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334 lines
16 KiB

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.
#ifndef _AudioEffectCompressor_F32
#define _AudioEffectCompressor_F32
#include <arm_math.h> //ARM DSP extensions.
#include <AudioStream_F32.h>
class AudioEffectCompressor_F32 : public AudioStream_F32
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
AudioEffectCompressor_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {
setDefaultValues(AUDIO_SAMPLE_RATE); resetStates();
AudioEffectCompressor_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) {
setDefaultValues(settings.sample_rate_Hz); resetStates();
void setDefaultValues(const float 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
//here's the method that does all the work
void update(void) {
//Serial.println("AudioEffectGain_F32: updating."); //for debugging.
audio_block_f32_t *audio_block = AudioStream_F32::receiveWritable_f32();
if (!audio_block) 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->data, audio_block->data, audio_block->length);
//apply the pre-gain...a negative gain value will disable
if (pre_gain > 0.0f) arm_scale_f32(audio_block->data, pre_gain, audio_block->data, audio_block->length); //use ARM DSP for speed!
//calculate the level of the audio (ie, calculate a smoothed version of the signal power)
audio_block_f32_t *audio_level_dB_block = AudioStream_F32::allocate_f32();
calcAudioLevel_dB(audio_block, audio_level_dB_block); //returns through audio_level_dB_block
//compute the desired gain based on the observed audio level
audio_block_f32_t *gain_block = AudioStream_F32::allocate_f32();
calcGain(audio_level_dB_block, gain_block); //returns through gain_block
//apply the desired the processed audio back into audio_block
arm_mult_f32(audio_block->data, gain_block->data, audio_block->data, audio_block->length);
//transmit the block and release memory
// 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 calcAudioLevel_dB(audio_block_f32_t *wav_block, audio_block_f32_t *level_dB_block) {
// calculate the instantaneous signal power (square the signal)
audio_block_f32_t *wav_pow_block = AudioStream_F32::allocate_f32();
arm_mult_f32(wav_block->data, wav_block->data, wav_pow_block->data, wav_block->length);
// low-pass filter and convert to dB
float c1 = level_lp_const, c2 = 1.0f - c1; //prepare constants
for (int i = 0; i < wav_pow_block->length; i++) {
// first-order low-pass filter to get a running estimate of the average power
wav_pow_block->data[i] = c1*prev_level_lp_pow + c2*wav_pow_block->data[i];
// save the state of the first-order low-pass filter
prev_level_lp_pow = wav_pow_block->data[i];
//now convert the signal power to dB (but not yet multiplied by 10.0)
level_dB_block->data[i] = log10f_approx(wav_pow_block->data[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->data, 10.0f, level_dB_block->data, level_dB_block->length); //use ARM DSP for speed!
//release memory and return
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 calcGain(audio_block_f32_t *audio_level_dB_block, audio_block_f32_t *gain_block) {
//first, calculate the instantaneous target gain based on the compression ratio
audio_block_f32_t *inst_targ_gain_dB_block = AudioStream_F32::allocate_f32();
calcInstantaneousTargetGain(audio_level_dB_block, inst_targ_gain_dB_block);
//second, smooth in time (attack and release) by stepping through each sample
audio_block_f32_t *gain_dB_block = AudioStream_F32::allocate_f32();
//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->data, 1.0f/20.0f, gain_dB_block->data, gain_dB_block->length); //divide by 20
for (int i = 0; i < gain_dB_block->length; i++) gain_block->data[i] = pow10f(gain_dB_block->data[i]); //do the 10^(x)
//release memory and return
return; //output is passed through gain_block
//Compute the instantaneous desired gain, including the compression ratio and
//threshold for where the comrpession kicks in
void calcInstantaneousTargetGain(audio_block_f32_t *audio_level_dB_block, audio_block_f32_t *inst_targ_gain_dB_block) {
// how much are we above the compression threshold?
audio_block_f32_t *above_thresh_dB_block = AudioStream_F32::allocate_f32();
arm_offset_f32(audio_level_dB_block->data, //CMSIS DSP for "add a constant value to all elements"
-thresh_dBFS, //this is the value to be added
above_thresh_dB_block->data, //this is the output
// 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->data, //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->data, //this is the output
// compute the instantaneous gain...which is the difference between the target level and the original level
arm_sub_f32(inst_targ_gain_dB_block->data, //CMSIS DSP for "subtract two vectors element-by-element"
above_thresh_dB_block->data, //this is the vector to be subtracted
inst_targ_gain_dB_block->data, //this is the output
// limit the target gain to attenuation only (this part of the compressor should not make things louder!)
for (int i=0; i < inst_targ_gain_dB_block->length; i++) {
if (inst_targ_gain_dB_block->data[i] > 0.0f) inst_targ_gain_dB_block->data[i] = 0.0f;
// release memory before returning
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 calcSmoothedGain_dB(audio_block_f32_t *inst_targ_gain_dB_block, audio_block_f32_t *gain_dB_block) {
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 (int i = 0; i < inst_targ_gain_dB_block->length; i++) {
gain_dB = inst_targ_gain_dB_block->data[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->data[i] = attack_const*prev_gain_dB + one_minus_attack_const*gain_dB;
} else { //or, we're in the release phase
gain_dB_block->data[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->data[i];
return; //the output here is gain_block
//methods to set parameters of this module
void 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 setPreGain(float g) { pre_gain = g; }
void setPreGain_dB(float gain_dB) { setPreGain(pow(10.0, gain_dB / 20.0)); }
void setCompressionRatio(float cr) {
comp_ratio = max(0.001, cr); //limit to positive values
void setAttack_sec(float a, float 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(min(attack_sec,release_sec) / 5.0, fs_Hz); //make the level time-constant one-fifth the gain time constants
void setRelease_sec(float r, float 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(min(attack_sec,release_sec) / 5.0, fs_Hz); //make the level time-constant one-fifth the gain time constants
void setLevelTimeConst_sec(float t_sec, float fs_Hz) {
const float min_t_sec = 0.002f; //this is the minimum allowed value
level_lp_sec = max(min_t_sec,t_sec);
level_lp_const = expf(-1.0f / (level_lp_sec * fs_Hz)); //expf() is much faster than exp()
void setThresh_dBFS(float val) {
thresh_dBFS = val;
setThreshPow(pow(10.0, thresh_dBFS / 10.0));
void enableHPFilter(boolean flag) { use_HP_prefilter = flag; };
//methods to return information about this module
float32_t getPreGain_dB(void) { return 20.0 * log10f_approx(pre_gain); }
float32_t getAttack_sec(void) { return attack_sec; }
float32_t getRelease_sec(void) { return release_sec; }
float32_t getLevelTimeConst_sec(void) { return level_lp_sec; }
float32_t getThresh_dBFS(void) { return thresh_dBFS; }
float32_t getCompressionRatio(void) { return comp_ratio; }
float32_t getCurrentLevel_dBFS(void) { return 10.0* log10f_approx(prev_level_lp_pow); }
float32_t getCurrentGain_dB(void) { return prev_gain_dB; }
void setHPFilterCoeff_N2IIR_Matlab(float32_t b[], float32_t a[]){
//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
//state-related variables
audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
float32_t prev_level_lp_pow = 1.0;
float32_t prev_gain_dB = 0.0; //last gain^2 used
//HP filter state-related variables
arm_biquad_casd_df1_inst_f32 hp_filt_struct;
static const uint8_t hp_nstages = 1;
float32_t hp_coeff[5 * hp_nstages] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later
float32_t hp_state[4 * hp_nstages];
void setHPFilterCoeff(void) {
//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
//private parameters related to gain calculation
float32_t attack_const, release_const, level_lp_const; //used in calcGain(). set by setAttack_sec() and setRelease_sec();
float32_t comp_ratio_const, thresh_pow_FS_wCR; //used in calcGain(); set in updateThresholdAndCompRatioConstants()
void updateThresholdAndCompRatioConstants(void) {
comp_ratio_const = 1.0f-(1.0f / comp_ratio);
thresh_pow_FS_wCR = powf(thresh_pow_FS, comp_ratio_const);
float32_t attack_sec, release_sec, level_lp_sec;
float32_t thresh_dBFS = 0.0; //threshold for compression, relative to digital full scale
float32_t thresh_pow_FS = 1.0f; //same as above, but not in dB
void setThreshPow(float t_pow) {
thresh_pow_FS = t_pow;
float32_t comp_ratio = 1.0; //compression ratio
float32_t pre_gain = -1.0; //gain to apply before the compression. negative value disables
boolean use_HP_prefilter;
// Accelerate the powf(10.0,x) function
static float32_t pow10f(float 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(float 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.
** ------------------------------------------------------------------- */
//float log2f_approx_coeff[4] = {1.23149591368684f, -4.11852516267426f, 6.02197014179219f, -3.13396450166353f};
static 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 = 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;