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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.
- */
-
- // http://stenzel.waldorfmusic.de/post/pink/
- // https://github.com/Stenzel/newshadeofpink
- // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- // New Shade of Pink
- // (c) 2014 Stefan Stenzel
- // stefan at waldorfmusic.de
- //
- // Terms of use:
- // Use for any purpose. If used in a commercial product, you should give me one.
- // - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
-
- #include <Arduino.h>
- #include "synth_pinknoise.h"
-
- int16_t AudioSynthNoisePink::instance_cnt = 0;
-
- // Let preprocessor and compiler calculate two lookup tables for 12-tap FIR Filter
- // with these coefficients: 1.190566, 0.162580, 0.002208, 0.025475, -0.001522,
- // 0.007322, 0.001774, 0.004529, -0.001561, 0.000776, -0.000486, 0.002017
- #define Fn(cf,m,shift) (2048*cf*(2*((m)>>shift&1)-1))
- #define FA(n) (int32_t)(Fn(1.190566,n,0)+Fn(0.162580,n,1)+Fn(0.002208,n,2)+\
- Fn(0.025475,n,3)+Fn(-0.001522,n,4)+Fn(0.007322,n,5))
- #define FB(n) (int32_t)(Fn(0.001774,n,0)+Fn(0.004529,n,1)+Fn(-0.001561,n,2)+\
- Fn(0.000776,n,3)+Fn(-0.000486,n,4)+Fn(0.002017,n,5))
- #define FA8(n) FA(n),FA(n+1),FA(n+2),FA(n+3),FA(n+4),FA(n+5),FA(n+6),FA(n+7)
- #define FB8(n) FB(n),FB(n+1),FB(n+2),FB(n+3),FB(n+4),FB(n+5),FB(n+6),FB(n+7)
- const int32_t AudioSynthNoisePink::pfira[64] = // 1st FIR lookup table
- {FA8(0),FA8(8),FA8(16),FA8(24),FA8(32),FA8(40),FA8(48),FA8(56)};
- const int32_t AudioSynthNoisePink::pfirb[64] = // 2nd FIR lookup table
- {FB8(0),FB8(8),FB8(16),FB8(24),FB8(32),FB8(40),FB8(48),FB8(56)};
-
- // bitreversed lookup table
- #define PM16(n) n,0x80,0x40,0x80,0x20,0x80,0x40,0x80,0x10,0x80,0x40,0x80,0x20,0x80,0x40,0x80
- const uint8_t AudioSynthNoisePink::pnmask[256] = {
- PM16(0),PM16(8),PM16(4),PM16(8),PM16(2),PM16(8),PM16(4),PM16(8),
- PM16(1),PM16(8),PM16(4),PM16(8),PM16(2),PM16(8),PM16(4),PM16(8)
- };
-
- #define PINT(bitmask, out) /* macro for processing: */\
- bit = lfsr >> 31; /* spill random to all bits */\
- dec &= ~bitmask; /* blank old decrement bit */\
- lfsr <<= 1; /* shift lfsr */\
- dec |= inc & bitmask; /* copy increment to decrement bit */\
- inc ^= bit & bitmask; /* new random bit */\
- accu += inc - dec; /* integrate */\
- lfsr ^= bit & taps; /* update lfsr */\
- out = accu + /* save output */\
- pfira[lfsr & 0x3F] + /* add 1st half precalculated FIR */\
- pfirb[lfsr >> 6 & 0x3F] /* add 2nd half, also correts bias */
-
- void AudioSynthNoisePink::update(void)
- {
- audio_block_t *block;
- uint32_t *p, *end;
- int32_t n1, n2;
- int32_t gain;
- int32_t inc, dec, accu, bit, lfsr;
- int32_t taps;
-
- gain = level;
- if (gain == 0) return;
- block = allocate();
- if (!block) return;
- p = (uint32_t *)(block->data);
- end = p + AUDIO_BLOCK_SAMPLES/2;
- taps = 0x46000001;
- inc = pinc;
- dec = pdec;
- accu = paccu;
- lfsr = plfsr;
- do {
- int32_t mask = pnmask[pncnt++];
- PINT(mask, n1);
- n1 = signed_multiply_32x16b(gain, n1);
- PINT(0x0800, n2);
- n2 = signed_multiply_32x16b(gain, n2);
- *p++ = pack_16b_16b(n2, n1);
- PINT(0x0400, n1);
- n1 = signed_multiply_32x16b(gain, n1);
- PINT(0x0800, n2);
- n2 = signed_multiply_32x16b(gain, n2);
- *p++ = pack_16b_16b(n2, n1);
- PINT(0x0200, n1);
- n1 = signed_multiply_32x16b(gain, n1);
- PINT(0x0800, n2);
- n2 = signed_multiply_32x16b(gain, n2);
- *p++ = pack_16b_16b(n2, n1);
- PINT(0x0400, n1);
- n1 = signed_multiply_32x16b(gain, n1);
- PINT(0x0800, n2);
- n2 = signed_multiply_32x16b(gain, n2);
- *p++ = pack_16b_16b(n2, n1);
- PINT(0x0100, n1);
- n1 = signed_multiply_32x16b(gain, n1);
- PINT(0x0800, n2);
- n2 = signed_multiply_32x16b(gain, n2);
- *p++ = pack_16b_16b(n2, n1);
- PINT(0x0400, n1);
- n1 = signed_multiply_32x16b(gain, n1);
- PINT(0x0800, n2);
- n2 = signed_multiply_32x16b(gain, n2);
- *p++ = pack_16b_16b(n2, n1);
- PINT(0x0200, n1);
- n1 = signed_multiply_32x16b(gain, n1);
- PINT(0x0800, n2);
- n2 = signed_multiply_32x16b(gain, n2);
- *p++ = pack_16b_16b(n2, n1);
- PINT(0x0400, n1);
- n1 = signed_multiply_32x16b(gain, n1);
- PINT(0x0800, n2);
- n2 = signed_multiply_32x16b(gain, n2);
- *p++ = pack_16b_16b(n2, n1);
- } while (p < end);
- pinc = inc;
- pdec = dec;
- paccu = accu;
- plfsr = lfsr;
- transmit(block);
- release(block);
- }
-
-
-
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