From 52e51de63a22408ef245a126fd8e24c9aeac32d1 Mon Sep 17 00:00:00 2001 From: Holger Wirtz Date: Wed, 30 Mar 2022 14:04:04 +0200 Subject: [PATCH] First try for adding a reverb effect. --- src/Makefile | 3 +- src/effect_platervbstereo.cpp | 493 ++++++++++++++++++++++++++++++++++ src/effect_platervbstereo.h | 222 +++++++++++++++ src/minidexed.cpp | 17 ++ src/minidexed.h | 3 + 5 files changed, 737 insertions(+), 1 deletion(-) create mode 100644 src/effect_platervbstereo.cpp create mode 100644 src/effect_platervbstereo.h diff --git a/src/Makefile b/src/Makefile index b9f2195..352dfca 100644 --- a/src/Makefile +++ b/src/Makefile @@ -8,7 +8,8 @@ CMSIS_DIR = ../CMSIS_5/CMSIS OBJS = main.o kernel.o minidexed.o config.o userinterface.o \ mididevice.o midikeyboard.o serialmididevice.o pckeyboard.o \ - sysexfileloader.o performanceconfig.o perftimer.o + sysexfileloader.o performanceconfig.o perftimer.o \ + effect_platervbstereo.o include ./Synth_Dexed.mk include ./Rules.mk diff --git a/src/effect_platervbstereo.cpp b/src/effect_platervbstereo.cpp new file mode 100644 index 0000000..be0e8ad --- /dev/null +++ b/src/effect_platervbstereo.cpp @@ -0,0 +1,493 @@ +/* Stereo plate reverb for Teensy 4 + * + * Adapted for use in MiniDexed (Holger Wirtz ) + * + * Author: Piotr Zapart + * www.hexefx.com + * + * Copyright (c) 2020 by Piotr Zapart + * + * Development of this audio library was funded by PJRC.COM, LLC by sales of + * Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop + * open source software by purchasing Teensy or other PJRC products. + * + * Permission is hereby granted, free of charge, to any person obtaining a copy + * of this software and associated documentation files (the "Software"), to deal + * in the Software without restriction, including without limitation the rights + * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell + * copies of the Software, and to permit persons to whom the Software is + * furnished to do so, subject to the following conditions: + * + * The above copyright notice, development funding notice, and this permission + * notice shall be included in all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE + * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER + * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, + * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN + * THE SOFTWARE. + */ + + +#include +#include +#include "effect_platervbstereo.h" + +#define INP_ALLP_COEFF (0.65f) // default input allpass coeff +#define LOOP_ALLOP_COEFF (0.65f) // default loop allpass coeff + +#define HI_LOSS_FREQ (0.3f) // scaled center freq for the treble loss filter +// #define HI_LOSS_FREQ_MAX (0.08f) +#define LO_LOSS_FREQ (0.06f) // scaled center freq for the bass loss filter + +#define LFO_AMPL_BITS (5) // 2^LFO_AMPL_BITS will be the LFO amplitude +#define LFO_AMPL ((1<>1) // read offset = half the amplitude +#define LFO_FRAC_BITS (16 - LFO_AMPL_BITS) // fractional part used for linear interpolation +#define LFO_FRAC_MASK ((1<> 24; // 8bit lookup table address + y0 = AudioWaveformSine[idx]; + y1 = AudioWaveformSine[idx+1]; + idx = lfo1_phase_acc & 0x00FFFFFF; // lower 24 bit = fractional part + y = (int64_t)y0 * (0x00FFFFFF - idx); + y += (int64_t)y1 * idx; + lfo1_out_sin = (int32_t) (y >> (32-8)); // 16bit output + idx = ((lfo1_phase_acc >> 24)+64) & 0xFF; + y0 = AudioWaveformSine[idx]; + y1 = AudioWaveformSine[idx + 1]; + y = (int64_t)y0 * (0x00FFFFFF - idx); + y += (int64_t)y1 * idx; + lfo1_out_cos = (int32_t) (y >> (32-8)); // 16bit output + + lfo2_phase_acc += lfo2_adder; + idx = lfo2_phase_acc >> 24; // 8bit lookup table address + y0 = AudioWaveformSine[idx]; + y1 = AudioWaveformSine[idx+1]; + idx = lfo2_phase_acc & 0x00FFFFFF; // lower 24 bit = fractional part + y = (int64_t)y0 * (0x00FFFFFF - idx); + y += (int64_t)y1 * idx; + lfo2_out_sin = (int32_t) (y >> (32-8)); //32-8->output 16bit, + idx = ((lfo2_phase_acc >> 24)+64) & 0xFF; + y0 = AudioWaveformSine[idx]; + y1 = AudioWaveformSine[idx + 1]; + y = (int64_t)y0 * (0x00FFFFFF - idx); + y += (int64_t)y1 * idx; + lfo2_out_cos = (int32_t) (y >> (32-8)); // 16bit output + + input = input_blockL[i] * input_attn; + // chained input allpasses, channel L + acc = in_allp1_bufL[in_allp1_idxL] + input * in_allp_k; + in_allp1_bufL[in_allp1_idxL] = input - in_allp_k * acc; + input = acc; + if (++in_allp1_idxL >= sizeof(in_allp1_bufL)/sizeof(float32_t)) in_allp1_idxL = 0; + + acc = in_allp2_bufL[in_allp2_idxL] + input * in_allp_k; + in_allp2_bufL[in_allp2_idxL] = input - in_allp_k * acc; + input = acc; + if (++in_allp2_idxL >= sizeof(in_allp2_bufL)/sizeof(float32_t)) in_allp2_idxL = 0; + + acc = in_allp3_bufL[in_allp3_idxL] + input * in_allp_k; + in_allp3_bufL[in_allp3_idxL] = input - in_allp_k * acc; + input = acc; + if (++in_allp3_idxL >= sizeof(in_allp3_bufL)/sizeof(float32_t)) in_allp3_idxL = 0; + + acc = in_allp4_bufL[in_allp4_idxL] + input * in_allp_k; + in_allp4_bufL[in_allp4_idxL] = input - in_allp_k * acc; + in_allp_out_L = acc; + if (++in_allp4_idxL >= sizeof(in_allp4_bufL)/sizeof(float32_t)) in_allp4_idxL = 0; + + input = input_blockR[i] * input_attn; + + // chained input allpasses, channel R + acc = in_allp1_bufR[in_allp1_idxR] + input * in_allp_k; + in_allp1_bufR[in_allp1_idxR] = input - in_allp_k * acc; + input = acc; + if (++in_allp1_idxR >= sizeof(in_allp1_bufR)/sizeof(float32_t)) in_allp1_idxR = 0; + + acc = in_allp2_bufR[in_allp2_idxR] + input * in_allp_k; + in_allp2_bufR[in_allp2_idxR] = input - in_allp_k * acc; + input = acc; + if (++in_allp2_idxR >= sizeof(in_allp2_bufR)/sizeof(float32_t)) in_allp2_idxR = 0; + + acc = in_allp3_bufR[in_allp3_idxR] + input * in_allp_k; + in_allp3_bufR[in_allp3_idxR] = input - in_allp_k * acc; + input = acc; + if (++in_allp3_idxR >= sizeof(in_allp3_bufR)/sizeof(float32_t)) in_allp3_idxR = 0; + + acc = in_allp4_bufR[in_allp4_idxR] + input * in_allp_k; + in_allp4_bufR[in_allp4_idxR] = input - in_allp_k * acc; + in_allp_out_R = acc; + if (++in_allp4_idxR >= sizeof(in_allp4_bufR)/sizeof(float32_t)) in_allp4_idxR = 0; + + // input allpases done, start loop allpases + input = lp_allp_out + in_allp_out_R; + acc = lp_allp1_buf[lp_allp1_idx] + input * loop_allp_k; // input is the lp allpass chain output + lp_allp1_buf[lp_allp1_idx] = input - loop_allp_k * acc; + input = acc; + if (++lp_allp1_idx >= sizeof(lp_allp1_buf)/sizeof(float32_t)) lp_allp1_idx = 0; + + acc = lp_dly1_buf[lp_dly1_idx]; // read the end of the delay + lp_dly1_buf[lp_dly1_idx] = input; // write new sample + input = acc; + if (++lp_dly1_idx >= sizeof(lp_dly1_buf)/sizeof(float32_t)) lp_dly1_idx = 0; // update index + + // hi/lo shelving filter + temp1 = input - lpf1; + lpf1 += temp1 * lp_lowpass_f; + temp2 = input - lpf1; + temp1 = lpf1 - hpf1; + hpf1 += temp1 * lp_hipass_f; + acc = lpf1 + temp2*lp_hidamp_k + hpf1*lp_lodamp_k; + acc = acc * rv_time * rv_time_scaler; // scale by the reveb time + + input = acc + in_allp_out_L; + + acc = lp_allp2_buf[lp_allp2_idx] + input * loop_allp_k; + lp_allp2_buf[lp_allp2_idx] = input - loop_allp_k * acc; + input = acc; + if (++lp_allp2_idx >= sizeof(lp_allp2_buf)/sizeof(float32_t)) lp_allp2_idx = 0; + acc = lp_dly2_buf[lp_dly2_idx]; // read the end of the delay + lp_dly2_buf[lp_dly2_idx] = input; // write new sample + input = acc; + if (++lp_dly2_idx >= sizeof(lp_dly2_buf)/sizeof(float32_t)) lp_dly2_idx = 0; // update index + // hi/lo shelving filter + temp1 = input - lpf2; + lpf2 += temp1 * lp_lowpass_f; + temp2 = input - lpf2; + temp1 = lpf2 - hpf2; + hpf2 += temp1 * lp_hipass_f; + acc = lpf2 + temp2*lp_hidamp_k + hpf2*lp_lodamp_k; + acc = acc * rv_time * rv_time_scaler; + + input = acc + in_allp_out_R; + + acc = lp_allp3_buf[lp_allp3_idx] + input * loop_allp_k; + lp_allp3_buf[lp_allp3_idx] = input - loop_allp_k * acc; + input = acc; + if (++lp_allp3_idx >= sizeof(lp_allp3_buf)/sizeof(float32_t)) lp_allp3_idx = 0; + acc = lp_dly3_buf[lp_dly3_idx]; // read the end of the delay + lp_dly3_buf[lp_dly3_idx] = input; // write new sample + input = acc; + if (++lp_dly3_idx >= sizeof(lp_dly3_buf)/sizeof(float32_t)) lp_dly3_idx = 0; // update index + // hi/lo shelving filter + temp1 = input - lpf3; + lpf3 += temp1 * lp_lowpass_f; + temp2 = input - lpf3; + temp1 = lpf3 - hpf3; + hpf3 += temp1 * lp_hipass_f; + acc = lpf3 + temp2*lp_hidamp_k + hpf3*lp_lodamp_k; + acc = acc * rv_time * rv_time_scaler; + + input = acc + in_allp_out_L; + + acc = lp_allp4_buf[lp_allp4_idx] + input * loop_allp_k; + lp_allp4_buf[lp_allp4_idx] = input - loop_allp_k * acc; + input = acc; + if (++lp_allp4_idx >= sizeof(lp_allp4_buf)/sizeof(float32_t)) lp_allp4_idx = 0; + acc = lp_dly4_buf[lp_dly4_idx]; // read the end of the delay + lp_dly4_buf[lp_dly4_idx] = input; // write new sample + input = acc; + if (++lp_dly4_idx >= sizeof(lp_dly4_buf)/sizeof(float32_t)) lp_dly4_idx= 0; // update index + // hi/lo shelving filter + temp1 = input - lpf4; + lpf4 += temp1 * lp_lowpass_f; + temp2 = input - lpf4; + temp1 = lpf4 - hpf4; + hpf4 += temp1 * lp_hipass_f; + acc = lpf4 + temp2*lp_hidamp_k + hpf4*lp_lodamp_k; + 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*(1.0f-input) + temp2*input)* 0.8f; +#else + temp16 = (lp_dly1_idx + lp_dly1_offset_L) % (sizeof(lp_dly1_buf)/sizeof(float32_t)); + acc = lp_dly1_buf[temp16]* 0.8f; +#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*(1.0f-input) + temp2*input)* 0.7f; +#else + temp16 = (lp_dly2_idx + lp_dly2_offset_L) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); + acc += (temp1*(1.0f-input) + temp2*input)* 0.6f; +#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*(1.0f-input) + temp2*input)* 0.6f; + + 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*(1.0f-input) + temp2*input)* 0.5f; + + // Master lowpass filter + temp1 = acc - master_lowpass_l; + master_lowpass_l += temp1 * master_lowpass_f; + + outblockL[i] =(int16_t)(master_lowpass_l * 32767.0f); //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*(1.0f-input) + temp2*input)* 0.8f; + #else + temp16 = (lp_dly1_idx + lp_dly1_offset_R) % (sizeof(lp_dly1_buf)/sizeof(float32_t)); + acc = lp_dly1_buf[temp16] * 0.8f; + #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*(1.0f-input) + temp2*input)* 0.7f; +#else + temp16 = (lp_dly2_idx + lp_dly2_offset_R) % (sizeof(lp_dly2_buf)/sizeof(float32_t)); + acc += (temp1*(1.0f-input) + temp2*input)* 0.7f; +#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*(1.0f-input) + temp2*input)* 0.6f; + + 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*(1.0f-input) + temp2*input)* 0.5f; + + // Master lowpass filter + temp1 = acc - master_lowpass_r; + master_lowpass_r += temp1 * master_lowpass_f; + outblockR[i] =(int16_t)(master_lowpass_r * 32767.0f); + + } + + if(input_blockL) + free(input_blockL); + if(input_blockR) + free(input_blockR); +} diff --git a/src/effect_platervbstereo.h b/src/effect_platervbstereo.h new file mode 100644 index 0000000..21f751b --- /dev/null +++ b/src/effect_platervbstereo.h @@ -0,0 +1,222 @@ +/* Stereo plate reverb for Teensy 4 + * + * Adapted for use in MiniDexed (Holger Wirtz ) + * + * 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 + * + */ + +#pragma once +#ifndef _EFFECT_PLATERVBSTEREO_H +#define _EFFECT_PLATERVBSTEREO_H + +#include "arm_math.h" +#include + +#define constrain(amt, low, high) ({ \ + __typeof__(amt) _amt = (amt); \ + __typeof__(low) _low = (low); \ + __typeof__(high) _high = (high); \ + (_amt < _low) ? _low : ((_amt > _high) ? _high : _amt); \ +}) + +/* +template +inline static T min(const T& a, const T& b) { + return a < b ? a : b; +} + +template +inline static T max(const T& a, const T& b) { + return a > b ? a : b; +} +*/ + +inline long my_map(long x, long in_min, long in_max, long out_min, long out_max) { + return (x - in_min) * (out_max - out_min) / (in_max - in_min) + out_min; +} + +/*** + * 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: + AudioEffectPlateReverb(float32_t samplerate); + void doReverb(uint16_t len, int16_t *inblockL, int16_t *inblockR, int16_t *outblockL, int16_t *outblockR); + + void size(float n) + { + n = constrain(n, 0.0f, 1.0f); + n = my_map(n, 0.0f, 1.0f, 0.2f, rv_time_k_max); + float32_t attn = my_map(n, 0.0f, rv_time_k_max, 0.5f, 0.25f); + //__disable_irq(); + rv_time_k = n; + input_attn = attn; + //__enable_irq(); + } + + void hidamp(float n) + { + n = constrain(n, 0.0f, 1.0f); + //__disable_irq(); + lp_hidamp_k = 1.0f - n; + //__enable_irq(); + } + + void lodamp(float n) + { + n = constrain(n, 0.0f, 1.0f); + //__disable_irq(); + lp_lodamp_k = -n; + rv_time_scaler = 1.0f - n * 0.12f; // limit the max reverb time, otherwise it will clip + //__enable_irq(); + } + + void lowpass(float n) + { + n = constrain(n, 0.0f, 1.0f); + n = my_map(n*n*n, 0.0f, 1.0f, 0.05f, 1.0f); + master_lowpass_f = n; + } + + void diffusion(float n) + { + n = constrain(n, 0.0f, 1.0f); + n = my_map(n, 0.0f, 1.0f, 0.005f, 0.65f); + //__disable_irq(); + in_allp_k = n; + loop_allp_k = n; + //__enable_irq(); + } + + float32_t get_size(void) {return rv_time_k;} + bool get_bypass(void) {return bypass;} + void set_bypass(bool state) {bypass = state;}; + void tgl_bypass(void) {bypass ^=1;} +private: + bool bypass = false; + float32_t* input_blockL; + float32_t* input_blockR; + float32_t input_attn; + + float32_t in_allp_k; // input allpass coeff + 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]; + 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 + 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]; + 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 + float32_t lp_allp1_buf[2303]; // loop allpass buffers + float32_t lp_allp2_buf[2905]; + float32_t lp_allp3_buf[3175]; + float32_t lp_allp4_buf[2398]; + 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 + float32_t lp_allp_out; + float32_t lp_dly1_buf[3423]; + float32_t lp_dly2_buf[4589]; + float32_t lp_dly3_buf[4365]; + float32_t lp_dly4_buf[3698]; + 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; // delay line tap offets + 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 = 0.95f; + 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 diff --git a/src/minidexed.cpp b/src/minidexed.cpp index 4279444..64285b1 100644 --- a/src/minidexed.cpp +++ b/src/minidexed.cpp @@ -108,6 +108,14 @@ CMiniDexed::CMiniDexed (CConfig *pConfig, CInterruptSystem *pInterrupt, m_CoreStatus[nCore] = CoreStatusInit; } #endif + + // Create reverb object + reverb = new AudioEffectPlateReverb(pConfig->GetSampleRate()); + reverb->size(0.3); + reverb->hidamp(0.8); + reverb->lodamp(0.5); + reverb->lowpass(0.3); + reverb->diffusion(0.2); }; bool CMiniDexed::Initialize (void) @@ -584,6 +592,15 @@ void CMiniDexed::ProcessSound (void) } } + // Test adding reverb + int16_t ReverbBuffer[nFrames][2]; + reverb->doReverb(nFrames,&SampleBuffer[0][0],&SampleBuffer[0][1],&ReverbBuffer[0][0],&ReverbBuffer[0][1]); + for (unsigned i = 0; i < nFrames; i++) + { + SampleBuffer[i][0] += ReverbBuffer[0][0]; + SampleBuffer[i][1] += ReverbBuffer[0][1]; + } + if ( m_pSoundDevice->Write (SampleBuffer, sizeof SampleBuffer) != (int) sizeof SampleBuffer) { diff --git a/src/minidexed.h b/src/minidexed.h index 7632544..2e9efc8 100644 --- a/src/minidexed.h +++ b/src/minidexed.h @@ -38,6 +38,7 @@ #include #include #include +#include "effect_platervbstereo.h" class CMiniDexed #ifdef ARM_ALLOW_MULTI_CORE @@ -124,6 +125,8 @@ private: CPerformanceTimer m_GetChunkTimer; bool m_bProfileEnabled; + + AudioEffectPlateReverb* reverb; }; #endif