OpenAudio_ArduinoLibrary/AudioFilterConvolution_F32.h

/**
  ******************************************************************************
  * @file    AudioFilterConvolution_F32.h
  * @author  Giuseppe Callipo - IK8YFW - ik8yfw@libero.it
  * @version V1.0.0
  * @date    02-05-2021
  * @brief   F32 Filter Convolution
  *
  ******************************************************************************
  ******************************************************************************
   This software is based on the  AudioFilterConvolution routine
   Written by Brian Millier on Mar 2017

   https://circuitcellar.com/research-design-hub/fancy-filtering-with-the-teensy-3-6/
   and modified by Giuseppe Callipo - ik8yfw.
   Modifications:
    1) Class refactoring, change some methods visibility;
    2) Filter coefficients calculation included into class;
    3) Change the class for running in both with F32
     OpenAudio_ArduinoLibrary for Teensy;
  4) Added initFilter method for single anf fast initialization and on
     the fly re-inititialization;
  5) Optimize it to use as output audio filter on SDR receiver.

  Brought to the Teensy F32 OpenAudio_ArduinoLibrary 2 Feb 2022, Bob Larkin
  ******************************************************************************/
/*Notes from Giuseppe Callipo, IK8YFW:

   *** How to use it.  ***

   Create an audio project based on chipaudette/OpenAudio_ArduinoLibrary
   like the Keiths SDR  Project, and just add the h and cpp file to your
   processing chain as unique output filter:

   ************************************************************

   1) Include the header
   #include "AudioFilterConvolution_F32.h" (or OpenAudio_ArduinoLibrary.h)

   2) Initialize as F32 block
   (the block must be 128 but the sample rate can be changed but must initialized)
     const float sample_rate_Hz = 96000.0f; // or 44100.0f or other
     const int   audio_block_samples = 128;
   AudioSettings_F32 audio_settings(sample_rate_Hz, audio_block_samples);

   AudioFilterConvolution_F32 FilterConv(audio_settings);

   3) Connect before output

   AudioConnection_F32 patchCord1(FilterConv,0,Output_i2s,0);
   AudioConnection_F32 patchCord2(FilterConv,0,Output_i2s,1);

   4) Set the filter value when you need - some examples:
   // CW - Centered at 800Hz, ( 40 db x oct ), 2=BPF, width = 1200Hz
   FilterConv.initFilter((float32_t)800, 40, 2, 1200.0);

   // SSB - Centered at 1500Hz, ( 60 db x oct ), 2=BPF, width = 3000Hz
   FilterConv.initFilter((float32_t)1500, 60, 2, 3000.0);
  ************************************************************** */
  /* Additional Notes from Bob
   * Measured 128 word update in update() is 248 microseconds (T4.x)
   * Comparison with a conventional FIR, from this library, the
   * AudioFilterFIRGeneral_F32 showed that a 512 tap FIR gave
   * essentially the same response and was slightly faster at
   * 225 microseconds per update.  Also, note that this form of
   * computation uses about 52 kB of memory where the direct FIR
   * uses about 10 kB.  The responses differ in only minor ways.
   * ************************************************************ */

#ifndef AudioFilterConvolution_F32_h_
#define AudioFilterConvolution_F32_h_

#include <AudioStream_F32.h>
#include <arm_math.h>
#include <arm_const_structs.h>

#define MAX_NUMCOEF 513
#define TPI           6.28318530717959f
#define PIH           1.57079632679490f
#define FOURPI        2.0 * TPI
#define SIXPI         3.0 * TPI

class AudioFilterConvolution_F32 :
 public AudioStream_F32
{
public:
  AudioFilterConvolution_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {
          fs = AUDIO_SAMPLE_RATE;
          //block_size = AUDIO_BLOCK_SAMPLES;
	  };
  AudioFilterConvolution_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) {
        // Performs the first initialize
        fs = settings.sample_rate_Hz;
      };

  boolean begin(int status);
  virtual void update(void);
  void passThrough(int stat);
  void initFilter (float32_t fc, float32_t Astop, int type, float dfc);

private:
  #define BUFFER_SIZE 128
  float32_t fs;

  audio_block_f32_t *inputQueueArray_f32[1];

  float32_t *sp_L;
  volatile uint8_t state;
  int i;
  int k;
  int l;
  int passThru;
  int enabled;
  float32_t FIR_Coef[MAX_NUMCOEF];
  const uint32_t FFT_length = 1024;
  float32_t FIR_coef[2048] __attribute__((aligned(4)));
  float32_t FIR_filter_mask[2048] __attribute__((aligned(4)));
  float32_t buffer[2048] __attribute__((aligned(4)));
  float32_t tbuffer[2048]__attribute__((aligned(4)));
  float32_t FFT_buffer[2048] __attribute__((aligned(4)));
  float32_t iFFT_buffer[2048] __attribute__((aligned(4)));
  float32_t float_buffer_L[512]__attribute__((aligned(4)));
  float32_t last_sample_buffer_L[512];
  void impulse(float32_t *coefs);
  float32_t Izero (float32_t x);
  float     m_sinc(int m, float fc);
  void      calc_FIR_coeffs (float * coeffs, int numCoeffs, float32_t fc, float32_t Astop, int type, float dfc, float Fsamprate);

};

#endif