/* Audio Library for Teensy 3.X
   Copyright (c) 2014, Pete (El Supremo)
   Copyright (c) 2019, Holger Wirtz

   Permission is hereby granted, free of charge, to any person obtaining a copy
   of this software and associated documentation files (the "Software"), to deal
   in the Software without restriction, including without limitation the rights
   to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
   copies of the Software, and to permit persons to whom the Software is
   furnished to do so, subject to the following conditions:

   The above copyright notice and this permission notice shall be included in
   all copies or substantial portions of the Software.

   THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
   IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
   FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
   AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
   LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
   OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
   THE SOFTWARE.
*/

#include <Arduino.h>
#include <Audio.h>
#include "limits.h"
#include "effect_modulated_delay.h"
#include "spline.h"
#include "config.h"

/******************************************************************/

// Based on;      A u d i o E f f e c t D e l a y
// Written by Pete (El Supremo) Jan 2014
// 140529 - change to handle mono stream - change modify() to voices()
// 140219 - correct storage class (not static)
// 190527 - added modulation input (by Holger Wirtz)

boolean AudioEffectModulatedDelay::begin(short *delayline, int d_length)
{
#if 0
  Serial.print(F("AudioEffectModulatedDelay.begin(Chorus delay line length = "));
  Serial.print(d_length);
  Serial.println(F(")"));
#endif

  _delayline = NULL;
  _delay_length = 0;
  _circ_idx = 0;

  if (delayline == NULL) {
    return (false);
  }
  if (d_length < 10) {
    return (false);
  }

  _delayline = delayline;
  _delay_length = _max_delay_length = d_length;

  // init filter
  filter.numStages = 1;
  filter.pState =  filter_state;
  filter.pCoeffs = filter_coeffs;
  calcModFilterCoeff(5000.0, 0.0, 5.0);

  return (true);
}

void AudioEffectModulatedDelay::update(void)
{
  audio_block_t *block;
  audio_block_t *modulation;

  if (_delayline == NULL)
    return;

  block = receiveWritable(0);
  modulation = receiveReadOnly(1);

  if (block && modulation)
  {
    int16_t *bp;
    float *mp;
    float mod_idx;
    float mod_number;
    float mod_fraction;

#ifdef INTERPOLATE_MODE
    int8_t j;
    float x[INTERPOLATION_WINDOW_SIZE];
    float y[INTERPOLATION_WINDOW_SIZE];
    Spline s(x, y, INTERPOLATION_WINDOW_SIZE, INTERPOLATE_MODE);
#endif

    // (Filter implementation: https://web.fhnw.ch/technik/projekte/eit/Fruehling2016/MuelZum/html/parametric__equalizer__example_8c_source.html)
    arm_q15_to_float(modulation->data, modulation_f32, AUDIO_BLOCK_SAMPLES);
    arm_biquad_cascade_df1_f32(&filter, modulation_f32, modulation_f32, AUDIO_BLOCK_SAMPLES);
    bp = block->data;
    mp = modulation_f32;

    for (uint16_t i = 0; i < AUDIO_BLOCK_SAMPLES; i++)
    {
      // write data into circular buffer
      if (_circ_idx >= _delay_length)
        _circ_idx = 0;
      _delayline[_circ_idx] = *bp;

      // Calculate modulation index as a float, for interpolation later.
      // The index is located around the half of the delay length multiplied by the current amount of the modulator
      mod_idx = *mp * float(_delay_length >> 1);
      mod_fraction = modff(mod_idx, &mod_number);

#ifdef INTERPOLATE_MODE
      // Generate a an array with the size of INTERPOLATION_WINDOW_SIZE of x/y values around mod_idx for interpolation
      uint8_t c;
      int16_t c_mod_idx = _circ_idx - int(round(mod_idx)); // This is the pointer to the value in the circular buffer at the current modulation index
      for (j = ~(INTERPOLATION_WINDOW_SIZE >> 1) | 0x01, c = 0; j <= INTERPOLATION_WINDOW_SIZE >> 1; j++, c++) // only another way to say: from -INTERPOLATION_WINDOW_SIZE/2 to INTERPOLATION_WINDOW_SIZE/2
      {
        int16_t jc_mod_idx = (c_mod_idx + j) % _delay_length; // The modulation index pointer plus the value of the current window pointer
        if (jc_mod_idx < 0)
          y[c] = float(_delayline[_delay_length + jc_mod_idx]);
        else
          y[c] = float(_delayline[jc_mod_idx]);
        x[c] = float(j);
      }
      *bp = int(round(s.value(mod_fraction)));
#else
      // Simple interpolation
      int16_t c_mod_idx = (_circ_idx - int(round(mod_idx))) % _delay_length;
      float value1, value2;

      if (c_mod_idx < 0)
      {
        value1 = _delayline[_delay_length + c_mod_idx - 1];
        value2 = _delayline[_delay_length + c_mod_idx];
      }
      else
      {
        value1 = _delayline[c_mod_idx - 1];
        value2 = _delayline[c_mod_idx];
      }
      *bp = mod_fraction * value1 + (1.0 - mod_fraction) * value2;
#endif

      bp++; // next audio data
      mp++; // next modulation data
      _circ_idx++; // next circular buffer index
    }
  }

  if (modulation)
    release(modulation);

  if (block)
  {
    transmit(block, 0);
    release(block);
  }
}

void AudioEffectModulatedDelay::setDelay(float milliseconds)
{
  _delay_length = min(AUDIO_SAMPLE_RATE * milliseconds / 500, _max_delay_length);
}

void AudioEffectModulatedDelay::calcModFilterCoeff(float32_t cFrq, float32_t gain, float32_t width)
{
  /* Calculate intermediate values */
  //  float32_t A = sqrt(pow(10, gain / 20.0f));
  //  float32_t w0 = 2.0f * PI * cFrq / ((float32_t)AUDIO_SAMPLE_RATE_EXACT);
  //  float32_t cosw0 = cos(w0);
  //  float32_t sinw0 = sin(w0);
  //  float32_t alpha = sinw0 / (2.0f * width);

  /* Calculate coefficients */
  //  float32_t b0 =  1.0f + alpha * A;
  //  float32_t b1 = -2.0f * cosw0;
  //  float32_t b2 =  1.0f - alpha * A;
  //  float32_t a0 =  1.0f + alpha / A;
  //  float32_t a1 = -2.0f * cosw0;
  //  float32_t a2 =  1.0f - alpha / A;

/* https://stackoverflow.com/questions/20924868/calculate-coefficients-of-2nd-order-butterworth-low-pass-filter/20932062
    ff=cutoff_frq/sample_rate=AUDIO_SAMPLE_RATE_EXACT/1000
    const double ita =1.0/ tan(M_PI*ff);
    const double q=sqrt(2.0);
    b0 = 1.0 / (1.0 + q*ita + ita*ita);
    b1= 2*b0;
    b2= b0;
    a1 = 2.0 * (ita*ita - 1.0) * b0;
    a2 = -(1.0 - q*ita + ita*ita) * b0;

    ff=1000/44117.64706=0.02266666666
    ita=804.60898525
    q=1.414213
    
 */
  // 1kHz 2nd order Butterworth lowpass filter coefficients
  // calculated with Iowa IIR FIlter Designer 6.5
  float32_t b0 = 0.124589380980617656;
  float32_t b1 = 0.124589380980617656;
  float32_t b2 = 0.0;
  float32_t a0 = 1.000000000000000000;
  float32_t a1 = -0.750821238038764660;
  float32_t a2 = 0.0;

  /* Normalize so a0 = 1 */
  filter_coeffs[0] = b0 / a0;
  filter_coeffs[1] = b1 / a0;
  filter_coeffs[2] = b2 / a0;
  filter_coeffs[3] = -a1 / a0;
  filter_coeffs[4] = -a2 / a0;
}

void AudioEffectModulatedDelay::setModFilter(float cFrq, float gain, float width)
{
  calcModFilterCoeff(cFrq, gain, width);
}