teensy-audio/synth_sine.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_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