uilleann/synth_waveform.cpp

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/* 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.
*/
#if defined(ADAFRUIT_TRELLIS_M4_EXPRESS)
#include <Arduino.h>
#include "synth_waveform.h"
#include "arm_math.h"
#include "utility/dspinst.h"
void AudioSynthWaveformModulated::update(void)
{
audio_block_t *block, *moddata, *shapedata;
int16_t *bp, *end;
int32_t val1, val2;
int16_t magnitude15;
uint32_t i, ph, index, index2, scale, priorphase;
const uint32_t inc = phase_increment;
moddata = receiveReadOnly(0);
shapedata = receiveReadOnly(1);
// Pre-compute the phase angle for every output sample of this update
ph = phase_accumulator;
priorphase = phasedata[AUDIO_BLOCK_SAMPLES-1];
if (moddata && modulation_type == 0) {
// Frequency Modulation
bp = moddata->data;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
int32_t n = (*bp++) * modulation_factor; // n is # of octaves to mod
int32_t ipart = n >> 27; // 4 integer bits
n &= 0x7FFFFFF; // 27 fractional bits
#ifdef IMPROVE_EXPONENTIAL_ACCURACY
// exp2 polynomial suggested by Stefan Stenzel on "music-dsp"
// mail list, Wed, 3 Sep 2014 10:08:55 +0200
int32_t x = n << 3;
n = multiply_accumulate_32x32_rshift32_rounded(536870912, x, 1494202713);
int32_t sq = multiply_32x32_rshift32_rounded(x, x);
n = multiply_accumulate_32x32_rshift32_rounded(n, sq, 1934101615);
n = n + (multiply_32x32_rshift32_rounded(sq,
multiply_32x32_rshift32_rounded(x, 1358044250)) << 1);
n = n << 1;
#else
// exp2 algorithm by Laurent de Soras
// https://www.musicdsp.org/en/latest/Other/106-fast-exp2-approximation.html
n = (n + 134217728) << 3;
n = multiply_32x32_rshift32_rounded(n, n);
n = multiply_32x32_rshift32_rounded(n, 715827883) << 3;
n = n + 715827882;
#endif
uint32_t scale = n >> (14 - ipart);
uint64_t phstep = (uint64_t)inc * scale;
uint32_t phstep_msw = phstep >> 32;
if (phstep_msw < 0x7FFE) {
ph += phstep >> 16;
} else {
ph += 0x7FFE0000;
}
phasedata[i] = ph;
}
release(moddata);
} else if (moddata) {
// Phase Modulation
bp = moddata->data;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
// more than +/- 180 deg shift by 32 bit overflow of "n"
uint32_t n = (uint16_t)(*bp++) * modulation_factor;
phasedata[i] = ph + n;
ph += inc;
}
release(moddata);
} else {
// No Modulation Input
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
phasedata[i] = ph;
ph += inc;
}
}
phase_accumulator = ph;
// If the amplitude is zero, no output, but phase still increments properly
if (magnitude == 0) {
if (shapedata) release(shapedata);
return;
}
block = allocate();
if (!block) {
if (shapedata) release(shapedata);
return;
}
bp = block->data;
// Now generate the output samples using the pre-computed phase angles
switch(tone_type) {
case WAVEFORM_SINE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[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);
}
break;
case WAVEFORM_ARBITRARY:
if (!arbdata) {
release(block);
if (shapedata) release(shapedata);
return;
}
// len = 256
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[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);
}
break;
case WAVEFORM_PULSE:
if (shapedata) {
magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
uint32_t width = ((shapedata->data[i] + 0x8000) & 0xFFFF) << 16;
if (phasedata[i] < width) {
*bp++ = magnitude15;
} else {
*bp++ = -magnitude15;
}
}
break;
} // else fall through to orginary square without shape modulation
case WAVEFORM_SQUARE:
magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
if (phasedata[i] & 0x80000000) {
*bp++ = -magnitude15;
} else {
*bp++ = magnitude15;
}
}
break;
case WAVEFORM_SAWTOOTH:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
*bp++ = signed_multiply_32x16t(magnitude, phasedata[i]);
}
break;
case WAVEFORM_SAWTOOTH_REVERSE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
*bp++ = signed_multiply_32x16t(0xFFFFFFFFu - magnitude, phasedata[i]);
}
break;
case WAVEFORM_TRIANGLE_VARIABLE:
if (shapedata) {
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
uint32_t width = (shapedata->data[i] + 0x8000) & 0xFFFF;
uint32_t rise = 0xFFFFFFFF / width;
uint32_t fall = 0xFFFFFFFF / (0xFFFF - width);
uint32_t halfwidth = width << 15;
uint32_t n;
ph = phasedata[i];
if (ph < halfwidth) {
n = (ph >> 16) * rise;
*bp++ = ((n >> 16) * magnitude) >> 16;
} else if (ph < 0xFFFFFFFF - halfwidth) {
n = 0x7FFFFFFF - (((ph - halfwidth) >> 16) * fall);
*bp++ = (((int32_t)n >> 16) * magnitude) >> 16;
} else {
n = ((ph + halfwidth) >> 16) * rise + 0x80000000;
*bp++ = (((int32_t)n >> 16) * magnitude) >> 16;
}
ph += inc;
}
break;
} // else fall through to orginary triangle without shape modulation
case WAVEFORM_TRIANGLE:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[i];
uint32_t phtop = ph >> 30;
if (phtop == 1 || phtop == 2) {
*bp++ = ((0xFFFF - (ph >> 15)) * magnitude) >> 16;
} else {
*bp++ = (((int32_t)ph >> 15) * magnitude) >> 16;
}
}
break;
case WAVEFORM_SAMPLE_HOLD:
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
ph = phasedata[i];
if (ph < priorphase) { // does not work for phase modulation
sample = random(magnitude) - (magnitude >> 1);
}
priorphase = ph;
*bp++ = sample;
}
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);
}
if (shapedata) release(shapedata);
transmit(block, 0);
release(block);
}
#endif