/* Audio Library for Teensy 3.X
 * Copyright (c) 2014, Pete (El Supremo)
 *
 * 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 "effect_flange.h"
#include "arm_math.h"

/******************************************************************/
//                A u d i o E f f e c t F l a n g e
// Written by Pete (El Supremo) Jan 2014
// 140529 - change to handle mono stream and change modify() to voices()
// 140207 - fix calculation of delay_rate_incr which is expressed as
//			a fraction of 2*PI
// 140207 - cosmetic fix to begin()
// 140219 - correct the calculation of "frac"

// circular addressing indices for left and right channels
//short AudioEffectFlange::l_circ_idx;
//short AudioEffectFlange::r_circ_idx;

//short * AudioEffectFlange::l_delayline = NULL;
//short * AudioEffectFlange::r_delayline = NULL;

// User-supplied offset for the delayed sample
// but start with passthru
//int AudioEffectFlange::delay_offset_idx = FLANGE_DELAY_PASSTHRU;
//int AudioEffectFlange::delay_length;

//int AudioEffectFlange::delay_depth;
//int AudioEffectFlange::delay_rate_incr;
//unsigned int AudioEffectFlange::l_delay_rate_index;
//unsigned int AudioEffectFlange::r_delay_rate_index;
// fails if the user provides unreasonable values but will
// coerce them and go ahead anyway. e.g. if the delay offset
// is >= CHORUS_DELAY_LENGTH, the code will force it to
// CHORUS_DELAY_LENGTH-1 and return false.
// delay_rate is the rate (in Hz) of the sine wave modulation
// delay_depth is the maximum variation around delay_offset
// i.e. the total offset is delay_offset + delay_depth * sin(delay_rate)
boolean AudioEffectFlange::begin(short *delayline,int d_length,int delay_offset,int d_depth,float delay_rate)
{
  boolean all_ok = true;

if(0) {
  Serial.print("AudioEffectFlange.begin(offset = ");
  Serial.print(delay_offset);
  Serial.print(", depth = ");
  Serial.print(d_depth);
  Serial.print(", rate = ");
  Serial.print(delay_rate,3);
  Serial.println(")");
  Serial.print("    FLANGE_DELAY_LENGTH = ");
  Serial.println(d_length);
}
  delay_length = d_length/2;
  l_delayline = delayline;
  
  delay_depth = d_depth;
  // initial index
  l_delay_rate_index = 0;
  l_circ_idx = 0;
  delay_rate_incr =(delay_rate * 2147483648.0)/ AUDIO_SAMPLE_RATE_EXACT;
//Serial.println(delay_rate_incr,HEX);

  delay_offset_idx = delay_offset;
  // Allow the passthru code to go through
  if(delay_offset_idx < -1) {
    delay_offset_idx = 0;
    all_ok = false;
  }
  if(delay_offset_idx >= delay_length) {
    delay_offset_idx = delay_length - 1;
    all_ok = false;
  }  
  return(all_ok);
}


boolean AudioEffectFlange::voices(int delay_offset,int d_depth,float delay_rate)
{
  boolean all_ok = true;
  
  delay_depth = d_depth;

  delay_rate_incr =(delay_rate * 2147483648.0)/ AUDIO_SAMPLE_RATE_EXACT;
  
  delay_offset_idx = delay_offset;
  // Allow the passthru code to go through
  if(delay_offset_idx < -1) {
    delay_offset_idx = 0;
    all_ok = false;
  }
  if(delay_offset_idx >= delay_length) {
    delay_offset_idx = delay_length - 1;
    all_ok = false;
  }
  l_delay_rate_index = 0;
  l_circ_idx = 0;
  return(all_ok);
}

void AudioEffectFlange::update(void)
{
  audio_block_t *block;
  int idx;
  short *bp;
  short frac;
  int idx1;

  if(l_delayline == NULL)return;

  // do passthru
  if(delay_offset_idx == FLANGE_DELAY_PASSTHRU) {
    // Just passthrough
    block = receiveWritable(0);
    if(block) {
      bp = block->data;
      // fill the delay line
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        l_circ_idx++;
        if(l_circ_idx >= delay_length) {
          l_circ_idx = 0;
        }
        l_delayline[l_circ_idx] = *bp++;
      }
      // transmit the unmodified block
      transmit(block,0);
      release(block);
    }
    return;
  }

  //          L E F T  C H A N N E L

  block = receiveWritable(0);
  if(block) {
    bp = block->data;
    for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
      // increment the index into the circular delay line buffer
      l_circ_idx++;
      // wrap the index around if necessary
      if(l_circ_idx >= delay_length) {
        l_circ_idx = 0;
      }
      // store the current sample in the delay line
      l_delayline[l_circ_idx] = *bp;
      // The argument to the arm_sin_q15 function is NOT in radians. It is
      // actually, in effect, the fraction remaining after the division
      // of radians/(2*PI) which is then expressed as a positive Q15
      // fraction in the interval [0 , +1) - this is l_delay_rate_index. 
      // l_delay_rate_index should probably be called l_delay_rate_phase
      // (sorry about that!)
      // It is a Q31 positive number of which the high order 16 bits are
      // used when calculating the sine. idx will have a value in the
      // interval [-1 , +1)
      frac = arm_sin_q15( (q15_t)((l_delay_rate_index >> 16) & 0x7fff));
      // multiply the sin by the delay depth
      idx = (frac * delay_depth) >> 15;
//Serial.println(idx);
      // Calculate the offset into the buffer
      idx = l_circ_idx - (delay_offset_idx + idx);
      // and adjust idx to point into the circular buffer
      if(idx < 0) {
        idx += delay_length;
      }
      if(idx >= delay_length) {
        idx -= delay_length;
      }

      // Here we interpolate between two indices but if the sine was negative
      // then we interpolate between idx and idx-1, otherwise the
      // interpolation is between idx and idx+1
      if(frac < 0)
        idx1 = idx - 1;
      else
        idx1 = idx + 1;
      // adjust idx1 in the circular buffer
      if(idx1 < 0) {
        idx1 += delay_length;
      }
      if(idx1 >= delay_length) {
        idx1 -= delay_length;
      }
      // Do the interpolation
      frac = (l_delay_rate_index >> 1) &0x7fff;
      frac = (( (int)(l_delayline[idx1] - l_delayline[idx])*frac) >> 15);
      *bp++ = (l_delayline[l_circ_idx]+ l_delayline[idx] + frac)/2;

      l_delay_rate_index += delay_rate_incr;
      if(l_delay_rate_index & 0x80000000) {
        l_delay_rate_index &= 0x7fffffff;
      }
    }
    // send the effect output to the left channel
    transmit(block,0);
    release(block);
  }
}