/* * 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; 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