/* From https://github.com/chipaudette/OpenAudio_ArduinoLibrary */
/*
AudioEffectCompressor
Created: Chip Audette, Dec 2016 - Jan 2017
Purpose; Apply dynamic range compression to the audio stream.
Assumes floating-point data.
This processes a single stream fo audio data (ie, it is mono)
MIT License. use at your own risk.
*/
#include <algorithm>
#include <circle/logger.h>
#include <cstdlib>
#include "effect_compressor.h"
LOGMODULE ("compressor");
Compressor::Compressor(const float32_t sample_rate_Hz) {
//setDefaultValues(AUDIO_SAMPLE_RATE); resetStates();
setDefaultValues(sample_rate_Hz);
resetStates();
}
void Compressor::setDefaultValues(const float32_t sample_rate_Hz) {
setThresh_dBFS(-20.0f); //set the default value for the threshold for compression
setCompressionRatio(5.0f); //set the default copression ratio
setAttack_sec(0.005f, sample_rate_Hz); //default to this value
setRelease_sec(0.200f, sample_rate_Hz); //default to this value
setHPFilterCoeff(); enableHPFilter(true); //enable the HP filter to remove any DC offset from the audio
}
//Compute the instantaneous desired gain, including the compression ratio and
//threshold for where the comrpession kicks in
void Compressor::calcInstantaneousTargetGain(float32_t *audio_level_dB_block, float32_t *inst_targ_gain_dB_block, uint16_t len)
{
// how much are we above the compression threshold?
float32_t above_thresh_dB_block[len];
//arm_copy_f32(zeroblock_f32,above_thresh_dB_block,len);
arm_offset_f32(audio_level_dB_block, //CMSIS DSP for "add a constant value to all elements"
-thresh_dBFS, //this is the value to be added
above_thresh_dB_block, //this is the output
len);
// scale by the compression ratio...this is what the output level should be (this is our target level)
arm_scale_f32(above_thresh_dB_block, //CMSIS DSP for "multiply all elements by a constant value"
1.0f / comp_ratio, //this is the value to be multiplied
inst_targ_gain_dB_block, //this is the output
len);
// compute the instantaneous gain...which is the difference between the target level and the original level
arm_sub_f32(inst_targ_gain_dB_block, //CMSIS DSP for "subtract two vectors element-by-element"
above_thresh_dB_block, //this is the vector to be subtracted
inst_targ_gain_dB_block, //this is the output
len);
// limit the target gain to attenuation only (this part of the compressor should not make things louder!)
for (uint16_t i=0; i < len; i++) {
if (inst_targ_gain_dB_block[i] > 0.0f) inst_targ_gain_dB_block[i] = 0.0f;
}
return; //output is passed through inst_targ_gain_dB_block
}
//this method applies the "attack" and "release" constants to smooth the
//target gain level through time.
void Compressor::calcSmoothedGain_dB(float32_t *inst_targ_gain_dB_block, float32_t *gain_dB_block, uint16_t len)
{
float32_t gain_dB;
float32_t one_minus_attack_const = 1.0f - attack_const;
float32_t one_minus_release_const = 1.0f - release_const;
for (uint16_t i = 0; i < len; i++) {
gain_dB = inst_targ_gain_dB_block[i];
//smooth the gain using the attack or release constants
if (gain_dB < prev_gain_dB) { //are we in the attack phase?
gain_dB_block[i] = attack_const*prev_gain_dB + one_minus_attack_const*gain_dB;
} else { //or, we're in the release phase
gain_dB_block[i] = release_const*prev_gain_dB + one_minus_release_const*gain_dB;
}
//save value for the next time through this loop
prev_gain_dB = gain_dB_block[i];
}
return; //the output here is gain_block
}
// Here's the method that estimates the level of the audio (in dB)
// It squares the signal and low-pass filters to get a time-averaged
// signal power. It then
void Compressor::calcAudioLevel_dB(float32_t *wav_block, float32_t *level_dB_block, uint16_t len) {
// calculate the instantaneous signal power (square the signal)
float32_t wav_pow_block[len];
//arm_copy_f32(zeroblock_f32,wav_pow_block,len);
arm_mult_f32(wav_block, wav_block, wav_pow_block, len);
// low-pass filter and convert to dB
float32_t c1 = level_lp_const, c2 = 1.0f - c1; //prepare constants
for (uint16_t i = 0; i < len; i++) {
// first-order low-pass filter to get a running estimate of the average power
wav_pow_block[i] = c1*prev_level_lp_pow + c2*wav_pow_block[i];
// save the state of the first-order low-pass filter
prev_level_lp_pow = wav_pow_block[i];
//now convert the signal power to dB (but not yet multiplied by 10.0)
level_dB_block[i] = log10f_approx(wav_pow_block[i]);
}
//limit the amount that the state of the smoothing filter can go toward negative infinity
if (prev_level_lp_pow < (1.0E-13)) prev_level_lp_pow = 1.0E-13; //never go less than -130 dBFS
//scale the wav_pow_block by 10.0 to complete the conversion to dB
arm_scale_f32(level_dB_block, 10.0f, level_dB_block, len); //use ARM DSP for speed!
return; //output is passed through level_dB_block
}
//This method computes the desired gain from the compressor, given an estimate
//of the signal level (in dB)
void Compressor::calcGain(float32_t *audio_level_dB_block, float32_t *gain_block,uint16_t len)
{
//first, calculate the instantaneous target gain based on the compression ratio
float32_t inst_targ_gain_dB_block[len];
//arm_copy_f32(zeroblock_f32,inst_targ_gain_dB_block,len);
calcInstantaneousTargetGain(audio_level_dB_block, inst_targ_gain_dB_block,len);
//second, smooth in time (attack and release) by stepping through each sample
float32_t gain_dB_block[len];
//arm_copy_f32(zeroblock_f32,gain_dB_block,len);
calcSmoothedGain_dB(inst_targ_gain_dB_block,gain_dB_block, len);
//finally, convert from dB to linear gain: gain = 10^(gain_dB/20); (ie this takes care of the sqrt, too!)
arm_scale_f32(gain_dB_block, 1.0f/20.0f, gain_dB_block, len); //divide by 20
for (uint16_t i = 0; i < len; i++) gain_block[i] = pow10f(gain_dB_block[i]); //do the 10^(x)
return; //output is passed through gain_block
}
//here's the method that does all the work
void Compressor::doCompression(float32_t *audio_block, uint16_t len) {
//Serial.println("AudioEffectGain_F32: updating."); //for debugging.
if (!audio_block) {
LOGERR("No audio_block available for Compressor!");
return;
}
//apply a high-pass filter to get rid of the DC offset
if (use_HP_prefilter)
arm_biquad_cascade_df1_f32(&hp_filt_struct, audio_block, audio_block, len);
//apply the pre-gain...a negative gain value will disable
if (pre_gain > 0.0f)
arm_scale_f32(audio_block, pre_gain, audio_block, len); //use ARM DSP for speed!
//calculate the level of the audio (ie, calculate a smoothed version of the signal power)
float32_t audio_level_dB_block[len];
//arm_copy_f32(zeroblock_f32,audio_level_dB_block,len);
calcAudioLevel_dB(audio_block, audio_level_dB_block, len); //returns through audio_level_dB_block
//compute the desired gain based on the observed audio level
float32_t gain_block[len];
//arm_copy_f32(zeroblock_f32,gain_block,len);
calcGain(audio_level_dB_block, gain_block, len); //returns through gain_block
//apply the desired gain...store the processed audio back into audio_block
arm_mult_f32(audio_block, gain_block, audio_block, len);
}
//methods to set parameters of this module
void Compressor::resetStates(void)
{
prev_level_lp_pow = 1.0f;
prev_gain_dB = 0.0f;
//initialize the HP filter. (This also resets the filter states,)
arm_biquad_cascade_df1_init_f32(&hp_filt_struct, hp_nstages, hp_coeff, hp_state);
}
void Compressor::setPreGain(float32_t g)
{
pre_gain = g;
}
void Compressor::setPreGain_dB(float32_t gain_dB)
{
setPreGain(pow(10.0, gain_dB / 20.0));
}
void Compressor::setCompressionRatio(float32_t cr)
{
comp_ratio = std::max(0.001f, cr); //limit to positive values
updateThresholdAndCompRatioConstants();
}
void Compressor::setAttack_sec(float32_t a, float32_t fs_Hz)
{
attack_sec = a;
attack_const = expf(-1.0f / (attack_sec * fs_Hz)); //expf() is much faster than exp()
//also update the time constant for the envelope extraction
setLevelTimeConst_sec(std::min(attack_sec,release_sec) / 5.0, fs_Hz); //make the level time-constant one-fifth the gain time constants
}
void Compressor::setRelease_sec(float32_t r, float32_t fs_Hz)
{
release_sec = r;
release_const = expf(-1.0f / (release_sec * fs_Hz)); //expf() is much faster than exp()
//also update the time constant for the envelope extraction
setLevelTimeConst_sec(std::min(attack_sec,release_sec) / 5.0, fs_Hz); //make the level time-constant one-fifth the gain time constants
}
void Compressor::setLevelTimeConst_sec(float32_t t_sec, float32_t fs_Hz)
{
const float32_t min_t_sec = 0.002f; //this is the minimum allowed value
level_lp_sec = std::max(min_t_sec,t_sec);
level_lp_const = expf(-1.0f / (level_lp_sec * fs_Hz)); //expf() is much faster than exp()
}
void Compressor::setThresh_dBFS(float32_t val)
{
thresh_dBFS = val;
setThreshPow(pow(10.0, thresh_dBFS / 10.0));
}
void Compressor::enableHPFilter(boolean flag)
{
use_HP_prefilter = flag;
}
void Compressor::setHPFilterCoeff_N2IIR_Matlab(float32_t b[], float32_t a[])
{
//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
//Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
hp_coeff[0] = b[0]; hp_coeff[1] = b[1]; hp_coeff[2] = b[2]; //here are the matlab "b" coefficients
hp_coeff[3] = -a[1]; hp_coeff[4] = -a[2]; //the DSP needs the "a" terms to have opposite sign vs Matlab
}
void Compressor::setHPFilterCoeff(void)
{
//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
//Use matlab to compute the coeff for HP at 20Hz: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
float32_t b[] = {9.979871156751189e-01, -1.995974231350238e+00, 9.979871156751189e-01}; //from Matlab
float32_t a[] = { 1.000000000000000e+00, -1.995970179642828e+00, 9.959782830576472e-01}; //from Matlab
setHPFilterCoeff_N2IIR_Matlab(b, a);
//hp_coeff[0] = b[0]; hp_coeff[1] = b[1]; hp_coeff[2] = b[2]; //here are the matlab "b" coefficients
//hp_coeff[3] = -a[1]; hp_coeff[4] = -a[2]; //the DSP needs the "a" terms to have opposite sign vs Matlab
}
void Compressor::updateThresholdAndCompRatioConstants(void)
{
comp_ratio_const = 1.0f-(1.0f / comp_ratio);
thresh_pow_FS_wCR = powf(thresh_pow_FS, comp_ratio_const);
}
void Compressor::setThreshPow(float32_t t_pow)
{
thresh_pow_FS = t_pow;
updateThresholdAndCompRatioConstants();
}
// Accelerate the powf(10.0,x) function
static float32_t pow10f(float32_t x)
{
//return powf(10.0f,x) //standard, but slower
return expf(2.302585092994f*x); //faster: exp(log(10.0f)*x)
}
// Accelerate the log10f(x) function?
static float32_t log10f_approx(float32_t x)
{
//return log10f(x); //standard, but slower
return log2f_approx(x)*0.3010299956639812f; //faster: log2(x)/log2(10)
}
/* ----------------------------------------------------------------------
** 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.
** ------------------------------------------------------------------- */
//https://community.arm.com/tools/f/discussions/4292/cmsis-dsp-new-functionality-proposal/22621#22621
//float32_t log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
static float32_t log2f_approx(float32_t X)
{
//float32_t *C = &log2f_approx_coeff[0];
float32_t Y;
float32_t 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 = 1.23149591368684f;
Y *= F;
//Y += (*C++);
Y += -4.11852516267426f;
Y *= F;
//Y += (*C++);
Y += 6.02197014179219f;
Y *= F;
//Y += (*C++);
Y += -3.13396450166353f;
Y += E;
return(Y);
}