/* * 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