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