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@ -1,403 +0,0 @@ |
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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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#include <Arduino.h> |
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#include "synth_waveform_extended.h" |
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#include "arm_math.h" |
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#include "utility/dspinst.h" |
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// uncomment for more accurate but more computationally expensive frequency modulation
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//#define IMPROVE_EXPONENTIAL_ACCURACY
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void AudioSynthWaveformExtended::update(void) |
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{ |
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audio_block_t *block; |
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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; |
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const uint32_t inc = phase_increment; |
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ph = phase_accumulator + phase_offset; |
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if (magnitude == 0) { |
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phase_accumulator += inc * AUDIO_BLOCK_SAMPLES; |
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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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phase_accumulator += inc * AUDIO_BLOCK_SAMPLES; |
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return; |
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} |
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bp = block->data; |
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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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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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ph += inc; |
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} |
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break; |
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|
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case WAVEFORM_ARBITRARY: |
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if (!arbdata) { |
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release(block); |
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phase_accumulator += inc * AUDIO_BLOCK_SAMPLES; |
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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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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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ph += inc; |
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} |
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break; |
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|
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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 (ph & 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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ph += inc; |
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} |
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break; |
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|
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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, ph); |
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ph += inc; |
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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, ph); |
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ph += inc; |
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} |
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break; |
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|
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case WAVEFORM_TRIANGLE: |
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for (i=0; i < AUDIO_BLOCK_SAMPLES; 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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ph += inc; |
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} |
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break; |
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case WAVEFORM_TRIANGLE_VARIABLE: |
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do { |
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uint32_t rise = 0xFFFFFFFF / (pulse_width >> 16); |
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uint32_t fall = 0xFFFFFFFF / (0xFFFF - (pulse_width >> 16)); |
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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if (ph < pulse_width/2) { |
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uint32_t n = (ph >> 16) * rise; |
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*bp++ = ((n >> 16) * magnitude) >> 16; |
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} else if (ph < 0xFFFFFFFF - pulse_width/2) { |
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uint32_t n = 0x7FFFFFFF - (((ph - pulse_width/2) >> 16) * fall); |
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*bp++ = (((int32_t)n >> 16) * magnitude) >> 16; |
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} else { |
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uint32_t n = ((ph + pulse_width/2) >> 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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} while (0); |
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break; |
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case WAVEFORM_PULSE: |
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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 (ph < pulse_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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ph += inc; |
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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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*bp++ = sample; |
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uint32_t newph = ph + inc; |
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if (newph < ph) { |
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sample = random(magnitude) - (magnitude >> 1); |
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} |
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ph = newph; |
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} |
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break; |
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} |
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phase_accumulator = ph - phase_offset; |
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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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transmit(block, 0); |
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release(block); |
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} |
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//--------------------------------------------------------------------------------
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void AudioSynthWaveformExtendedModulated::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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@ -1,221 +0,0 @@ |
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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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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 |
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THE SOFTWARE. |
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*/ |
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|
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/*
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Extension for setting the waveform while running by H. Wirtz <wirtz@parasitstudio.de> |
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*/ |
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#ifndef synth_waveform_extended_h_ |
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#define synth_waveform_extended_h_ |
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#include <Arduino.h> |
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#include "AudioStream.h" |
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#include "arm_math.h" |
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// waveforms.c
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extern "C" { |
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extern const int16_t AudioWaveformSine[257]; |
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} |
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#define WAVEFORM_SINE 0 |
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#define WAVEFORM_SAWTOOTH 1 |
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#define WAVEFORM_SQUARE 2 |
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#define WAVEFORM_TRIANGLE 3 |
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#define WAVEFORM_ARBITRARY 4 |
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#define WAVEFORM_PULSE 5 |
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#define WAVEFORM_SAWTOOTH_REVERSE 6 |
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#define WAVEFORM_SAMPLE_HOLD 7 |
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#define WAVEFORM_TRIANGLE_VARIABLE 8 |
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class AudioSynthWaveformExtended : public AudioStream |
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{ |
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public: |
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AudioSynthWaveformExtended(void) : AudioStream(0, NULL), |
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phase_accumulator(0), phase_increment(0), phase_offset(0), |
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magnitude(0), pulse_width(0x40000000), |
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arbdata(NULL), sample(0), tone_type(WAVEFORM_SINE), |
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tone_offset(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 phase(float angle) { |
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if (angle < 0.0) { |
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angle = 0.0; |
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} else if (angle > 360.0) { |
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angle = angle - 360.0; |
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if (angle >= 360.0) return; |
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} |
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phase_offset = angle * (4294967296.0 / 360.0); |
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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 pulseWidth(float n) { // 0.0 to 1.0
|
||||
if (n < 0) { |
||||
n = 0; |
||||
} else if (n > 1.0) { |
||||
n = 1.0; |
||||
} |
||||
pulse_width = n * 4294967296.0; |
||||
} |
||||
void waveform(short t_type) { |
||||
phase_offset = 0; |
||||
tone_type = t_type; |
||||
} |
||||
void begin(short t_type) { |
||||
phase_offset = 0; |
||||
tone_type = t_type; |
||||
} |
||||
void begin(float t_amp, float t_freq, short t_type) { |
||||
amplitude(t_amp); |
||||
frequency(t_freq); |
||||
phase_offset = 0; |
||||
tone_type = t_type; |
||||
} |
||||
void arbitraryWaveform(const int16_t *data, float maxFreq) { |
||||
arbdata = data; |
||||
} |
||||
virtual void update(void); |
||||
|
||||
private: |
||||
uint32_t phase_accumulator; |
||||
uint32_t phase_increment; |
||||
uint32_t phase_offset; |
||||
int32_t magnitude; |
||||
uint32_t pulse_width; |
||||
const int16_t *arbdata; |
||||
int16_t sample; // for WAVEFORM_SAMPLE_HOLD
|
||||
short tone_type; |
||||
int16_t tone_offset; |
||||
}; |
||||
|
||||
|
||||
class AudioSynthWaveformExtendedModulated : public AudioStream |
||||
{ |
||||
public: |
||||
AudioSynthWaveformExtendedModulated(void) : AudioStream(2, inputQueueArray), |
||||
phase_accumulator(0), phase_increment(0), modulation_factor(32768), |
||||
magnitude(0), arbdata(NULL), sample(0), tone_offset(0), |
||||
tone_type(WAVEFORM_SINE), modulation_type(0) { |
||||
} |
||||
|
||||
void frequency(float freq) { |
||||
if (freq < 0.0) { |
||||
freq = 0.0; |
||||
} else if (freq > AUDIO_SAMPLE_RATE_EXACT / 2) { |
||||
freq = AUDIO_SAMPLE_RATE_EXACT / 2; |
||||
} |
||||
phase_increment = freq * (4294967296.0 / AUDIO_SAMPLE_RATE_EXACT); |
||||
if (phase_increment > 0x7FFE0000u) phase_increment = 0x7FFE0000; |
||||
} |
||||
void amplitude(float n) { // 0 to 1.0
|
||||
if (n < 0) { |
||||
n = 0; |
||||
} else if (n > 1.0) { |
||||
n = 1.0; |
||||
} |
||||
magnitude = n * 65536.0; |
||||
} |
||||
void offset(float n) { |
||||
if (n < -1.0) { |
||||
n = -1.0; |
||||
} else if (n > 1.0) { |
||||
n = 1.0; |
||||
} |
||||
tone_offset = n * 32767.0; |
||||
} |
||||
void waveform(short t_type) { |
||||
tone_type = t_type; |
||||
} |
||||
void begin(short t_type) { |
||||
tone_type = t_type; |
||||
} |
||||
void begin(float t_amp, float t_freq, short t_type) { |
||||
amplitude(t_amp); |
||||
frequency(t_freq); |
||||
tone_type = t_type; |
||||
} |
||||
void arbitraryWaveform(const int16_t *data, float maxFreq) { |
||||
arbdata = data; |
||||
} |
||||
void frequencyModulation(float octaves) { |
||||
if (octaves > 12.0) { |
||||
octaves = 12.0; |
||||
} else if (octaves < 0.1) { |
||||
octaves = 0.1; |
||||
} |
||||
modulation_factor = octaves * 4096.0; |
||||
modulation_type = 0; |
||||
} |
||||
void phaseModulation(float degrees) { |
||||
if (degrees > 9000.0) { |
||||
degrees = 9000.0; |
||||
} else if (degrees < 30.0) { |
||||
degrees = 30.0; |
||||
} |
||||
modulation_factor = degrees * (65536.0 / 180.0); |
||||
modulation_type = 1; |
||||
} |
||||
virtual void update(void); |
||||
|
||||
private: |
||||
audio_block_t *inputQueueArray[2]; |
||||
uint32_t phase_accumulator; |
||||
uint32_t phase_increment; |
||||
uint32_t modulation_factor; |
||||
int32_t magnitude; |
||||
const int16_t *arbdata; |
||||
uint32_t phasedata[AUDIO_BLOCK_SAMPLES]; |
||||
int16_t sample; // for WAVEFORM_SAMPLE_HOLD
|
||||
int16_t tone_offset; |
||||
uint8_t tone_type; |
||||
uint8_t modulation_type; |
||||
}; |
||||
|
||||
|
||||
#endif |
||||
Loading…
Reference in new issue