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276 lines
9.3 KiB
276 lines
9.3 KiB
/**
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******************************************************************************
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* @file AudioFilterConvolution_F32.cpp
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* @author Giuseppe Callipo - IK8YFW - ik8yfw@libero.it
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* @version V1.0.0
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* @date 02-05-2021
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* @brief F32 Filter Convolution
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*
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******************************************************************************
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******************************************************************************
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This software is based on the AudioFilterConvolution routine
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Written by Brian Millier on Mar 2017
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https://circuitcellar.com/research-design-hub/fancy-filtering-with-the-teensy-3-6/
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and modified by Giuseppe Callipo - ik8yfw.
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Modifications:
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1) Class refactoring, change some methods visibility;
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2) Filter coefficients calculation included into class;
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3) Change the class for running in both with F32
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OpenAudio_ArduinoLibrary for Teensy;
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4) Added initFilter method for single anf fast initialization and on
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the fly reinititializzation;
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5) Optimize it to use as output audio filter on SDR receiver.
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*******************************************************************/
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#include "AudioFilterConvolution_F32.h"
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boolean AudioFilterConvolution_F32::begin(int status)
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{
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enabled = status;
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return(true);
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}
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void AudioFilterConvolution_F32::passThrough(int stat)
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{
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passThru=stat;
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}
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void AudioFilterConvolution_F32::impulse(float32_t *FIR_coef) {
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// arm_q15_to_float(coefs, FIR_coef, 513); // convert int_buffer to float 32bit
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int k = 0;
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int i = 0;
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enabled = 0; // shut off audio stream while impulse is loading
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for (i = 0; i < (FFT_length / 2) + 1; i++)
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{
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FIR_filter_mask[k++] = FIR_coef[i];
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FIR_filter_mask[k++] = 0;
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}
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for (i = FFT_length + 1; i < FFT_length * 2; i++)
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{
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FIR_filter_mask[i] = 0.0;
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}
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arm_cfft_f32( &arm_cfft_sR_f32_len1024, FIR_filter_mask, 0, 1);
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for (int i = 0; i < 1024; i++) {
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// Serial.println(FIR_filter_mask[i] * 32768);
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}
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// for 1st time thru, zero out the last sample buffer to 0
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memset(last_sample_buffer_L, 0, sizeof(last_sample_buffer_L));
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state = 0;
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enabled = 1; //enable audio stream again
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}
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void AudioFilterConvolution_F32::update(void)
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{
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audio_block_f32_t *block;
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float32_t *bp;
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if (enabled != 1 ) return;
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block = receiveWritable_f32(0); // MUST be Writable, as convolution results are written into block
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if (block) {
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switch (state) {
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case 0:
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bp = block->data;
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
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buffer[i] = *bp++;
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}
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bp = block->data;
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
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*bp++ = tbuffer[i]; // tbuffer contains results of last FFT/multiply/iFFT processing (convolution filtering)
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}
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AudioStream_F32::transmit(block);
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AudioStream_F32::release(block);
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state = 1;
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break;
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case 1:
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bp = block->data;
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
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buffer[128+i] = *bp++;
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}
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bp = block->data;
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
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*bp++ = tbuffer[i+128]; // tbuffer contains results of last FFT/multiply/iFFT processing (convolution filtering)
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}
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AudioStream_F32::transmit(block);
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AudioStream_F32::release(block);
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state = 2;
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break;
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case 2:
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bp = block->data;
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
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buffer[256 + i] = *bp++;
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}
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bp = block->data;
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
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*bp++ = tbuffer[i+256]; // tbuffer contains results of last FFT/multiply/iFFT processing (convolution filtering)
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}
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AudioStream_F32::transmit(block);
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AudioStream_F32::release(block);
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state = 3;
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break;
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case 3:
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bp = block->data;
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
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buffer[384 + i] = *bp++;
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}
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bp = block->data;
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for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
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*bp++ = tbuffer[i + 384]; // tbuffer contains results of last FFT/multiply/iFFT processing (convolution filtering)
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}
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AudioStream_F32::transmit(block);
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AudioStream_F32::release(block);
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state = 0;
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// 4 blocks are in- now do the FFT1024,complex multiply and iFFT1024 on 512samples of data
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// using the overlap/add method
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// 1st convert Q15 samples to float
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//arm_q15_to_float(buffer, float_buffer_L, 512);
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arm_copy_f32 (buffer, float_buffer_L, 512);
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// float_buffer samples are now standardized from > -1.0 to < 1.0
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if (passThru ==0) {
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memset(FFT_buffer + 1024, 0, sizeof(FFT_buffer) / 2); // zero pad last half of array- necessary to prevent aliasing in FFT
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//fill FFT_buffer with current audio samples
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k = 0;
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for (i = 0; i < 512; i++)
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{
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FFT_buffer[k++] = float_buffer_L[i]; // real
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FFT_buffer[k++] = float_buffer_L[i]; // imag
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}
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// calculations are performed in-place in FFT routines
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arm_cfft_f32(&arm_cfft_sR_f32_len1024, FFT_buffer, 0, 1);// perform complex FFT
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arm_cmplx_mult_cmplx_f32(FFT_buffer, FIR_filter_mask, iFFT_buffer, FFT_length); // complex multiplication in Freq domain = convolution in time domain
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arm_cfft_f32(&arm_cfft_sR_f32_len1024, iFFT_buffer, 1, 1); // perform complex inverse FFT
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k = 0;
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l = 1024;
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for (int i = 0; i < 512; i++) {
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float_buffer_L[i] = last_sample_buffer_L[i] + iFFT_buffer[k++]; // this performs the "ADD" in overlap/Add
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last_sample_buffer_L[i] = iFFT_buffer[l++]; // this saves 512 samples (overlap) for next time around
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k++;
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l++;
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}
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} //end if passTHru
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// convert floats to Q15 and save in temporary array tbuffer
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//arm_float_to_q15(&float_buffer_L[0], &tbuffer[0], BUFFER_SIZE*4);
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arm_copy_f32 (&float_buffer_L[0], &tbuffer[0], BUFFER_SIZE*4);
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break;
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}
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}
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}
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float32_t AudioFilterConvolution_F32::Izero (float32_t x)
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{
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float32_t x2 = x / 2.0;
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float32_t summe = 1.0;
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float32_t ds = 1.0;
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float32_t di = 1.0;
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float32_t errorlimit = 1e-9;
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float32_t tmp;
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do
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{
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tmp = x2 / di;
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tmp *= tmp;
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ds *= tmp;
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summe += ds;
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di += 1.0;
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} while (ds >= errorlimit * summe);
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return (summe);
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} // END Izero
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float AudioFilterConvolution_F32::m_sinc(int m, float fc)
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{ // fc is f_cut/(Fsamp/2)
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// m is between -M and M step 2
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//
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float x = m*PIH;
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if(m == 0)
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return 1.0f;
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else
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return sinf(x*fc)/(fc*x);
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}
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void AudioFilterConvolution_F32::calc_FIR_coeffs (float * coeffs, int numCoeffs, float32_t fc, float32_t Astop, int type, float dfc, float Fsamprate)
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// pointer to coefficients variable, no. of coefficients to calculate, frequency where it happens, stopband attenuation in dB,
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// filter type, half-filter bandwidth (only for bandpass and notch)
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{ // modified by WMXZ and DD4WH after
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// Wheatley, M. (2011): CuteSDR Technical Manual. www.metronix.com, pages 118 - 120, FIR with Kaiser-Bessel Window
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// assess required number of coefficients by
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// numCoeffs = (Astop - 8.0) / (2.285 * TPI * normFtrans);
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// selecting high-pass, numCoeffs is forced to an even number for better frequency response
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int ii,jj;
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float32_t Beta;
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float32_t izb;
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float fcf = fc;
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int nc = numCoeffs;
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fc = fc / Fsamprate;
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dfc = dfc / Fsamprate;
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// calculate Kaiser-Bessel window shape factor beta from stop-band attenuation
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if (Astop < 20.96)
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Beta = 0.0;
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else if (Astop >= 50.0)
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Beta = 0.1102 * (Astop - 8.71);
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else
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Beta = 0.5842 * powf((Astop - 20.96), 0.4) + 0.07886 * (Astop - 20.96);
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izb = Izero (Beta);
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if(type == 0) // low pass filter
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// { fcf = fc;
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{ fcf = fc * 2.0;
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nc = numCoeffs;
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}
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else if(type == 1) // high-pass filter
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{ fcf = -fc;
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nc = 2*(numCoeffs/2);
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}
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else if ((type == 2) || (type==3)) // band-pass filter
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{
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fcf = dfc;
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nc = 2*(numCoeffs/2); // maybe not needed
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}
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else if (type==4) // Hilbert transform
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{
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nc = 2*(numCoeffs/2);
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// clear coefficients
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for(ii=0; ii< 2*(nc-1); ii++) coeffs[ii]=0;
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// set real delay
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coeffs[nc]=1;
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// set imaginary Hilbert coefficients
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for(ii=1; ii< (nc+1); ii+=2)
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{
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if(2*ii==nc) continue;
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float x =(float)(2*ii - nc)/(float)nc;
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float w = Izero(Beta*sqrtf(1.0f - x*x))/izb; // Kaiser window
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coeffs[2*ii+1] = 1.0f/(PIH*(float)(ii-nc/2)) * w ;
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}
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return;
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}
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for(ii= - nc, jj=0; ii< nc; ii+=2,jj++)
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{
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float x =(float)ii/(float)nc;
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float w = Izero(Beta*sqrtf(1.0f - x*x))/izb; // Kaiser window
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coeffs[jj] = fcf * m_sinc(ii,fcf) * w;
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}
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if(type==1)
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{
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coeffs[nc/2] += 1;
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}
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else if (type==2)
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{
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for(jj=0; jj< nc+1; jj++) coeffs[jj] *= 2.0f*cosf(PIH*(2*jj-nc)*fc);
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}
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else if (type==3)
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{
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for(jj=0; jj< nc+1; jj++) coeffs[jj] *= -2.0f*cosf(PIH*(2*jj-nc)*fc);
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coeffs[nc/2] += 1;
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}
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} // END calc_FIR_coef
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void AudioFilterConvolution_F32::initFilter ( float32_t fc, float32_t Astop, int type, float dfc){
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//Init Fir
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calc_FIR_coeffs (FIR_Coef, MAX_NUMCOEF, fc, Astop, type, dfc, fs);
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begin(0); // set to zero to disable audio processing until impulse has been loaded
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impulse(FIR_Coef); // generates Filter Mask and enables the audio stream
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}
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