/*
* AudioFilterBiquad_F32.h
* Chip Audette, OpenAudio, Apr 2017
* MIT License, Use at your own risk.
*
* This filter has been running in F32 as a single stage. This
* would work by using multiple instantations, but compute time and
* latency suffer. So, Feb 2021 convert to MAX_STAGES 4 as is the I16
* Teensy Audio library. Bob Larrkin
*
* Float vs Double. There are times when double precision in the
* BiQuad calculation is needed to prevent
* serious numerical errors. This can be a processor time problem for
* T3.x. This routine (NOT QUITE YET) allows for either by
* a function with float as the default. This allows different BiQuads
* to use float or double. RSL
*
* NOTE: If your INO is broken, we had to do it.
* Some setting of coefficients also did a
* begin of the ARM CMSIS. We can't do that with multiple stages. If you
* encouter this, add myBiquad.begin(); to your INO after the
* coefficients have been set. Feb 2021
*
* The sign of the coefficients for feedback, the a[], here use the
* convention of the ARM CMSIS library. Matlab reverses the signs of these.
* I believe these are treated per those rules!! Bob
*
* Algorithm for CMSIS library
* Each Biquad stage implements a second order filter using the difference equation:
* y[n] = b0 * x[n] + b1 * x[n-1] + b2 * x[n-2] + a1 * y[n-1] + a2 * y[n-2]
* The a1 and a2 coeccicients do not have minus signs as do the Matlab ones.
*/
#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) ,,,,,REMOVE???? WE HAVE DO_BIQUAD THAT DOES THISS
#define IIR_F32_PASSTHRU ((const float32_t *) 1)
// Changed Feb 2021
#define IIR_MAX_STAGES 4
// T4.x can generally use doubles, they may be a burden for T3.x
// Leave commented out to compile for BOTH float and double
// See the function useDouble(bool d) below
// #define NEVER_DOUBLE
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) {
setSampleRate_Hz(AUDIO_SAMPLE_RATE_EXACT);
sampleRate_Hz = AUDIO_SAMPLE_RATE_EXACT; // <<<<<<<<<<<<<<<<<<<<<< CHECK IF NEEDED??
doClassInit();
}
AudioFilterBiquad_F32(const AudioSettings_F32 &settings):
AudioStream_F32(1,inputQueueArray) {
setSampleRate_Hz(settings.sample_rate_Hz);
doClassInit();
}
void doClassInit(void) {
for(int ii=0; ii<5*IIR_MAX_STAGES; ii++) {
coeff32[ii] = 0.0;
coeff64[ii] = 0.0;
}
for(int ii=0; ii<4; ii++) {
coeff32[5*ii] = 1.0; // b0 = 1 for pass through
coeff64[5*ii] = 1.0;
}
numStagesUsed = 0; // Can be 0 to 4
doBiquad = false; // This is the way to jump over the biquad
}
// Up to 4 stages are allowed. Coefficients, either by design function
// or from direct setCoefficients() need to be added to the double array
// and also to the float
void setCoefficients(int iStage, double *cf) {
if (iStage > IIR_MAX_STAGES) {
if (Serial) {
Serial.print("AudioFilterBiquad_F32: setCoefficients:");
Serial.println(" *** MaxStages Error");
}
return;
}
if((iStage + 1) > numStagesUsed)
numStagesUsed = iStage + 1; // There may be blank pass throughs
for(int ii=0; ii<5; ii++) {
coeff64[ii + 5*iStage] = cf[ii]; // The local collection of double coefficients
coeff32[ii + 5*iStage] = (float)cf[ii]; // and of floats
}
doBiquad = true;
}
// ARM DSP Math library filter instance.
// Does the initialization of ARM CMSIS DSP BiQuad structure. This MUST follow the
// setting of coefficients to catch the max number of stages and do the
// double to float conversion for the CMSIS routine.
void begin(void) {
// Initialize BiQuad instance (ARM DSP Math Library)
//https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html
arm_biquad_cascade_df1_init_f32(&iir_inst, numStagesUsed, &coeff32[0], &StateF32[0]);
}
void end(void) {
doBiquad = false;
}
void setSampleRate_Hz(float _fs_Hz) { sampleRate_Hz = _fs_Hz; }
// Deprecated
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
double b[] = {8.173653471988667e-01, -1.634730694397733e+00, 8.173653471988667e-01}; //from Matlab
double a[] = { 1.000000000000000e+00, -1.601092394183619e+00, 6.683689946118476e-01}; //from Matlab
setFilterCoeff_Matlab(b, a);
}
void setFilterCoeff_Matlab(double b[], double a[]) { //one stage of N=2 IIR
double coeff[5];
//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
setCoefficients(0, coeff);
}
//Two update() options, floats or doubles
void useDouble(bool ud) {
useDoubleCoefs = ud; // true is to use doubles
useDoubleCoefs = false; // Not implemented yet
}
// 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(int stage, float frequency, float q) {
double coeff[5];
double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
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 setHighpass(uint32_t stage, float frequency, float q) {
double coeff[5];
double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
double sinW0 = sin(w0);
double alpha = sinW0 / ((double)q * 2.0);
double cosW0 = cos(w0);
double scale = 1.0 / (1.0+alpha);
/* 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) {
double coeff[5];
double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
double sinW0 = sin(w0);
double alpha = sinW0 / ((double)q * 2.0);
double cosW0 = cos(w0);
double scale = 1.0 / (1.0+alpha);
/* 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);
}
// frequency in Hz. q makes the response stay close to 0.0dB until
// close to the notch frequency. q up to 100 or more seem stable.
void setNotch(uint32_t stage, float frequency, float q) {
double coeff[5];
double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
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) {
double coeff[5];
double a = pow(10.0, gain/40.0);
double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
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) {
double coeff[5];
double a = pow(10.0, gain/40.0);
double w0 = frequency * (2 * 3.141592654 / sampleRate_Hz);
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);
}
double* getCoeffs(void) {
return coeff64; // Pointer to 20 coefficients in double.
}
void update(void);
private:
audio_block_f32_t *inputQueueArray[1];
float coeff32[5 * IIR_MAX_STAGES]; // Local copies to be transferred with begin()
double coeff64[5 * IIR_MAX_STAGES];
float StateF32[4*IIR_MAX_STAGES];
//double StateF64[4*IIR_MAX_STAGES]; // Will need this for 64 bit version
float sampleRate_Hz = AUDIO_SAMPLE_RATE_EXACT; //default. from AudioStream.h??
int numStagesUsed = 0;
bool useDoubleCoefs = false; // As of now, all float <<<<<<<<<<<<<<<<<<<<
bool doBiquad = false;
/* Info - The structure from arm_biquad_casd_df1_inst_f32 consists of
* uint32_t numStages;
* const float32_t *pCoeffs; //Points to the array of coefficients, length 5*numStages.
* float32_t *pState; //Points to the array of state variables, length 4*numStages.
*/
// ARM DSP Math library filter instance.
arm_biquad_casd_df1_inst_f32 iir_inst;
};
#endif