Holger Wirtz
4 years ago
parent
fecebf1b4a
commit
6c4a11c91e
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/* Stereo plate reverb for Teensy 4
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* |
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* Author: Piotr Zapart |
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* www.hexefx.com |
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* |
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* Copyright (c) 2020 by Piotr Zapart |
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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 "effect_platervbstereo.h" |
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#include "utility/dspinst.h" |
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#include "synth_waveform.h" |
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#define INP_ALLP_COEFF (0.65) |
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#define LOOP_ALLOP_COEFF (0.65) |
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#define HI_LOSS_FREQ (0.3) |
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#define HI_LOSS_FREQ_MAX (0.08) |
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#define LO_LOSS_FREQ (0.06) |
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#define LFO_AMPL_BITS (5) // 2^LFO_AMPL_BITS will be the LFO amplitude
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#define LFO_AMPL ((1<<LFO_AMPL_BITS) + 1) // lfo amplitude
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#define LFO_READ_OFFSET (LFO_AMPL>>1) // read offset = half the amplitude
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#define LFO_FRAC_BITS (16 - LFO_AMPL_BITS) // fractional part used for linear interpolation
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#define LFO_FRAC_MASK ((1<<LFO_FRAC_BITS)-1) // mask for the above
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#define LFO1_FREQ_HZ (1.37) // LFO1 frequency in Hz
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#define LFO2_FREQ_HZ (1.52) // LFO2 frequency in Hz
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#define RV_MASTER_LOWPASS_F (0.6) // master lowpass scaled frequency coeff.
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extern "C" { |
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extern const int16_t AudioWaveformSine[257]; |
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} |
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#ifdef REVERB_USE_DMAMEM |
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float32_t DMAMEM input_blockL[AUDIO_BLOCK_SAMPLES]; |
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float32_t DMAMEM input_blockR[AUDIO_BLOCK_SAMPLES]; |
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float32_t DMAMEM in_allp1_bufL[224]; // input allpass buffers
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float32_t DMAMEM in_allp2_bufL[420]; |
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float32_t DMAMEM in_allp3_bufL[856]; |
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float32_t DMAMEM in_allp4_bufL[1089]; |
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float32_t DMAMEM in_allp1_bufR[156]; // input allpass buffers
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float32_t DMAMEM in_allp2_bufR[520]; |
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float32_t DMAMEM in_allp3_bufR[956]; |
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float32_t DMAMEM in_allp4_bufR[1289]; |
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float32_t DMAMEM lp_allp1_buf[1303]; // loop allpass buffers
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float32_t DMAMEM lp_allp2_buf[905]; |
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float32_t DMAMEM lp_allp3_buf[1175]; |
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float32_t DMAMEM lp_allp4_buf[1398]; |
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float32_t DMAMEM lp_dly1_buf[1423]; |
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float32_t DMAMEM lp_dly2_buf[1589]; |
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float32_t DMAMEM lp_dly3_buf[1365]; |
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float32_t DMAMEM lp_dly4_buf[1698]; |
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#endif |
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AudioEffectPlateReverb::AudioEffectPlateReverb() : AudioStream(2, inputQueueArray) |
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{ |
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input_attn = 0.5; |
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in_allp_k = INP_ALLP_COEFF; |
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memset(in_allp1_bufL, 0, sizeof(in_allp1_bufL)); |
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memset(in_allp2_bufL, 0, sizeof(in_allp2_bufL)); |
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memset(in_allp3_bufL, 0, sizeof(in_allp3_bufL)); |
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memset(in_allp4_bufL, 0, sizeof(in_allp4_bufL)); |
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in_allp1_idxL = 0; |
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in_allp2_idxL = 0; |
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in_allp3_idxL = 0; |
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in_allp4_idxL = 0; |
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memset(in_allp1_bufR, 0, sizeof(in_allp1_bufR)); |
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memset(in_allp2_bufR, 0, sizeof(in_allp2_bufR)); |
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memset(in_allp3_bufR, 0, sizeof(in_allp3_bufR)); |
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memset(in_allp4_bufR, 0, sizeof(in_allp4_bufR)); |
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in_allp1_idxR = 0; |
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in_allp2_idxR = 0; |
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in_allp3_idxR = 0; |
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in_allp4_idxR = 0; |
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in_allp_out_R = 0; |
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memset(lp_allp1_buf, 0, sizeof(lp_allp1_buf)); |
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memset(lp_allp2_buf, 0, sizeof(lp_allp2_buf)); |
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memset(lp_allp3_buf, 0, sizeof(lp_allp3_buf)); |
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memset(lp_allp4_buf, 0, sizeof(lp_allp4_buf)); |
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lp_allp1_idx = 0; |
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lp_allp2_idx = 0; |
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lp_allp3_idx = 0; |
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lp_allp4_idx = 0; |
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loop_allp_k = LOOP_ALLOP_COEFF; |
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lp_allp_out = 0; |
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memset(lp_dly1_buf, 0, sizeof(lp_dly1_buf)); |
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memset(lp_dly2_buf, 0, sizeof(lp_dly2_buf)); |
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memset(lp_dly3_buf, 0, sizeof(lp_dly3_buf)); |
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memset(lp_dly4_buf, 0, sizeof(lp_dly4_buf)); |
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lp_dly1_idx = 0; |
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lp_dly2_idx = 0; |
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lp_dly3_idx = 0; |
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lp_dly4_idx = 0; |
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lp_hidamp_k = 1.0; |
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lp_lodamp_k = 0.0; |
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lp_lowpass_f = HI_LOSS_FREQ; |
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lp_hipass_f = LO_LOSS_FREQ; |
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lpf1 = 0; |
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lpf2 = 0; |
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lpf3 = 0; |
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lpf4 = 0; |
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hpf1 = 0; |
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hpf2 = 0; |
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hpf3 = 0; |
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hpf4 = 0; |
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master_lowpass_f = RV_MASTER_LOWPASS_F; |
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master_lowpass_l = 0; |
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master_lowpass_r = 0; |
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lfo1_phase_acc = 0; |
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lfo1_adder = (UINT32_MAX + 1)/(AUDIO_SAMPLE_RATE_EXACT * LFO1_FREQ_HZ); |
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lfo2_phase_acc = 0; |
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lfo2_adder = (UINT32_MAX + 1)/(AUDIO_SAMPLE_RATE_EXACT * LFO2_FREQ_HZ);
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} |
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#define sat16(n, rshift) signed_saturate_rshift((n), 16, (rshift)) |
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// TODO: move this to one of the data files, use in output_adat.cpp, output_tdm.cpp, etc
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static const audio_block_t zeroblock = { |
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0, 0, 0, { |
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, |
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#if AUDIO_BLOCK_SAMPLES > 16 |
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, |
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#endif |
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#if AUDIO_BLOCK_SAMPLES > 32 |
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, |
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#endif |
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#if AUDIO_BLOCK_SAMPLES > 48 |
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, |
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#endif |
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#if AUDIO_BLOCK_SAMPLES > 64 |
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, |
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#endif |
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#if AUDIO_BLOCK_SAMPLES > 80 |
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, |
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#endif |
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#if AUDIO_BLOCK_SAMPLES > 96 |
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, |
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#endif |
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#if AUDIO_BLOCK_SAMPLES > 112 |
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, |
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#endif |
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} }; |
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void AudioEffectPlateReverb::update() |
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{ |
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const audio_block_t *blockL, *blockR; |
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#if defined(__ARM_ARCH_7EM__) |
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audio_block_t *outblockL; |
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audio_block_t *outblockR; |
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int i; |
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float32_t input, acc, temp1, temp2; |
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uint16_t temp16; |
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float32_t rv_time; |
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// for LFOs:
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int16_t lfo1_out_sin, lfo1_out_cos, lfo2_out_sin, lfo2_out_cos; |
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int32_t y0, y1; |
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int64_t y; |
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uint32_t idx; |
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blockL = receiveReadOnly(0); |
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blockR = receiveReadOnly(1); |
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outblockL = allocate(); |
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outblockR = allocate(); |
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if (!outblockL || !outblockR) { |
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if (outblockL) release(outblockL); |
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if (outblockR) release(outblockR); |
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if (blockL) release((audio_block_t *)blockL); |
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if (blockR) release((audio_block_t *)blockR); |
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return; |
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} |
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if (!blockL) blockL = &zeroblock; |
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if (!blockR) blockR = &zeroblock; |
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// convert data to float32
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arm_q15_to_float((q15_t *)blockL->data, input_blockL, AUDIO_BLOCK_SAMPLES); |
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arm_q15_to_float((q15_t *)blockR->data, input_blockR, AUDIO_BLOCK_SAMPLES); |
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rv_time = rv_time_k; |
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for (i=0; i < AUDIO_BLOCK_SAMPLES; i++)
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{ |
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// do the LFOs
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lfo1_phase_acc += lfo1_adder; |
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idx = lfo1_phase_acc >> 24; // 8bit lookup table address
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y0 = AudioWaveformSine[idx]; |
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y1 = AudioWaveformSine[idx+1]; |
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idx = lfo1_phase_acc & 0x00FFFFFF; // lower 24 bit = fractional part
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y = (int64_t)y0 * (0x00FFFFFF - idx); |
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y += (int64_t)y1 * idx; |
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lfo1_out_sin = (int32_t) (y >> (32-8)); // 16bit output
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idx = ((lfo1_phase_acc >> 24)+64) & 0xFF; |
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y0 = AudioWaveformSine[idx]; |
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y1 = AudioWaveformSine[idx + 1]; |
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y = (int64_t)y0 * (0x00FFFFFF - idx); |
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y += (int64_t)y1 * idx; |
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lfo1_out_cos = (int32_t) (y >> (32-8)); // 16bit output
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lfo2_phase_acc += lfo2_adder; |
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idx = lfo2_phase_acc >> 24; // 8bit lookup table address
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y0 = AudioWaveformSine[idx]; |
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y1 = AudioWaveformSine[idx+1]; |
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idx = lfo2_phase_acc & 0x00FFFFFF; // lower 24 bit = fractional part
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y = (int64_t)y0 * (0x00FFFFFF - idx); |
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y += (int64_t)y1 * idx; |
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lfo2_out_sin = (int32_t) (y >> (32-8)); //32-8->output 16bit,
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idx = ((lfo2_phase_acc >> 24)+64) & 0xFF; |
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y0 = AudioWaveformSine[idx]; |
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y1 = AudioWaveformSine[idx + 1]; |
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y = (int64_t)y0 * (0x00FFFFFF - idx); |
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y += (int64_t)y1 * idx; |
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lfo2_out_cos = (int32_t) (y >> (32-8)); // 16bit output
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input = input_blockL[i] * input_attn; |
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// chained input allpasses, channel L
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acc = in_allp1_bufL[in_allp1_idxL] + input * in_allp_k;
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in_allp1_bufL[in_allp1_idxL] = input - in_allp_k * acc; |
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input = acc; |
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if (++in_allp1_idxL >= sizeof(in_allp1_bufL)/sizeof(float32_t)) in_allp1_idxL = 0; |
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acc = in_allp2_bufL[in_allp2_idxL] + input * in_allp_k;
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in_allp2_bufL[in_allp2_idxL] = input - in_allp_k * acc; |
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input = acc; |
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if (++in_allp2_idxL >= sizeof(in_allp2_bufL)/sizeof(float32_t)) in_allp2_idxL = 0; |
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acc = in_allp3_bufL[in_allp3_idxL] + input * in_allp_k;
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in_allp3_bufL[in_allp3_idxL] = input - in_allp_k * acc; |
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input = acc; |
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if (++in_allp3_idxL >= sizeof(in_allp3_bufL)/sizeof(float32_t)) in_allp3_idxL = 0; |
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acc = in_allp4_bufL[in_allp4_idxL] + input * in_allp_k;
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in_allp4_bufL[in_allp4_idxL] = input - in_allp_k * acc; |
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in_allp_out_L = acc; |
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if (++in_allp4_idxL >= sizeof(in_allp4_bufL)/sizeof(float32_t)) in_allp4_idxL = 0; |
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input = input_blockR[i] * input_attn; |
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// chained input allpasses, channel R
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acc = in_allp1_bufR[in_allp1_idxR] + input * in_allp_k;
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in_allp1_bufR[in_allp1_idxR] = input - in_allp_k * acc; |
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input = acc; |
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if (++in_allp1_idxR >= sizeof(in_allp1_bufR)/sizeof(float32_t)) in_allp1_idxR = 0; |
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acc = in_allp2_bufR[in_allp2_idxR] + input * in_allp_k;
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in_allp2_bufR[in_allp2_idxR] = input - in_allp_k * acc; |
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input = acc; |
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if (++in_allp2_idxR >= sizeof(in_allp2_bufR)/sizeof(float32_t)) in_allp2_idxR = 0; |
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acc = in_allp3_bufR[in_allp3_idxR] + input * in_allp_k;
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in_allp3_bufR[in_allp3_idxR] = input - in_allp_k * acc; |
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input = acc; |
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if (++in_allp3_idxR >= sizeof(in_allp3_bufR)/sizeof(float32_t)) in_allp3_idxR = 0; |
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acc = in_allp4_bufR[in_allp4_idxR] + input * in_allp_k;
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in_allp4_bufR[in_allp4_idxR] = input - in_allp_k * acc; |
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in_allp_out_R = acc; |
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if (++in_allp4_idxR >= sizeof(in_allp4_bufR)/sizeof(float32_t)) in_allp4_idxR = 0; |
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// input allpases done, start loop allpases
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input = lp_allp_out + in_allp_out_R;
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acc = lp_allp1_buf[lp_allp1_idx] + input * loop_allp_k; // input is the lp allpass chain output
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lp_allp1_buf[lp_allp1_idx] = input - loop_allp_k * acc; |
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input = acc; |
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if (++lp_allp1_idx >= sizeof(lp_allp1_buf)/sizeof(float32_t)) lp_allp1_idx = 0; |
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acc = lp_dly1_buf[lp_dly1_idx]; // read the end of the delay
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lp_dly1_buf[lp_dly1_idx] = input; // write new sample
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input = acc; |
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if (++lp_dly1_idx >= sizeof(lp_dly1_buf)/sizeof(float32_t)) lp_dly1_idx = 0; // update index
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// hi/lo shelving filter
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temp1 = input - lpf1; |
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lpf1 += temp1 * lp_lowpass_f; |
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temp2 = input - lpf1; |
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temp1 = lpf1 - hpf1; |
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hpf1 += temp1 * lp_hipass_f; |
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acc = lpf1 + temp2*lp_hidamp_k + hpf1*lp_lodamp_k; |
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acc = acc * rv_time * rv_time_scaler; // scale by the reveb time
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input = acc + in_allp_out_L; |
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acc = lp_allp2_buf[lp_allp2_idx] + input * loop_allp_k;
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lp_allp2_buf[lp_allp2_idx] = input - loop_allp_k * acc; |
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input = acc; |
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if (++lp_allp2_idx >= sizeof(lp_allp2_buf)/sizeof(float32_t)) lp_allp2_idx = 0; |
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acc = lp_dly2_buf[lp_dly2_idx]; // read the end of the delay
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lp_dly2_buf[lp_dly2_idx] = input; // write new sample
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input = acc; |
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if (++lp_dly2_idx >= sizeof(lp_dly2_buf)/sizeof(float32_t)) lp_dly2_idx = 0; // update index
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// hi/lo shelving filter
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temp1 = input - lpf2; |
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lpf2 += temp1 * lp_lowpass_f; |
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temp2 = input - lpf2; |
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temp1 = lpf2 - hpf2; |
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hpf2 += temp1 * lp_hipass_f; |
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acc = lpf2 + temp2*lp_hidamp_k + hpf2*lp_lodamp_k; |
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acc = acc * rv_time * rv_time_scaler;
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input = acc + in_allp_out_R; |
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acc = lp_allp3_buf[lp_allp3_idx] + input * loop_allp_k;
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lp_allp3_buf[lp_allp3_idx] = input - loop_allp_k * acc; |
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input = acc; |
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if (++lp_allp3_idx >= sizeof(lp_allp3_buf)/sizeof(float32_t)) lp_allp3_idx = 0; |
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acc = lp_dly3_buf[lp_dly3_idx]; // read the end of the delay
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lp_dly3_buf[lp_dly3_idx] = input; // write new sample
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input = acc; |
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if (++lp_dly3_idx >= sizeof(lp_dly3_buf)/sizeof(float32_t)) lp_dly3_idx = 0; // update index
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// hi/lo shelving filter
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temp1 = input - lpf3; |
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lpf3 += temp1 * lp_lowpass_f; |
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temp2 = input - lpf3; |
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temp1 = lpf3 - hpf3; |
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hpf3 += temp1 * lp_hipass_f; |
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acc = lpf3 + temp2*lp_hidamp_k + hpf3*lp_lodamp_k; |
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acc = acc * rv_time * rv_time_scaler;
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input = acc + in_allp_out_L;
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acc = lp_allp4_buf[lp_allp4_idx] + input * loop_allp_k;
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lp_allp4_buf[lp_allp4_idx] = input - loop_allp_k * acc; |
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input = acc; |
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if (++lp_allp4_idx >= sizeof(lp_allp4_buf)/sizeof(float32_t)) lp_allp4_idx = 0; |
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acc = lp_dly4_buf[lp_dly4_idx]; // read the end of the delay
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lp_dly4_buf[lp_dly4_idx] = input; // write new sample
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input = acc; |
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if (++lp_dly4_idx >= sizeof(lp_dly4_buf)/sizeof(float32_t)) lp_dly4_idx= 0; // update index
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// hi/lo shelving filter
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temp1 = input - lpf4; |
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lpf4 += temp1 * lp_lowpass_f; |
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temp2 = input - lpf4; |
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temp1 = lpf4 - hpf4; |
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hpf4 += temp1 * lp_hipass_f; |
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acc = lpf4 + temp2*lp_hidamp_k + hpf4*lp_lodamp_k; |
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acc = acc * rv_time * rv_time_scaler;
|
||||
|
||||
lp_allp_out = acc; |
||||
|
||||
// channel L:
|
||||
#ifdef TAP1_MODULATED |
||||
temp16 = (lp_dly1_idx + lp_dly1_offset_L + (lfo1_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly1_buf)/sizeof(float32_t)); |
||||
temp1 = lp_dly1_buf[temp16++]; // sample now
|
||||
if (temp16 >= sizeof(lp_dly1_buf)/sizeof(float32_t)) temp16 = 0; |
||||
temp2 = lp_dly1_buf[temp16]; // sample next
|
||||
input = (float32_t)(lfo1_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
|
||||
acc = (temp1*(1.0-input) + temp2*input)* 0.8; |
||||
#else |
||||
temp16 = (lp_dly1_idx + lp_dly1_offset_L) % (sizeof(lp_dly1_buf)/sizeof(float32_t)); |
||||
acc = lp_dly1_buf[temp16]* 0.8; |
||||
#endif |
||||
|
||||
|
||||
#ifdef TAP2_MODULATED |
||||
temp16 = (lp_dly2_idx + lp_dly2_offset_L + (lfo1_out_sin>>LFO_FRAC_BITS)) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); |
||||
temp1 = lp_dly2_buf[temp16++]; |
||||
if (temp16 >= sizeof(lp_dly2_buf)/sizeof(float32_t)) temp16 = 0; |
||||
temp2 = lp_dly2_buf[temp16];
|
||||
input = (float32_t)(lfo1_out_sin & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
|
||||
acc += (temp1*(1.0-input) + temp2*input)* 0.7; |
||||
#else |
||||
temp16 = (lp_dly2_idx + lp_dly2_offset_L) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); |
||||
acc += (temp1*(1.0-input) + temp2*input)* 0.6; |
||||
#endif |
||||
|
||||
temp16 = (lp_dly3_idx + lp_dly3_offset_L + (lfo2_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly3_buf)/sizeof(float32_t)); |
||||
temp1 = lp_dly3_buf[temp16++]; |
||||
if (temp16 >= sizeof(lp_dly3_buf)/sizeof(float32_t)) temp16 = 0; |
||||
temp2 = lp_dly3_buf[temp16];
|
||||
input = (float32_t)(lfo2_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
|
||||
acc += (temp1*(1.0-input) + temp2*input)* 0.6; |
||||
|
||||
temp16 = (lp_dly4_idx + lp_dly4_offset_L + (lfo2_out_sin>>LFO_FRAC_BITS)) % (sizeof(lp_dly4_buf)/sizeof(float32_t)); |
||||
temp1 = lp_dly4_buf[temp16++]; |
||||
if (temp16 >= sizeof(lp_dly4_buf)/sizeof(float32_t)) temp16 = 0; |
||||
temp2 = lp_dly4_buf[temp16];
|
||||
input = (float32_t)(lfo2_out_sin & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
|
||||
acc += (temp1*(1.0-input) + temp2*input)* 0.5; |
||||
|
||||
// Master lowpass filter
|
||||
temp1 = acc - master_lowpass_l; |
||||
master_lowpass_l += temp1 * master_lowpass_f; |
||||
|
||||
outblockL->data[i] =(int16_t)(master_lowpass_l * 32767.0); //sat16(output * 30, 0);
|
||||
|
||||
// Channel R
|
||||
#ifdef TAP1_MODULATED |
||||
temp16 = (lp_dly1_idx + lp_dly1_offset_R + (lfo2_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly1_buf)/sizeof(float32_t)); |
||||
temp1 = lp_dly1_buf[temp16++]; // sample now
|
||||
if (temp16 >= sizeof(lp_dly1_buf)/sizeof(float32_t)) temp16 = 0; |
||||
temp2 = lp_dly1_buf[temp16]; // sample next
|
||||
input = (float32_t)(lfo2_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
|
||||
|
||||
acc = (temp1*(1.0-input) + temp2*input)* 0.8; |
||||
#else |
||||
temp16 = (lp_dly1_idx + lp_dly1_offset_R) % (sizeof(lp_dly1_buf)/sizeof(float32_t)); |
||||
acc = lp_dly1_buf[temp16] * 0.8; |
||||
#endif |
||||
#ifdef TAP2_MODULATED |
||||
temp16 = (lp_dly2_idx + lp_dly2_offset_R + (lfo1_out_cos>>LFO_FRAC_BITS)) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); |
||||
temp1 = lp_dly2_buf[temp16++]; |
||||
if (temp16 >= sizeof(lp_dly2_buf)/sizeof(float32_t)) temp16 = 0; |
||||
temp2 = lp_dly2_buf[temp16];
|
||||
input = (float32_t)(lfo1_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
|
||||
acc += (temp1*(1.0-input) + temp2*input)* 0.7; |
||||
#else |
||||
temp16 = (lp_dly2_idx + lp_dly2_offset_R) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); |
||||
acc += (temp1*(1.0-input) + temp2*input)* 0.7; |
||||
#endif |
||||
temp16 = (lp_dly3_idx + lp_dly3_offset_R + (lfo2_out_sin>>LFO_FRAC_BITS)) % (sizeof(lp_dly3_buf)/sizeof(float32_t)); |
||||
temp1 = lp_dly3_buf[temp16++]; |
||||
if (temp16 >= sizeof(lp_dly3_buf)/sizeof(float32_t)) temp16 = 0; |
||||
temp2 = lp_dly3_buf[temp16];
|
||||
input = (float32_t)(lfo2_out_sin & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
|
||||
acc += (temp1*(1.0-input) + temp2*input)* 0.6; |
||||
|
||||
temp16 = (lp_dly4_idx + lp_dly4_offset_R + (lfo1_out_sin>>LFO_FRAC_BITS)) % (sizeof(lp_dly4_buf)/sizeof(float32_t)); |
||||
temp1 = lp_dly4_buf[temp16++]; |
||||
if (temp16 >= sizeof(lp_dly4_buf)/sizeof(float32_t)) temp16 = 0; |
||||
temp2 = lp_dly4_buf[temp16];
|
||||
input = (float32_t)(lfo2_out_cos & LFO_FRAC_MASK) / ((float32_t)LFO_FRAC_MASK); // interp. k
|
||||
acc += (temp1*(1.0-input) + temp2*input)* 0.5; |
||||
|
||||
// Master lowpass filter
|
||||
temp1 = acc - master_lowpass_r; |
||||
master_lowpass_r += temp1 * master_lowpass_f; |
||||
outblockR->data[i] =(int16_t)(master_lowpass_r * 32767.0); |
||||
|
||||
} |
||||
transmit(outblockL, 0); |
||||
transmit(outblockR, 1); |
||||
release(outblockL); |
||||
release(outblockR); |
||||
if (blockL != &zeroblock) release((audio_block_t *)blockL); |
||||
if (blockR != &zeroblock) release((audio_block_t *)blockR); |
||||
|
||||
#elif defined(KINETISL) |
||||
blockL = receiveReadOnly(0); |
||||
if (blockL) release(blockL); |
||||
blockR = receiveReadOnly(1); |
||||
if (blockR) release(blockR); |
||||
#endif |
||||
} |
||||
@ -0,0 +1,211 @@ |
||||
/* Stereo plate reverb for Teensy 4
|
||||
* |
||||
* Author: Piotr Zapart |
||||
* www.hexefx.com |
||||
* |
||||
* Copyright (c) 2020 by Piotr Zapart |
||||
* |
||||
* 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 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. |
||||
*/ |
||||
|
||||
/***
|
||||
* Algorithm based on plate reverbs developed for SpinSemi FV-1 DSP chip |
||||
*
|
||||
* Allpass + modulated delay line based lush plate reverb |
||||
*
|
||||
* Input parameters are float in range 0.0 to 1.0: |
||||
*
|
||||
* size - reverb time |
||||
* hidamp - hi frequency loss in the reverb tail |
||||
* lodamp - low frequency loss in the reverb tail |
||||
* lowpass - output/master lowpass filter, useful for darkening the reverb sound
|
||||
* diffusion - lower settings will make the reverb tail more "echoey", optimal value 0.65 |
||||
*
|
||||
*/ |
||||
|
||||
|
||||
#ifndef _EFFECT_PLATERVBSTEREO_H |
||||
#define _EFFECT_PLATERVBSTEREO_H |
||||
|
||||
#include <Arduino.h> |
||||
#include "Audio.h" |
||||
#include "AudioStream.h" |
||||
#include "arm_math.h" |
||||
|
||||
|
||||
// if uncommented will place all the buffers in the DMAMEM section ofd the memory
|
||||
// works with single instance of the reverb only
|
||||
#define REVERB_USE_DMAMEM |
||||
|
||||
/***
|
||||
* Loop delay modulation: comment/uncomment to switch sin/cos
|
||||
* modulation for the 1st or 2nd tap, 3rd tap is always modulated |
||||
* more modulation means more chorus type sounding reverb tail |
||||
*/ |
||||
//#define TAP1_MODULATED
|
||||
#define TAP2_MODULATED |
||||
|
||||
class AudioEffectPlateReverb : public AudioStream |
||||
{ |
||||
public: |
||||
AudioEffectPlateReverb(); |
||||
virtual void update(); |
||||
|
||||
void size(float n) |
||||
{ |
||||
n = constrain(n, 0.0, 1.0); |
||||
n = map (n, 0.0, 1.0, 0.2, rv_time_k_max); |
||||
float32_t attn = 0.5 * map(n, 0.0, rv_time_k_max, 0.5, 1.0); |
||||
AudioNoInterrupts(); |
||||
rv_time_k = n; |
||||
input_attn = attn; |
||||
AudioInterrupts(); |
||||
} |
||||
|
||||
void hidamp(float n) |
||||
{ |
||||
n = constrain(n, 0.0, 1.0); |
||||
AudioNoInterrupts(); |
||||
lp_hidamp_k = 1.0 - n; |
||||
AudioInterrupts(); |
||||
} |
||||
|
||||
void lodamp(float n) |
||||
{ |
||||
n = constrain(n, 0.0, 1.0); |
||||
AudioNoInterrupts(); |
||||
lp_lodamp_k = -n; |
||||
rv_time_scaler = 1.0 - n * 0.12; // limit the max reverb time, otherwise it will clip
|
||||
AudioInterrupts(); |
||||
} |
||||
|
||||
void lowpass(float n) |
||||
{ |
||||
n = constrain(n, 0.0, 1.0); |
||||
n = map(n*n*n, 0.0, 1.0, 0.05, 1.0); |
||||
master_lowpass_f = n; |
||||
} |
||||
|
||||
void diffusion(float n) |
||||
{ |
||||
n = constrain(n, 0.0, 1.0); |
||||
n = map(n, 0.0, 1.0, 0.005, 0.65); |
||||
AudioNoInterrupts(); |
||||
in_allp_k = n; |
||||
loop_allp_k = n; |
||||
AudioInterrupts();
|
||||
} |
||||
|
||||
float32_t get_size(void) {return rv_time_k;} |
||||
private: |
||||
audio_block_t *inputQueueArray[2]; |
||||
#ifndef REVERB_USE_DMAMEM |
||||
float32_t input_blockL[AUDIO_BLOCK_SAMPLES]; |
||||
float32_t input_blockR[AUDIO_BLOCK_SAMPLES]; |
||||
#endif |
||||
float32_t input_attn; |
||||
|
||||
float32_t in_allp_k; // input allpass coeff (default 0.6)
|
||||
#ifndef REVERB_USE_DMAMEM |
||||
float32_t in_allp1_bufL[224]; // input allpass buffers
|
||||
float32_t in_allp2_bufL[420]; |
||||
float32_t in_allp3_bufL[856]; |
||||
float32_t in_allp4_bufL[1089]; |
||||
#endif |
||||
uint16_t in_allp1_idxL; |
||||
uint16_t in_allp2_idxL; |
||||
uint16_t in_allp3_idxL; |
||||
uint16_t in_allp4_idxL; |
||||
float32_t in_allp_out_L; // L allpass chain output
|
||||
#ifndef REVERB_USE_DMAMEM |
||||
float32_t in_allp1_bufR[156]; // input allpass buffers
|
||||
float32_t in_allp2_bufR[520]; |
||||
float32_t in_allp3_bufR[956]; |
||||
float32_t in_allp4_bufR[1289]; |
||||
#endif |
||||
uint16_t in_allp1_idxR; |
||||
uint16_t in_allp2_idxR; |
||||
uint16_t in_allp3_idxR; |
||||
uint16_t in_allp4_idxR; |
||||
float32_t in_allp_out_R; // R allpass chain output
|
||||
#ifndef REVERB_USE_DMAMEM |
||||
float32_t lp_allp1_buf[1303]; // loop allpass buffers
|
||||
float32_t lp_allp2_buf[1905]; |
||||
float32_t lp_allp3_buf[1175]; |
||||
float32_t lp_allp4_buf[1398]; |
||||
#endif |
||||
uint16_t lp_allp1_idx; |
||||
uint16_t lp_allp2_idx; |
||||
uint16_t lp_allp3_idx; |
||||
uint16_t lp_allp4_idx; |
||||
float32_t loop_allp_k; // loop allpass coeff (default 0.6)
|
||||
float32_t lp_allp_out; |
||||
#ifndef REVERB_USE_DMAMEM |
||||
float32_t lp_dly1_buf[1423]; |
||||
float32_t lp_dly2_buf[1589]; |
||||
float32_t lp_dly3_buf[1365]; |
||||
float32_t lp_dly4_buf[1698]; |
||||
#endif |
||||
uint16_t lp_dly1_idx; |
||||
uint16_t lp_dly2_idx; |
||||
uint16_t lp_dly3_idx; |
||||
uint16_t lp_dly4_idx; |
||||
|
||||
const uint16_t lp_dly1_offset_L = 201; |
||||
const uint16_t lp_dly2_offset_L = 145; |
||||
const uint16_t lp_dly3_offset_L = 1897; |
||||
const uint16_t lp_dly4_offset_L = 280; |
||||
|
||||
const uint16_t lp_dly1_offset_R = 1897; |
||||
const uint16_t lp_dly2_offset_R = 1245; |
||||
const uint16_t lp_dly3_offset_R = 487; |
||||
const uint16_t lp_dly4_offset_R = 780;
|
||||
|
||||
float32_t lp_hidamp_k; // loop high band damping coeff
|
||||
float32_t lp_lodamp_k; // loop low baand damping coeff
|
||||
|
||||
float32_t lpf1; // lowpass filters
|
||||
float32_t lpf2; |
||||
float32_t lpf3; |
||||
float32_t lpf4; |
||||
|
||||
float32_t hpf1; // highpass filters
|
||||
float32_t hpf2; |
||||
float32_t hpf3; |
||||
float32_t hpf4; |
||||
|
||||
float32_t lp_lowpass_f; // loop lowpass scaled frequency
|
||||
float32_t lp_hipass_f; // loop highpass scaled frequency
|
||||
|
||||
float32_t master_lowpass_f; |
||||
float32_t master_lowpass_l; |
||||
float32_t master_lowpass_r; |
||||
|
||||
const float32_t rv_time_k_max = 0.95; |
||||
float32_t rv_time_k; // reverb time coeff
|
||||
float32_t rv_time_scaler; // with high lodamp settings lower the max reverb time to avoid clipping
|
||||
|
||||
uint32_t lfo1_phase_acc; // LFO 1
|
||||
uint32_t lfo1_adder; |
||||
|
||||
uint32_t lfo2_phase_acc; // LFO 2
|
||||
uint32_t lfo2_adder; |
||||
}; |
||||
|
||||
#endif // _EFFECT_PLATEREV_H
|
||||
Loading…
Reference in new issue