/* 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 #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