is an attempt to simplify and further comment the Chip Audette WDRC compressor.
* Derived from: Chip Audette (OpenAudio) Feb 2017
* Which was derived From: WDRC_circuit from CHAPRO from BTNRC: https://github.com/BTNRH/chapro
* As of Feb 2017, CHAPRO license is listed as "Creative Commons?"
*
* MIT License. Use at your own risk.
*/
/*
* WDRC2 Wide dynamic range compressor #2. Amplifies input signals by a fixed amoount
* when the input is low. Above a first knee, the gain is reduce progressively more as
* the input gets greater. On a dB out vs. dB in curve, this shows as a chnge in the
* original 1:1 slope to a lesser slope of 1:cr1 where cr1 is the first compression ratio.
* Finally there is a second knee where the gain is reduced at an even greater rate. In the
* extreme this becomes a hard limiter, but it can continue to increase slightly at a dB
* rate of 1:cr2, with cr2 the second compression ratio.
Vout dB
|
|
0.0| **********#
| **********
| @********** 1:cr2
| ****
| ***
| ***
| *** 1:cr1
| ***
| @***
| *
| *
| *
| * Vout = Vin + g0 (all in dB)
| * 1:1
| *
| * * Vout vs. Vin in dB *
|* Knees (breakpoints) are shown with '@'
* Zero, zero intersection shown with '#'
*| Slopes are ratio of: output:input (in dB)
* |
* |________|___________________|____________________________|_________ Vin dB
k1 k2 0.0
* The graph shows the changes in gain on a log or dB scale. A 1:1 slope represents
* a constant gain with level. When the slope is less, say cr1:1 where cr1 might be 3,
* the voltage gain is decreasing as the input level increases.
*
* The model here is, I believe, the same as the two references above (Audette and CHAPRO).
* The variable names have been changed to avoid confusion with those of audiologists and
* to be easier to follow for non-audiologists. Here goes:
* gain0DB Gain, in dB of the compressor for low level inputs (g0 on graph) [38 dB]
* knee1dB First knee on the gain curve where the dB gain slope decreases(k1) [-50 dB]
* cr1 Compression ratio on dB curve between knee1dB and knee2dB [3.0]
* knee2dB Second knee on the gain curve where the dB gain slope decreases further (k2) [-20 dB]
* cr2 Compression ratio on dB curve above knee2dB [10.0]
*
* The presets for the above quantities, shown in square brackest, are qite aggressive,
* with a lot of compression (up to 38 dB). This is for demonstration, and each
* situation will have different settings. For the presets, the following data
* was measured, essentiallly as predicted:
* vIn (rel full scale)=0.001 vInDB=-60.05 vOutDB-vInDB=38.00
* vIn (rel full scale)=0.003 vInDB=-50.47 vOutDB-vInDB=38.00
* vIn (rel full scale)=0.01 vInDB=-40.00 vOutDB-vInDB=31.38
* vIn (rel full scale)=0.03 vInDB=-30.45 vOutDB-vInDB=24.97
* vIn (rel full scale)=0.1 vInDB=-19.98 vOutDB-vInDB=19.98
* vIn (rel full scale)=0.3 vInDB=-10.45 vOutDB-vInDB= 9.40
* vIn (rel full scale)=1.0 vInDB= 0.01 vOutDB-vInDB=-0.01
*
* vInDB refers to the time averaged envelope voltage.
* Needing a zero reference, this has been chosen as full ADC range output. This is ±1.0
* peak or 0.707 RMS in F32 terminology. If this is fixed, the low-level gain will also be
* determined. This is calculated in the constructor.
*
* The curve is for gainOffsetDB = 0.0. This parameter raises and lowers the entire gain
* curve by this many dB. This is equivalent to a post-compressor gain (or loss).
*
* Note: This is all done in conventional 10 based dB. This ends up with scaling in
* several places that could be eliminated by using 2B instead of dB, i.e.,
* use log2() and 2^(). This would seem to be faster, but less "readable."
*
* *********** UNDER DEVELOPMENT SUBJECT TO CHANGE!!!! *********
*/
#ifndef _AudioEffectCompWDRC2_F32
#define _AudioEffectCompWDRC2_F32
#include <Arduino.h>
#include <AudioStream_F32.h>
class AudioEffectWDRC2_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName: CompressWDRC2
public:
AudioEffectWDRC2_F32(void): AudioStream_F32(1,inputQueueArray) {
setAttackReleaseSec(0.005f, 0.100f);
setLowLevelGain(); // Not an independent variable, set by knees, cr's and 0,0 intersection
// setSampleRate_Hz(AUDIO_SAMPLE_RATE);
}
//AudioEffectCompWDRC_F32(AudioSettings_F32 settings): AudioStream_F32(1,inputQueueArray) {
// setSampleRate_Hz(settings.sample_rate_Hz);
//}
// Here is the method called automatically by the audio library
void update(void) {
float vAbs, vPeak;
float vInDB, vOutDB;
float targetGain;
// Receive the input audio data
audio_block_f32_t *block = AudioStream_F32::receiveWritable_f32();
if (!block) return;
// Allocate memory for the output
audio_block_f32_t *out_block = AudioStream_F32::allocate_f32();
if (!out_block)
{
release(block);
return;
}
// Find the smoothed envelope, target gain and compressed output
vPeak = vPeakSave;
for (int k=0; k<block->length; k++) {
vAbs = (block->data[k] >= 0.0f) ? block->data[k] : -block->data[k];
if (vAbs >= vPeak) { // Attack (rising level)
vPeak = alpha * vPeak + (oneMinusAlpha) * vAbs;
} else { // Release (decay for falling level)
vPeak = beta * vPeak;
}
// Convert to dB
// At all levels and quite frequency flat, this under estimates by 1.05 dB
vInDB = v2DB_Approx(vPeak) + 1.05f;
// Convert to desired Vout_DB, this is the compression curve
if(vInDB<=knee1DB)
vOutDB = vInDB + gain0DB; // No compression
else if(vInDB<knee2DB)
vOutDB = vInDB + gain0DB + (knee1DB - vInDB)*(cr1 - 1.0f)/cr1; // Middle region
else
vOutDB = vInDB + gain0DB + (knee2DB - vInDB)*(cr2 - 1.0f)/cr2 +
(knee1DB - knee2DB)*(cr1 - 1)/cr1; // High level region
// A note: from the latter, algebra says for a 0, 0 intersection of vInDB and vOutDB
// See setLowLevelGain()
// Convert the needed gain back to a voltage ratio 10^(db/20)
targetGain = pow10f(0.05f*(vOutDB - vInDB + gainOffsetDB));
// And apply target gain to signal stream from the delayed data. The
// delay buffer is circular because of delayBufferMask and length 2^m m<=8.
out_block->data[k] = targetGain * delayData[(k + in_index) & delayBufferMask];
if(printIO) {
Serial.print(block->data[k],6);
Serial.print("," );
Serial.print(delayData[(k + in_index) & delayBufferMask],6);
Serial.print("," );
Serial.println(targetGain);
}
// Put the new data into the delay line, delaySize positions ahead.
// If delaySize==256, this will be the same location as we just got data from.
delayData[(k + in_index + delaySize) & delayBufferMask] = block->data[k];
}
vPeakSave = vPeak; // save last vPeak for next time
sampleInputDB = vInDB; // Last values for get...() functions
sampleGainDB = vOutDB - vInDB;
// transmit the block and release memory
AudioStream_F32::release(block);
AudioStream_F32::transmit(out_block); // send the FIR output
AudioStream_F32::release(out_block);
// Update pointer in_index to delay line for next 128 update
in_index = (in_index + block->length) & delayBufferMask;
}
// gain0DB is the gain at low levels, below compression. Not an independent variable,
// so this should becalled after any change is made to knees and compression ratios.
void setLowLevelGain(void)
{
gain0DB = knee2DB*(1.0f - cr2)/cr2 + (knee2DB - knee1DB)*(cr1 - 1.0f)/cr1; // Low-level gain
}
void setOutputGainOffsetDB(float _gOff) { gainOffsetDB = _gOff; }
void setKnee1LowDB(float _k1) { knee1DB = _k1; }
void setCompressionRatioMiddleDB(float _cr1) { cr1 = _cr1; }
void setKnee2HighDB(float _k2) { knee2DB = _k2; }
void setCompressionRatioHighDB(float _cr2) { cr2 = _cr2; }
// A delay of 256 samples is 256/44100 = 0.0058 sec = 5.8 mSec
void setDelayBufferSize(int16_t _delaySize) { // Any power of 2, i.e., 256, 128, 64, etc.
delaySize = _delaySize;
delayBufferMask = _delaySize - 1;
in_index = 0;
}
void printOn(bool _printIO) { printIO = _printIO; } // Diagnostics ONLY. Not for general INO
float getLowLevelGainDB(void) { return gain0DB; }
float getCurrentInputDB(void) { return sampleInputDB; }
float getCurrentGainDB(void) { return sampleGainDB; }
//convert time constants from seconds to unitless parameters, from CHAPRO, agc_prepare.c
void setAttackReleaseSec(const float atk_sec, const float rel_sec) {
// convert ANSI attack & release times to filter time constants
float ansi_atk = atk_sec * sample_rate_Hz / 2.425f;
float ansi_rel = rel_sec * sample_rate_Hz / 1.782f;
alpha = (float) (ansi_atk / (1.0f + ansi_atk));
oneMinusAlpha = 1.0f - alpha;
beta = (float) (ansi_rel / (1.0f + ansi_rel));
}
// Accelerate the powf(10.0,x) function (from Chip's single slope compressor)
float pow10f(float x) {
//return powf(10.0f,x) //standard, but slower
return expf(2.30258509f*x); //faster: exp(log(10.0f)*x)
}
/* See https://github.com/Tympan/Tympan_Library/blob/master/src/AudioCalcGainWDRC_F32.h
* Dr Paul Beckmann
* 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. Y is log2(X)
*/
float v2DB_Approx(float volts) {
float Y, F;
int E;
// This is the approximation to log2()
F = frexpf(volts, &E); // first separate power of 2;
// Y = C[0]*F*F*F + C[1]*F*F + C[2]*F + C[3] + E;
Y = 1.23149591; //C[0]
Y *= F;
Y += -4.11852516f; //C[1]
Y *= F;
Y += 6.02197014f; //C[2]
Y *= F;
Y += -3.13396450f; //C[3]
Y += E;
// Convert to dB = 20 Log10(volts)
return 6.020599f * Y; // (20.0f/log2(10.0))*Y;
}
private:
audio_block_f32_t *inputQueueArray[1];
float delayData[256]; // The circular delay line for the signal
uint16_t in_index = 0; // Pointer to next block update entry
// And a mask to make the circular buffer limit to a power of 2
uint16_t delayBufferMask = 0X00FF;
uint16_t delaySize = 256;
float sample_rate_Hz = 44100;
float attackSec = 0.005f; // Q: Can this be reduced with the delay line added to the signal path??
float releaseSec = 0.100f;
// This alpha, beta for 5 ms attack, 100ms release, about 0.07 dB max ripple at 1000 Hz
float alpha = 0.98912216f;
float oneMinusAlpha = 0.01087784f;
float beta = 0.9995961f;
// Presets here should be studied/experimented with for each application
float gain0DB = 38.0f; // Gain, in dB for low level inputs
float gainOffsetDB = 0.0f; // Raise/lower entire gain curve by this amount (post gain)
float knee1DB = -50.0f; // First knee on the gain curve
float cr1 = 3.0f; // Compression ratio on dB curve between knee1dB and knee2dB
float knee2DB = -20.0f; // Second knee on the gain curve
float cr2 = 10.0f; // Compression ratio on dB curve above knee2dB
float vPeakSave = 0.0f;
bool printIO = false; // Diagnostics Only
float sampleInputDB, sampleGainDB;
};
#endif