/*
* AudioFilterBiquad_F32
*
* Created: Chip Audette (OpenAudio) Feb 2017
*
* License: MIT License. Use at your own risk.
*
*/
#ifndef _filter_iir_f32
#define _filter_iir_f32
#include "Arduino.h"
#include "AudioStream_F32.h"
#include "arm_math.h"
// Indicates that the code should just pass through the audio
// without any filtering (as opposed to doing nothing at all)
#define IIR_F32_PASSTHRU ((const float32_t *) 1)
#define IIR_MAX_STAGES 1 //meaningless right now
class AudioFilterBiquad_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:IIR
public:
AudioFilterBiquad_F32(void): AudioStream_F32(1,inputQueueArray), coeff_p(IIR_F32_PASSTHRU) {
setSampleRate_Hz(AUDIO_SAMPLE_RATE_EXACT);
}
AudioFilterBiquad_F32(const AudioSettings_F32 &settings):
AudioStream_F32(1,inputQueueArray), coeff_p(IIR_F32_PASSTHRU) {
setSampleRate_Hz(settings.sample_rate_Hz);
}
void begin(const float32_t *cp, int n_stages = 1) {
coeff_p = cp;
// Initialize FIR instance (ARM DSP Math Library)
if (coeff_p && (coeff_p != IIR_F32_PASSTHRU) && n_stages <= IIR_MAX_STAGES) {
//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html
arm_biquad_cascade_df1_init_f32(&iir_inst, n_stages, (float32_t *)coeff_p, &StateF32[0]);
}
}
void end(void) {
coeff_p = NULL;
}
void setSampleRate_Hz(float _fs_Hz) { sampleRate_Hz = _fs_Hz; }
void setBlockDC(void) {
//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
//Use matlab to compute the coeff for HP at 40Hz: [b,a]=butter(2,40/(44100/2),'high'); %assumes fs_Hz = 44100
float32_t b[] = {8.173653471988667e-01, -1.634730694397733e+00, 8.173653471988667e-01}; //from Matlab
float32_t a[] = { 1.000000000000000e+00, -1.601092394183619e+00, 6.683689946118476e-01}; //from Matlab
setFilterCoeff_Matlab(b, a);
}
void setFilterCoeff_Matlab(float32_t b[], float32_t a[]) { //one stage of N=2 IIR
//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
//Use matlab to compute the coeff, such as: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
coeff[0] = b[0]; coeff[1] = b[1]; coeff[2] = b[2]; //here are the matlab "b" coefficients
coeff[3] = -a[1]; coeff[4] = -a[2]; //the DSP needs the "a" terms to have opposite sign vs Matlab ;
begin(coeff);
}
//note: stage is currently ignored
void setCoefficients(int stage, float c[]) {
if (stage > 0) {
if (Serial) {
Serial.println(F("AudioFilterBiquad_F32: setCoefficients: *** ERROR ***"));
Serial.print(F(" : This module only accepts one stage."));
Serial.print(F(" : You are attempting to set stage "));Serial.print(stage);
Serial.print(F(" : Ignoring this filter."));
}
return;
}
coeff[0] = c[0];
coeff[1] = c[1];
coeff[2] = c[2];
coeff[3] = -c[3];
coeff[4] = -c[4];
begin(coeff);
}
// Compute common filter functions
// http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
//void setLowpass(uint32_t stage, float frequency, float q = 0.7071) {
void setLowpass(uint32_t stage, float frequency, float q = 0.7071) {
//int coeff[5];
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
double alpha = sinW0 / ((double)q * 2.0);
double cosW0 = cos(w0);
//double scale = 1073741824.0 / (1.0 + alpha);
double scale = 1.0 / (1.0+alpha); // which is equal to 1.0 / a0
/* b0 */ coeff[0] = ((1.0 - cosW0) / 2.0) * scale;
/* b1 */ coeff[1] = (1.0 - cosW0) * scale;
/* b2 */ coeff[2] = coeff[0];
/* a1 */ coeff[3] = (-2.0 * cosW0) * scale;
/* a2 */ coeff[4] = (1.0 - alpha) * scale;
setCoefficients(stage, coeff);
}
void setHighpass(uint32_t stage, float frequency, float q = 0.7071) {
//int coeff[5];
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
double alpha = sinW0 / ((double)q * 2.0);
double cosW0 = cos(w0);
double scale = 1.0 / (1.0+alpha); // which is equal to 1.0 / a0
/* b0 */ coeff[0] = ((1.0 + cosW0) / 2.0) * scale;
/* b1 */ coeff[1] = -(1.0 + cosW0) * scale;
/* b2 */ coeff[2] = coeff[0];
/* a1 */ coeff[3] = (-2.0 * cosW0) * scale;
/* a2 */ coeff[4] = (1.0 - alpha) * scale;
setCoefficients(stage, coeff);
}
void setBandpass(uint32_t stage, float frequency, float q = 1.0) {
//int coeff[5];
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
double alpha = sinW0 / ((double)q * 2.0);
double cosW0 = cos(w0);
double scale = 1.0 / (1.0+alpha); // which is equal to 1.0 / a0
/* b0 */ coeff[0] = alpha * scale;
/* b1 */ coeff[1] = 0;
/* b2 */ coeff[2] = (-alpha) * scale;
/* a1 */ coeff[3] = (-2.0 * cosW0) * scale;
/* a2 */ coeff[4] = (1.0 - alpha) * scale;
setCoefficients(stage, coeff);
}
void setNotch(uint32_t stage, float frequency, float q = 1.0) {
//int coeff[5];
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
double alpha = sinW0 / ((double)q * 2.0);
double cosW0 = cos(w0);
double scale = 1.0 / (1.0+alpha); // which is equal to 1.0 / a0
/* b0 */ coeff[0] = scale;
/* b1 */ coeff[1] = (-2.0 * cosW0) * scale;
/* b2 */ coeff[2] = coeff[0];
/* a1 */ coeff[3] = (-2.0 * cosW0) * scale;
/* a2 */ coeff[4] = (1.0 - alpha) * scale;
setCoefficients(stage, coeff);
}
void setLowShelf(uint32_t stage, float frequency, float gain, float slope = 1.0f) {
//int coeff[5];
double a = pow(10.0, gain/40.0);
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
//double alpha = (sinW0 * sqrt((a+1/a)*(1/slope-1)+2) ) / 2.0;
double cosW0 = cos(w0);
//generate three helper-values (intermediate results):
double sinsq = sinW0 * sqrt( (pow(a,2.0)+1.0)*(1.0/slope-1.0)+2.0*a );
double aMinus = (a-1.0)*cosW0;
double aPlus = (a+1.0)*cosW0;
double scale = 1.0 / ( (a+1.0) + aMinus + sinsq);
/* b0 */ coeff[0] = a * ( (a+1.0) - aMinus + sinsq ) * scale;
/* b1 */ coeff[1] = 2.0*a * ( (a-1.0) - aPlus ) * scale;
/* b2 */ coeff[2] = a * ( (a+1.0) - aMinus - sinsq ) * scale;
/* a1 */ coeff[3] = -2.0* ( (a-1.0) + aPlus ) * scale;
/* a2 */ coeff[4] = ( (a+1.0) + aMinus - sinsq ) * scale;
setCoefficients(stage, coeff);
}
void setHighShelf(uint32_t stage, float frequency, float gain, float slope = 1.0f) {
//int coeff[5];
double a = pow(10.0, gain/40.0);
double w0 = frequency * (2 * 3.141592654 / AUDIO_SAMPLE_RATE_EXACT);
double sinW0 = sin(w0);
//double alpha = (sinW0 * sqrt((a+1/a)*(1/slope-1)+2) ) / 2.0;
double cosW0 = cos(w0);
//generate three helper-values (intermediate results):
double sinsq = sinW0 * sqrt( (pow(a,2.0)+1.0)*(1.0/slope-1.0)+2.0*a );
double aMinus = (a-1.0)*cosW0;
double aPlus = (a+1.0)*cosW0;
double scale = 1.0 / ( (a+1.0) - aMinus + sinsq);
/* b0 */ coeff[0] = a * ( (a+1.0) + aMinus + sinsq ) * scale;
/* b1 */ coeff[1] = -2.0*a * ( (a-1.0) + aPlus ) * scale;
/* b2 */ coeff[2] = a * ( (a+1.0) + aMinus - sinsq ) * scale;
/* a1 */ coeff[3] = 2.0* ( (a-1.0) - aPlus ) * scale;
/* a2 */ coeff[4] = ( (a+1.0) - aMinus - sinsq ) * scale;
setCoefficients(stage, coeff);
}
void update(void);
private:
audio_block_f32_t *inputQueueArray[1];
float32_t coeff[5 * 1] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later
float sampleRate_Hz = AUDIO_SAMPLE_RATE_EXACT; //default. from AudioStream.h??
// pointer to current coefficients or NULL or FIR_PASSTHRU
const float32_t *coeff_p;
// ARM DSP Math library filter instance
arm_biquad_casd_df1_inst_f32 iir_inst;
float32_t StateF32[4*IIR_MAX_STAGES];
};
#endif