/* Extended to f32 data Created: Chip Audette, OpenAudio, Feb 2017 Included I16 to F32 conversion here. Bob Larkin June 2020 License: MIT License. Use at your own risk. */ /* 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 "synth_pinknoise_f32.h" int16_t AudioSynthNoisePink_F32::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_F32::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_F32::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_F32::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_F32::update(void) { audio_block_t *block; audio_block_f32_t *block_f32; uint32_t *p, *end; int32_t n1, n2; int32_t gain; int32_t inc, dec, accu, bit, lfsr; int32_t taps; if (!enabled) return; gain = level; if (gain == 0) return; block = AudioStream::allocate(); block_f32 = AudioStream_F32::allocate_f32(); if (!block | !block_f32) return; p = (uint32_t *)(block->data); //end = p + AUDIO_BLOCK_SAMPLES/2; end = p + (block_f32->length)/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; //convert int16 to f32 #define I16_TO_F32_NORM_FACTOR (3.051757812500000E-05) //which is 1/32768 for (int i=0; ilength; i++) block_f32->data[i] = (float32_t)block->data[i] * I16_TO_F32_NORM_FACTOR; AudioStream_F32::transmit(block_f32); AudioStream_F32::release(block_f32); AudioStream::release(block); }