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AddWDRC2 object and example

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boblark 4 years ago
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  1. 272
      AudioEffectWDRC2_F32.h
  2. BIN
      examples/testWDRC2/WDRC2Transients.gif
  3. 203
      examples/testWDRC2/testWDRC2.ino

272
AudioEffectWDRC2_F32.h
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/*
* 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

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examples/testWDRC2/WDRC2Transients.gif
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203
examples/testWDRC2/testWDRC2.ino
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/* TestWDRC2.ino Bob Larkin 8 Dec 2020
*
* Test of AudioEffectWDRC2_F32 (Wide Dynamic Range Compressor)
* See AudioEffectWDRC2_F32.h for much detail and explanation.
* Choice of test signals is a single sine wave, a random sequence
* of sine waves of varying frequency and amplitude, a power
* sweep or a pulse of sine wave to see transient behavior.
*
* This version is for the Chip Audette OpenAudio_F32 Library. and
* thus has that interface structure.
*
* NOTE: As of 20 Dec 2020, the compressor AudioEffectWDRC2_F32.h
* was not finalized and could change in detail. Use here with
* this in mind.
*/
#include "Audio.h"
#include "OpenAudio_ArduinoLibrary.h"
#include "AudioEffectWDRC2_F32.h"
#define CW 0
#define RANDOM 1
#define POWER_SWEEP 2
#define PULSE 3
// Edit in one of the last four, here:
#define SIGNAL_SOURCE PULSE
AudioSynthWaveformSine_F32 sine1; // Test signal
AudioPlayQueue_F32 queue0; // Amplitude set of input
AudioMultiply_F32 mult1;
AudioEffectWDRC2_F32 compressor1; // Wide Dynamic Range Compressor
AudioFilterFIR_F32 fir;
AudioEffectGain_F32 gain0; // Sets volume sent to output
AudioEffectGain_F32 gain1; // Sets the same
AudioConvert_F32toI16 convert0; // Allow integer output driver
AudioConvert_F32toI16 convert1;
AudioOutputI2S i2sOut;
AudioConnection_F32 patchCord0(sine1, 0, mult1, 0);
AudioConnection_F32 patchCord1(queue0, 0, mult1, 1);
AudioConnection_F32 patchCord2(mult1, 0, fir, 0);
AudioConnection_F32 patchCord3(fir, 0, compressor1, 0);
AudioConnection_F32 patchCord4(compressor1, 0, gain0, 0);
AudioConnection_F32 patchCord5(fir, 0, gain1, 0);
AudioConnection_F32 patchCord6(gain0, 0, convert0, 0);
AudioConnection_F32 patchCord7(gain1, 0, convert1, 0);
AudioConnection patchCord8(convert0, 0, i2sOut, 0);
AudioConnection patchCord9(convert1, 0, i2sOut, 1);
AudioControlSGTL5000 sgtl5000_1;
uint16_t count17, count27;
float level = 0.05f;
void setup(void) {
AudioMemory(50);
AudioMemory_F32(100);
Serial.begin(300); delay(1000);
Serial.println("*** Test WDRC2 Gain Compressor **");
sine1.frequency(1000.0f);
sine1.amplitude(0.05f);
// CAUTION - If using ears on the output, adjust the following two carefully
gain0.setGain_dB(-25.0f); // Consider (-50.0f);
gain1.setGain_dB(13.0f); // Consider (-30.0f);
sgtl5000_1.enable();
// Fir Filter needs coefs, now it ts just a pass through.
count17 = 0;
count27 = 0;
#if 0
// For reference, here are the defaults from AudioEffectsWDRC_F32.h
int16_t delaySize = 256; // Any power of 2, i.e., 256, 128, 64, etc.
float gain0DB = 38.0f; // Gain, in dB for low level inputs (dependent variable)
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
#endif
// Edit the following to change settings
// Note: gain0 is a dependent variable, and not available as an input
compressor1.setDelayBufferSize(128);
compressor1.setOutputGainOffsetDB(0.0f);
compressor1.setKnee1LowDB(-50.0f);
compressor1.setCompressionRatioMiddleDB(3.0f);
compressor1.setKnee2HighDB(-20.0f);
compressor1.setCompressionRatioHighDB(10.0f);
}
void loop(void)
{
float32_t* pBuff;
static uint16_t kk;
#if SIGNAL_SOURCE == CW
// Literally Continuous Wave. Edit frequency and amplitude below
pBuff = queue0.getBuffer();
if (pBuff)
{
if(count27++ == 227) // about 0.7 sec
{
sine1.frequency(1000.0f); // <--
sine1.amplitude(0.01f); // <--
Serial.print(" LowLevDB= "); Serial.print( compressor1.getLowLevelGainDB(), 3);
Serial.print(" CurInDB= "); Serial.print( compressor1.getCurrentInputDB(), 3);
Serial.print(" CurrentGainDB= "); Serial.println( compressor1.getCurrentGainDB(), 3);
count27 = 0;
}
// Multiply by 1.0 by filling queue1
for(int ii=0; ii<128; ii++)
*(pBuff + ii) = 1.0f; // Fill buffer with constant level
queue0.playBuffer(); // Starr up new 128 values
}
#elif SIGNAL_SOURCE == RANDOM
/* To give an audio signal with interest, we alter the frequency
* every 17 blocks (49 msec) and alter the level every 27 b;ocks
* (78.4 msec) The pattern keeps changing to be more interesting
* Janet thinks it is aliens. */
pBuff = queue0.getBuffer();
if (pBuff)
{
Serial.print(" CurInDB= "); Serial.print( compressor1.getCurrentInputDB(), 3);
Serial.print(" CurrentGainDB= "); Serial.println( compressor1.getCurrentGainDB(), 3);
if(count17++ == 17)
{
// Put a delay in, like between words
if(randUniform() < 0.03)
delay( (int)(1500.0*randUniform()) );
count17 = 0;
float ff = 350.0f + 700.0f*sqrtf( randUniform() );
sine1.frequency(ff); //Serial.println(ff);
}
if(count27++ == 27)
{
count27 = 0;
level = 1.0f*powf( randUniform(), 2 ); // 0 to 1, emphasizing 0 end
}
for(int ii=0; ii<128; ii++)
*(pBuff + ii) = level; // Fill buffer with constant level
queue0.playBuffer(); // Starr up new 128 values
}
#elif SIGNAL_SOURCE == POWER_SWEEP
pBuff = queue0.getBuffer();
if (pBuff)
{
if(count17++ == 17)
{
count17 = 0;
level *= 1.05f;
if(level > 0.99)
{
level=0.001;
delay(200);
}
Serial.print(" CurInDB= "); Serial.print( compressor1.getCurrentInputDB(), 3);
Serial.print(" CurrentGainDB= "); Serial.println( compressor1.getCurrentGainDB(), 3);
}
for(int ii=0; ii<128; ii++)
*(pBuff + ii) = level;
queue0.playBuffer();
}
#elif SIGNAL_SOURCE == PULSE
pBuff = queue0.getBuffer();
if (pBuff)
{
for(int ii=0; ii<128; ii++)
*(pBuff + ii) = 1.0f;
queue0.playBuffer();
// A pulse, repeats every 3 minutes or so
if(count17 == 5) sine1.amplitude(0.0f); // Settling
else if(count17 == 498) compressor1.printOn(true); //record it
else if(count17 == 500) sine1.amplitude(0.03f);
else if(count17 == 510) sine1.amplitude(0.0f);
else if(count17 == 700) compressor1.printOn(false);
// or build your own transient test pulse here
count17++;
}
#endif
}
/* randUniform() - Returns random number, uniform on (0, 1)
* The "Even Quicker" uniform random sample generator from D. E. Knuth and
* H. W. Lewis and described in Chapter 7 of "Numerical Receipes in C",
* 2nd ed, with the comment "this is about as good as any 32-bit linear
* congruential generator, entirely adequate for many uses."
*/
#define FL_ONE 0X3F800000
#define FL_MASK 0X007FFFFF
float randUniform(void)
{
static uint32_t idum = 12345;
union {
uint32_t i32;
float f32;
} uinf;
idum = (uint32_t)1664525 * idum + (uint32_t)1013904223;
// return (*(float *)&it); // Cute convert to float but gets compiler warning
uinf.i32 = FL_ONE | (FL_MASK & idum); // So do the same thing with a union
return uinf.f32 - 1.0f;
}
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