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/* 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_sine.h"
#include "utility/dspinst.h"
// data_waveforms.c
extern "C" {
extern const int16_t AudioWaveformSine[257];
}
void AudioSynthWaveformSine::update(void)
{
audio_block_t *block;
uint32_t i, ph, inc, index, scale;
int32_t val1, val2;
if (magnitude) {
block = allocate();
if (block) {
ph = phase_accumulator;
inc = phase_increment;
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;
#if defined(__ARM_ARCH_7EM__)
block->data[i] = multiply_32x32_rshift32(val1 + val2, magnitude);
#elif defined(KINETISL)
block->data[i] = (((val1 + val2) >> 16) * magnitude) >> 16;
#endif
ph += inc;
}
phase_accumulator = ph;
transmit(block);
release(block);
return;
}
}
phase_accumulator += phase_increment * AUDIO_BLOCK_SAMPLES;
}
#if defined(__ARM_ARCH_7EM__)
// High accuracy 11th order Taylor Series Approximation
// input is 0 to 0xFFFFFFFF, representing 0 to 360 degree phase
// output is 32 bit signed integer, top 25 bits should be very good
static int32_t taylor(uint32_t ph)
{
int32_t angle, sum, p1, p2, p3, p5, p7, p9, p11;
if (ph >= 0xC0000000 || ph < 0x40000000) { // ph: 0.32
angle = (int32_t)ph; // valid from -90 to +90 degrees
} else {
angle = (int32_t)(0x80000000u - ph); // angle: 2.30
}
p1 = multiply_32x32_rshift32_rounded(angle, 1686629713) << 2; // p1: 2.30
p2 = multiply_32x32_rshift32_rounded(p1, p1) << 1; // p2: 3.29
p3 = multiply_32x32_rshift32_rounded(p2, p1) << 2; // p3: 3.29
sum = multiply_subtract_32x32_rshift32_rounded(p1, p3, 1431655765); // sum: 2.30
p5 = multiply_32x32_rshift32_rounded(p3, p2); // p5: 6.26
sum = multiply_accumulate_32x32_rshift32_rounded(sum, p5, 572662306);
p7 = multiply_32x32_rshift32_rounded(p5, p2); // p7: 9.23
sum = multiply_subtract_32x32_rshift32_rounded(sum, p7, 109078534);
p9 = multiply_32x32_rshift32_rounded(p7, p2); // p9: 12.20
sum = multiply_accumulate_32x32_rshift32_rounded(sum, p9, 12119837);
p11 = multiply_32x32_rshift32_rounded(p9, p2); // p11: 15.17
sum = multiply_subtract_32x32_rshift32_rounded(sum, p11, 881443);
return sum <<= 1; // return: 1.31
}
#endif
void AudioSynthWaveformSineHires::update(void)
{
#if defined(__ARM_ARCH_7EM__)
audio_block_t *msw, *lsw;
uint32_t i, ph, inc;
int32_t val;
if (magnitude) {
msw = allocate();
lsw = allocate();
if (msw && lsw) {
ph = phase_accumulator;
inc = phase_increment;
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
val = taylor(ph);
msw->data[i] = val >> 16;
lsw->data[i] = val & 0xFFFF;
ph += inc;
}
phase_accumulator = ph;
transmit(msw, 0);
release(msw);
transmit(lsw, 1);
release(lsw);
return;
} else {
if (msw) release(msw);
if (lsw) release(lsw);
}
}
phase_accumulator += phase_increment * AUDIO_BLOCK_SAMPLES;
#endif
}
#if defined(__ARM_ARCH_7EM__)
void AudioSynthWaveformSineModulated::update(void)
{
audio_block_t *block, *modinput;
uint32_t i, ph, inc, index, scale;
int32_t val1, val2;
int16_t mod;
modinput = receiveReadOnly();
ph = phase_accumulator;
inc = phase_increment;
block = allocate();
if (!block) {
// unable to allocate memory, so we'll send nothing
if (modinput) {
// but if we got modulation data, update the phase
for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
mod = modinput->data[i];
ph += inc + (multiply_32x32_rshift32(inc, mod << 16) << 1);
}
release(modinput);
} else {
ph += phase_increment * AUDIO_BLOCK_SAMPLES;
}
phase_accumulator = ph;
return;
}
if (modinput) {
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;
//block->data[i] = (((val1 + val2) >> 16) * magnitude) >> 16;
block->data[i] = multiply_32x32_rshift32(val1 + val2, magnitude);
// -32768 = no phase increment
// 32767 = double phase increment
mod = modinput->data[i];
ph += inc + (multiply_32x32_rshift32(inc, mod << 16) << 1);
//ph += inc + (((int64_t)inc * (mod << 16)) >> 31);
}
release(modinput);
} else {
ph = phase_accumulator;
inc = phase_increment;
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;
block->data[i] = multiply_32x32_rshift32(val1 + val2, magnitude);
ph += inc;
}
}
phase_accumulator = ph;
transmit(block);
release(block);
}
#elif defined(KINETISL)
void AudioSynthWaveformSineModulated::update(void)
{
audio_block_t *block;
block = receiveReadOnly();
if (block) release(block);
}
#endif
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