/**
  ******************************************************************************
  * @file    AudioFilterConvolution_F32.cpp
  * @author  Giuseppe Callipo - IK8YFW - ik8yfw@libero.it
  * @version V2.0.0
  * @date    06-02-2021
  * @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 reinititializzation;
    5) Optimize it to use as output audio filter on SDR receiver.
    6) Optimize the time execution
  *******************************************************************/
  // Revised for OpenAudio_Arduino Teensy F32 library,  8 Feb 2022

#include "AudioFilterConvolution_F32.h"

void AudioFilterConvolution_F32::passThrough(int stat)
{
  passThru=stat;
}

// Function to pre-calculate the multiplying frequency function, the "mask."
void AudioFilterConvolution_F32::impulse(float32_t *FIR_coef)
{
    uint32_t k = 0;
    uint32_t i = 0;
    enabled = 0; // shut off audio stream while impulse is loading
    for (i = 0; i < (FFT_length / 2) + 1; i++)
    {
        FIR_filter_mask[k++] = FIR_coef[i];
        FIR_filter_mask[k++] = 0;
    }

    for (i = FFT_length + 1; i < FFT_length * 2; i++)
    {
        FIR_filter_mask[i] = 0.0;
    }
    arm_cfft_f32( &arm_cfft_sR_f32_len1024, FIR_filter_mask, 0, 1);

    // for 1st time thru, zero out the last sample buffer to 0
  arm_fill_f32(0, last_sample_buffer_L, BUFFER_SIZE *4);

    state = 0;
    enabled = 1;  //enable audio stream again
}

void AudioFilterConvolution_F32::update(void)
{
    audio_block_f32_t *block;
    float32_t *bp;

    if (enabled != 1 ) return;
    block = receiveWritable_f32(0);   // MUST be Writable, as convolution results are written into block
    if (block) {
        switch (state) {
        case 0:
            if (passThru ==0) {
                arm_cmplx_mult_cmplx_f32(FFT_buffer, FIR_filter_mask, iFFT_buffer, FFT_length);   // complex multiplication in Freq domain = convolution in time domain
                arm_cfft_f32(&arm_cfft_sR_f32_len1024, iFFT_buffer, 1, 1);  // perform complex inverse FFT
                k = 0;
                l = 1024;
                for (int i = 0; i < 512; i++) {
                  buffer[i] = last_sample_buffer_L[i] + iFFT_buffer[k++];   // this performs the "ADD" in overlap/Add
                  last_sample_buffer_L[i] = iFFT_buffer[l++];       // this saves 512 samples (overlap) for next time around
                  k++;
                  l++;
                }
            }
            arm_copy_f32 (&buffer[0], &tbuffer[0], BUFFER_SIZE*4);
            bp = block->data;
            for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
                buffer[i] = *bp;
                *bp++ = tbuffer[i];
            }
            AudioStream_F32::transmit(block);
            AudioStream_F32::release(block);
            state = 1;
            break;

        case 1:
            bp = block->data;
            for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
                buffer[128+i] = *bp;
                *bp++ = tbuffer[i+128];
            }
            AudioStream_F32::transmit(block);
            AudioStream_F32::release(block);
            state = 2;
            break;

        case 2:
            bp = block->data;
            for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
                buffer[256 + i] = *bp;
                *bp++ = tbuffer[i+256];  // tbuffer contains results of last FFT/multiply/iFFT processing (convolution filtering)
            }

            AudioStream_F32::transmit(block);
            AudioStream_F32::release(block);
             // zero pad last half of array- necessary to prevent aliasing in FFT
             arm_fill_f32(0, FFT_buffer + 1024, FFT_length);
            state = 3;
            break;

        case 3:
            bp = block->data;
            for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
                buffer[384 + i] = *bp;
                *bp++ = tbuffer[i + 384];  // tbuffer contains results of last FFT/multiply/iFFT processing (convolution filtering)
            }
            AudioStream_F32::transmit(block);
            AudioStream_F32::release(block);
            state = 0;
            // 4 blocks are in- now do the FFT1024,complex multiply and iFFT1024  on 512samples of data
            // using the overlap/add method
            if (passThru ==0) {
                //fill FFT_buffer with current audio samples
                k = 0;
                for (i = 0; i < 512; i++)
                {
                    FFT_buffer[k++] = buffer[i];   // real
                    FFT_buffer[k++] = buffer[i];   // imag
                }
                // calculations are performed in-place in FFT routines
                arm_cfft_f32(&arm_cfft_sR_f32_len1024, FFT_buffer, 0, 1);// perform complex FFT
             } //end if passTHru
            break;
        }
    }
}

float32_t AudioFilterConvolution_F32::Izero (float32_t x)
{
    float32_t x2 = x / 2.0;
    float32_t summe = 1.0;
    float32_t ds = 1.0;
    float32_t di = 1.0;
    float32_t errorlimit = 1e-9;
    float32_t tmp;
    do
    {
        tmp = x2 / di;
        tmp *= tmp;
        ds *= tmp;
        summe += ds;
        di += 1.0;
    }   while (ds >= errorlimit * summe);
    return (summe);
}  // END Izero

float AudioFilterConvolution_F32::m_sinc(int m, float fc)
{ // fc is f_cut/(Fsamp/2)
  // m is between -M and M step 2
  //
  float x = m*PIH;
  if(m == 0)
    return 1.0f;
  else
    return sinf(x*fc)/(fc*x);
}

void AudioFilterConvolution_F32::calc_FIR_coeffs
      (float *coeffs, int numCoeffs, float32_t fc, float32_t Astop,
       int type, float dfc, float Fsamprate){
    // pointer to coefficients variable, no. of coefficients to calculate,
    // frequency where it happens, stopband attenuation in dB,
    // filter type, half-filter bandwidth (only for bandpass and notch).
    // Modified by WMXZ and DD4WH after
    // Wheatley, M. (2011): CuteSDR Technical Manual. www.metronix.com,
    // pages 118 - 120, FIR with Kaiser-Bessel Window.
    // Assess required number of coefficients by
    //     numCoeffs = (Astop - 8.0) / (2.285 * TPI * normFtrans);
    // selecting high-pass, numCoeffs is forced to an even number for
    // better frequency response
     int ii,jj;
     float32_t Beta;
     float32_t izb;
     float fcf = fc;
     int nc = numCoeffs;
     fc = 2.0f * fc / Fsamprate;    // Corrected
     dfc = dfc / Fsamprate;
     // calculate Kaiser-Bessel window shape factor beta from stop-band attenuation
     if (Astop < 20.96)
       Beta = 0.0;
     else if (Astop >= 50.0)
       Beta = 0.1102 * (Astop - 8.71);
     else
       Beta = 0.5842 * powf((Astop - 20.96), 0.4) + 0.07886 * (Astop - 20.96);

     izb = Izero (Beta);
     if(type == LOWPASS)
     {  fcf = fc;
      nc =  numCoeffs;
     }
     else if(type == HIGHPASS)
     {  fcf = -fc;
      nc =  2*(numCoeffs/2);
     }
     else if ((type == BANDPASS) || (type==BANDREJECT))
     {
       fcf = dfc;
       nc =  2*(numCoeffs/2); // maybe not needed
     }

/*
     else if (type==HILBERT)
     {
       nc =  2*(numCoeffs/2);
       // clear coefficients
       for(ii=0; ii< 2*(nc-1); ii++) coeffs[ii]=0;
       // set real delay
       coeffs[nc]=1;                  // <<<<<??????????
       // set imaginary Hilbert coefficients
       for(ii=1; ii< (nc+1); ii+=2)
       {
         if(2*ii==nc) continue;
         float x =(float)(2*ii - nc)/(float)nc;
         float w = Izero(Beta*sqrtf(1.0f - x*x))/izb; // Kaiser window
         coeffs[ii-1] = 1.0f/(PIH*(float)(ii-nc/2)) * w ;
         //coeffs[2*ii+1] = 1.0f/(PIH*(float)(ii-nc/2)) * w ;
       }
       return;  // From Hilbert design
     }
 */

     for(ii= - nc, jj=0; ii< nc; ii+=2,jj++)
     {
       float x =(float)ii/(float)nc;
       float w = Izero(Beta*sqrtf(1.0f - x*x))/izb; // Kaiser window
       coeffs[jj] = fcf * m_sinc(ii,fcf) * w;
     }

     if(type==HIGHPASS)
     {
       coeffs[nc/2] += 1;
     }
     else if (type==BANDPASS)
     {
       for(jj=0; jj< nc+1; jj++) coeffs[jj] *= 2.0f*cosf(PIH*(2*jj-nc)*fc);
     }
     else if (type==BANDREJECT)
     {
       for(jj=0; jj< nc+1; jj++) coeffs[jj] *= -2.0f*cosf(PIH*(2*jj-nc)*fc);
       coeffs[nc/2] += 1;
     }
} // END calc_FIR_coef

void AudioFilterConvolution_F32::initFilter ( float32_t fc, float32_t Astop, int type, float dfc){
    //Init Fir
    calc_FIR_coeffs (FIR_Coef, MAX_NUMCOEF, fc, Astop, type, dfc, fs);
    enabled = 0;   // set to zero to disable audio processing until impulse has been loaded
    impulse(FIR_Coef);  // generates Filter Mask and enables the audio stream
}