teensy-audio/synth_waveform.cpp

/* Audio Library for Teensy 3.X
 * Copyright (c) 2018, 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;
	int16_t *bp, *end;
	int32_t val1, val2;
	int16_t magnitude15;
	uint32_t i, ph, index, index2, scale;
	const uint32_t inc = phase_increment;

	ph = phase_accumulator + phase_offset;
	if (magnitude == 0) {
		phase_accumulator += inc * AUDIO_BLOCK_SAMPLES;
		return;
	}
	block = allocate();
	if (!block) {
		phase_accumulator += inc * AUDIO_BLOCK_SAMPLES;
		return;
	}
	bp = block->data;

	switch(tone_type) {
	case WAVEFORM_SINE:
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			index = ph >> 24;
			val1 = AudioWaveformSine[index];
			val2 = AudioWaveformSine[index+1];
			scale = (ph >> 8) & 0xFFFF;
			val2 *= scale;
			val1 *= 0x10000 - scale;
			*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
			ph += inc;
		}
		break;

	case WAVEFORM_ARBITRARY:
		if (!arbdata) {
			release(block);
			phase_accumulator += inc * AUDIO_BLOCK_SAMPLES;
			return;
		}
		// len = 256
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			index = ph >> 24;
			index2 = index + 1;
			if (index2 >= 256) index2 = 0;
			val1 = *(arbdata + index);
			val2 = *(arbdata + index2);
			scale = (ph >> 8) & 0xFFFF;
			val2 *= scale;
			val1 *= 0x10000 - scale;
			*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
			ph += inc;
		}
		break;

	case WAVEFORM_SQUARE:
		magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			if (ph & 0x80000000) {
				*bp++ = -magnitude15;
			} else {
				*bp++ = magnitude15;
			}
			ph += inc;
		}
		break;

	case WAVEFORM_SAWTOOTH:
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			*bp++ = signed_multiply_32x16t(magnitude, ph);
			ph += inc;
		}
		break;

	case WAVEFORM_SAWTOOTH_REVERSE:
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			*bp++ = signed_multiply_32x16t(0xFFFFFFFFu - magnitude, ph);
			ph += inc;
		}
		break;

	case WAVEFORM_TRIANGLE:
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			uint32_t phtop = ph >> 30;
			if (phtop == 1 || phtop == 2) {
				*bp++ = ((0x10000 - (ph >> 15)) * magnitude) >> 16;
			} else {
				*bp++ = ((ph >> 15) * magnitude) >> 16;
			}
			ph += inc;
		}
		break;

	case WAVEFORM_PULSE:
		magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			if (ph < pulse_width) {
				*bp++ = magnitude15;
			} else {
				*bp++ = -magnitude15;
			}
			ph += inc;
		}
		break;

	case WAVEFORM_SAMPLE_HOLD:
		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
			*bp++ = sample;
			uint32_t newph = ph + inc;
			if (newph < ph) {
				sample = random(magnitude) - (magnitude >> 1);
			}
			ph = newph;
		}
		break;
	}
	phase_accumulator = ph - phase_offset;

	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);
}