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OpenAudio_ArduinoLibrary/AudioFilterIIR_F32.h

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3.4 KiB

/*
* AudioFilterFIR_F32
*
* Created: Chip Audette (OpenAudio) Feb 2017
* - Building from AudioFilterFIR from Teensy Audio Library (AudioFilterFIR credited to Pete (El Supremo))
*
*/
#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 AudioFilterIIR_F32 : public AudioStream_F32
{
public:
AudioFilterIIR_F32(void): AudioStream_F32(1,inputQueueArray), coeff_p(FIR_F32_PASSTHRU) {
}
void begin(const float32_t *cp, int n_stages) {
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 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
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
uint8_t n_stages = 1;
arm_biquad_cascade_df1_init_f32(&iir_inst, n_stages, hp_coeff, &StateF32[0]);
}
virtual void update(void);
private:
audio_block_f32_t *inputQueueArray[1];
float32_t hp_coeff[5 * 1] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later
// 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];
};
void AudioFilterIIR_F32::update(void)
{
audio_block_f32_t *block, *b_new;
block = AudioStream_F32::receiveWritable_f32();
if (!block) return;
// If there's no coefficient table, give up.
if (coeff_p == NULL) {
AudioStream_F32::release(block);
return;
}
// do passthru
if (coeff_p == IIR_F32_PASSTHRU) {
// Just passthrough
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
return;
}
// do IIR
arm_biquad_cascade_df1_f32(&iir_inst, block->data, block->data, block->length);
AudioStream_F32::transmit(block); // send the IIR output
AudioStream_F32::release(block);
}
#endif