trying to locate source of freeze
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/* Audio Library for Teensy 3.X
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* Copyright (c) 2018, Paul Stoffregen, paul@pjrc.com
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*
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* Development of this audio library was funded by PJRC.COM, LLC by sales of
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* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
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* open source software by purchasing Teensy or other PJRC products.
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice, development funding notice, and this permission
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* notice shall be included in all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#if defined(ADAFRUIT_TRELLIS_M4_EXPRESS)
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#include <Arduino.h>
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#include "synth_waveform.h"
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#include "arm_math.h"
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#include "utility/dspinst.h"
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void AudioSynthWaveformModulated::update(void)
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{
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audio_block_t *block, *moddata, *shapedata;
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int16_t *bp, *end;
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int32_t val1, val2;
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int16_t magnitude15;
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uint32_t i, ph, index, index2, scale, priorphase;
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const uint32_t inc = phase_increment;
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moddata = receiveReadOnly(0);
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shapedata = receiveReadOnly(1);
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// Pre-compute the phase angle for every output sample of this update
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ph = phase_accumulator;
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priorphase = phasedata[AUDIO_BLOCK_SAMPLES-1];
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if (moddata && modulation_type == 0) {
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// Frequency Modulation
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bp = moddata->data;
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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int32_t n = (*bp++) * modulation_factor; // n is # of octaves to mod
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int32_t ipart = n >> 27; // 4 integer bits
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n &= 0x7FFFFFF; // 27 fractional bits
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#ifdef IMPROVE_EXPONENTIAL_ACCURACY
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// exp2 polynomial suggested by Stefan Stenzel on "music-dsp"
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// mail list, Wed, 3 Sep 2014 10:08:55 +0200
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int32_t x = n << 3;
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n = multiply_accumulate_32x32_rshift32_rounded(536870912, x, 1494202713);
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int32_t sq = multiply_32x32_rshift32_rounded(x, x);
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n = multiply_accumulate_32x32_rshift32_rounded(n, sq, 1934101615);
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n = n + (multiply_32x32_rshift32_rounded(sq,
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multiply_32x32_rshift32_rounded(x, 1358044250)) << 1);
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n = n << 1;
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#else
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// exp2 algorithm by Laurent de Soras
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// https://www.musicdsp.org/en/latest/Other/106-fast-exp2-approximation.html
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n = (n + 134217728) << 3;
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n = multiply_32x32_rshift32_rounded(n, n);
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n = multiply_32x32_rshift32_rounded(n, 715827883) << 3;
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n = n + 715827882;
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#endif
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uint32_t scale = n >> (14 - ipart);
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uint64_t phstep = (uint64_t)inc * scale;
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uint32_t phstep_msw = phstep >> 32;
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if (phstep_msw < 0x7FFE) {
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ph += phstep >> 16;
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} else {
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ph += 0x7FFE0000;
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}
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phasedata[i] = ph;
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}
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release(moddata);
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} else if (moddata) {
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// Phase Modulation
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bp = moddata->data;
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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// more than +/- 180 deg shift by 32 bit overflow of "n"
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uint32_t n = (uint16_t)(*bp++) * modulation_factor;
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phasedata[i] = ph + n;
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ph += inc;
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}
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release(moddata);
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} else {
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// No Modulation Input
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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phasedata[i] = ph;
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ph += inc;
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}
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}
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phase_accumulator = ph;
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// If the amplitude is zero, no output, but phase still increments properly
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if (magnitude == 0) {
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if (shapedata) release(shapedata);
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return;
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}
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block = allocate();
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if (!block) {
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if (shapedata) release(shapedata);
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return;
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}
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bp = block->data;
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// Now generate the output samples using the pre-computed phase angles
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switch(tone_type) {
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case WAVEFORM_SINE:
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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ph = phasedata[i];
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index = ph >> 24;
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val1 = AudioWaveformSine[index];
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val2 = AudioWaveformSine[index+1];
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scale = (ph >> 8) & 0xFFFF;
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val2 *= scale;
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val1 *= 0x10000 - scale;
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*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
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}
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break;
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case WAVEFORM_ARBITRARY:
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if (!arbdata) {
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release(block);
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if (shapedata) release(shapedata);
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return;
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}
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// len = 256
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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ph = phasedata[i];
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index = ph >> 24;
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index2 = index + 1;
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if (index2 >= 256) index2 = 0;
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val1 = *(arbdata + index);
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val2 = *(arbdata + index2);
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scale = (ph >> 8) & 0xFFFF;
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val2 *= scale;
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val1 *= 0x10000 - scale;
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*bp++ = multiply_32x32_rshift32(val1 + val2, magnitude);
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}
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break;
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case WAVEFORM_PULSE:
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if (shapedata) {
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magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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uint32_t width = ((shapedata->data[i] + 0x8000) & 0xFFFF) << 16;
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if (phasedata[i] < width) {
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*bp++ = magnitude15;
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} else {
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*bp++ = -magnitude15;
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}
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}
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break;
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} // else fall through to orginary square without shape modulation
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case WAVEFORM_SQUARE:
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magnitude15 = signed_saturate_rshift(magnitude, 16, 1);
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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if (phasedata[i] & 0x80000000) {
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*bp++ = -magnitude15;
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} else {
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*bp++ = magnitude15;
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}
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}
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break;
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case WAVEFORM_SAWTOOTH:
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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*bp++ = signed_multiply_32x16t(magnitude, phasedata[i]);
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}
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break;
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case WAVEFORM_SAWTOOTH_REVERSE:
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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*bp++ = signed_multiply_32x16t(0xFFFFFFFFu - magnitude, phasedata[i]);
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}
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break;
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case WAVEFORM_TRIANGLE_VARIABLE:
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if (shapedata) {
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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uint32_t width = (shapedata->data[i] + 0x8000) & 0xFFFF;
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uint32_t rise = 0xFFFFFFFF / width;
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uint32_t fall = 0xFFFFFFFF / (0xFFFF - width);
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uint32_t halfwidth = width << 15;
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uint32_t n;
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ph = phasedata[i];
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if (ph < halfwidth) {
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n = (ph >> 16) * rise;
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*bp++ = ((n >> 16) * magnitude) >> 16;
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} else if (ph < 0xFFFFFFFF - halfwidth) {
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n = 0x7FFFFFFF - (((ph - halfwidth) >> 16) * fall);
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*bp++ = (((int32_t)n >> 16) * magnitude) >> 16;
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} else {
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n = ((ph + halfwidth) >> 16) * rise + 0x80000000;
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*bp++ = (((int32_t)n >> 16) * magnitude) >> 16;
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}
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ph += inc;
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}
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break;
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} // else fall through to orginary triangle without shape modulation
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case WAVEFORM_TRIANGLE:
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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ph = phasedata[i];
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uint32_t phtop = ph >> 30;
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if (phtop == 1 || phtop == 2) {
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*bp++ = ((0xFFFF - (ph >> 15)) * magnitude) >> 16;
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} else {
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*bp++ = (((int32_t)ph >> 15) * magnitude) >> 16;
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}
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}
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break;
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case WAVEFORM_SAMPLE_HOLD:
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
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ph = phasedata[i];
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if (ph < priorphase) { // does not work for phase modulation
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sample = random(magnitude) - (magnitude >> 1);
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}
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priorphase = ph;
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*bp++ = sample;
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}
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break;
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}
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if (tone_offset) {
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bp = block->data;
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end = bp + AUDIO_BLOCK_SAMPLES;
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do {
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val1 = *bp;
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*bp++ = signed_saturate_rshift(val1 + tone_offset, 16, 0);
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} while (bp < end);
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}
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if (shapedata) release(shapedata);
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transmit(block, 0);
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release(block);
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}
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#endif
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@ -0,0 +1,121 @@
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/** Backport modulated waveform to Adafruit fork
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*/
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#pragma once
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/* Audio Library for Teensy 3.X
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* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
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*
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* Development of this audio library was funded by PJRC.COM, LLC by sales of
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* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
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* open source software by purchasing Teensy or other PJRC products.
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice, development funding notice, and this permission
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* notice shall be included in all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#if defined(ADAFRUIT_TRELLIS_M4_EXPRESS)
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#include <Arduino.h>
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#include <Audio.h>
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#include "AudioStream.h"
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#include "arm_math.h"
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#define WAVEFORM_TRIANGLE_VARIABLE 8
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class AudioSynthWaveformModulated : public AudioStream
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{
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public:
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AudioSynthWaveformModulated(void) : AudioStream(2, inputQueueArray),
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phase_accumulator(0), phase_increment(0), modulation_factor(32768),
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magnitude(0), arbdata(NULL), sample(0), tone_offset(0),
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tone_type(WAVEFORM_SINE), modulation_type(0) {
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}
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void frequency(float freq) {
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if (freq < 0.0) {
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freq = 0.0;
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} else if (freq > AUDIO_SAMPLE_RATE_EXACT / 2) {
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freq = AUDIO_SAMPLE_RATE_EXACT / 2;
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}
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phase_increment = freq * (4294967296.0 / AUDIO_SAMPLE_RATE_EXACT);
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if (phase_increment > 0x7FFE0000u) phase_increment = 0x7FFE0000;
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}
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void amplitude(float n) { // 0 to 1.0
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if (n < 0) {
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n = 0;
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} else if (n > 1.0) {
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n = 1.0;
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}
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magnitude = n * 65536.0;
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}
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void offset(float n) {
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if (n < -1.0) {
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n = -1.0;
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} else if (n > 1.0) {
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n = 1.0;
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}
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tone_offset = n * 32767.0;
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}
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void begin(short t_type) {
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tone_type = t_type;
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}
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void begin(float t_amp, float t_freq, short t_type) {
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amplitude(t_amp);
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frequency(t_freq);
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tone_type = t_type;
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}
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void arbitraryWaveform(const int16_t *data, float maxFreq) {
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arbdata = data;
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}
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void frequencyModulation(float octaves) {
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if (octaves > 12.0) {
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octaves = 12.0;
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} else if (octaves < 0.1) {
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octaves = 0.1;
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}
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modulation_factor = octaves * 4096.0;
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modulation_type = 0;
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}
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void phaseModulation(float degrees) {
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if (degrees > 9000.0) {
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degrees = 9000.0;
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} else if (degrees < 30.0) {
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degrees = 30.0;
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}
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modulation_factor = degrees * (65536.0 / 180.0);
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modulation_type = 1;
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}
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virtual void update(void);
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private:
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audio_block_t *inputQueueArray[2];
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uint32_t phase_accumulator;
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uint32_t phase_increment;
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uint32_t modulation_factor;
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int32_t magnitude;
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const int16_t *arbdata;
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uint32_t phasedata[AUDIO_BLOCK_SAMPLES];
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int16_t sample; // for WAVEFORM_SAMPLE_HOLD
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int16_t tone_offset;
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uint8_t tone_type;
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uint8_t modulation_type;
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};
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#endif
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