teensy-audio/synth_waveform.cpp

/* Audio Library for Teensy 3.X
 * Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
 *
 * Development of this audio library was funded by PJRC.COM, LLC by sales of
 * Teensy and Audio Adaptor boards.  Please support PJRC's efforts to develop
 * open source software by purchasing Teensy or other PJRC products.
 *
 * 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, development funding 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 "synth_waveform.h"
#include "arm_math.h"
#include "utility/dspinst.h"


/******************************************************************/
// PAH 140415 - change sin to use Paul's interpolation which is much
//				faster than arm's sin function
// PAH 140316 - fix calculation of sample (amplitude error)
// PAH 140314 - change t_hi from int to float


void AudioSynthWaveform::update(void)
{
  audio_block_t *block;
  short *bp, *end;
  int32_t val1, val2, val3;
  uint32_t index, scale;
  
  // temporaries for TRIANGLE
  uint32_t mag;
  short tmp_amp;
  
  if(tone_amp == 0) return;
  block = allocate();
  if (block) {
    bp = block->data;
    switch(tone_type) {
    case WAVEFORM_SINE:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
      	// Calculate interpolated sin
		index = tone_phase >> 23;
		val1 = AudioWaveformSine[index];
		val2 = AudioWaveformSine[index+1];
		scale = (tone_phase >> 7) & 0xFFFF;
		val2 *= scale;
		val1 *= 0xFFFF - scale;
		val3 = (val1 + val2) >> 16;
		*bp++ = (short)((val3 * tone_amp) >> 15);
        
        // phase and incr are both unsigned 32-bit fractions
        tone_phase += tone_incr;
        // If tone_phase has overflowed, truncate the top bit 
        if(tone_phase & 0x80000000)tone_phase &= 0x7fffffff;
      }
      break;

    case WAVEFORM_ARBITRARY:
      if (!arbdata) {
		release(block);
		return;
      }
      // len = 256
      for (int i = 0; i < AUDIO_BLOCK_SAMPLES;i++) {
		index = tone_phase >> 23;
		val1 = *(arbdata + index);
		val2 = *(arbdata + ((index + 1) & 255));
		scale = (tone_phase >> 7) & 0xFFFF;
		val2 *= scale;
		val1 *= 0xFFFF - scale;
		val3 = (val1 + val2) >> 16;
		*bp++ = (short)((val3 * tone_amp) >> 15);
		tone_phase += tone_incr;
		tone_phase &= 0x7fffffff;
      }
      break;
      
    case WAVEFORM_SQUARE:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        if(tone_phase & 0x40000000)*bp++ = -tone_amp;
        else *bp++ = tone_amp;
        // phase and incr are both unsigned 32-bit fractions
        tone_phase += tone_incr;
      }
      break;
      
    case WAVEFORM_SAWTOOTH:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        *bp++ = ((short)(tone_phase>>15)*tone_amp) >> 15;
        // phase and incr are both unsigned 32-bit fractions
        tone_phase += tone_incr;    
      }
      break;

    case WAVEFORM_SAWTOOTH_REVERSE:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        *bp++ = ((short)(tone_phase>>15)*tone_amp) >> 15;
         // phase and incr are both unsigned 32-bit fractions
         tone_phase -= tone_incr;
      }
      break;

    case WAVEFORM_TRIANGLE:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        if(tone_phase & 0x80000000) {
          // negative half-cycle
          tmp_amp = -tone_amp;
        } 
        else {
          // positive half-cycle
          tmp_amp = tone_amp;
        }
        mag = tone_phase << 2;
        // Determine which quadrant
        if(tone_phase & 0x40000000) {
          // negate the magnitude
          mag = ~mag + 1;
        }
        *bp++ = ((short)(mag>>17)*tmp_amp) >> 15;
        tone_phase += 2*tone_incr;
      }
      break;
      
    case WAVEFORM_PULSE:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        if(tone_phase < tone_width)*bp++ = -tone_amp;
        else *bp++ = tone_amp;
        tone_phase += tone_incr;
      }
      break;
      
    case WAVEFORM_SAMPLE_HOLD:
      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
        if(tone_phase < tone_incr) {
          sample = random(-tone_amp, tone_amp);
        }
        *bp++ = sample;
        tone_phase += tone_incr;
      }
      break;
    }
    if (tone_offset) {
	bp = block->data;
	end = bp + AUDIO_BLOCK_SAMPLES;
	do {
		val1 = *bp;
		*bp++ = signed_saturate_rshift(val1 + tone_offset, 16, 0);
	} while (bp < end);
    }
    transmit(block,0);
    release(block);
  }
}