mirror of https://github.com/probonopd/MiniDexed
Added code for enabling compressor (compressor code is in Synth_Dexed) / Fixes for README.md / Added stereo plate reverb (#59)
* Additions for compressor. * Added code for enabling compressor of Synth_Dexed (with default values). * First try for adding a reverb effect. * Fixes for reverb. * Added full implementation of plate reverb. * -o KERNEL_MAX_SIZE=0x400000 https://github.com/probonopd/MiniDexed/pull/59#issuecomment-1086713102 Co-authored-by: Holger Wirtz <wirtz@parasitstudio.de> Co-authored-by: probonopd <probonopd@users.noreply.github.com>pull/61/head
Holger
3 years ago
committed by
GitHub
parent
99cefaefd3
commit
3b51ed8477
@ -1 +1 @@ |
|||||||
Subproject commit 5bceb75ce5b8a7620cdd469ca7bacb7d0afcd5e1 |
Subproject commit 70293ae5998643c706244b090504dde8b4097851 |
||||||
@ -0,0 +1,467 @@ |
|||||||
|
/* Stereo plate reverb for Teensy 4
|
||||||
|
* |
||||||
|
* Adapted for MiniDexed (Holger Wirtz <dcoredump@googlemail.com>) |
||||||
|
* |
||||||
|
* 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 <stdio.h> |
||||||
|
#include <cstdlib> |
||||||
|
#include <assert.h> |
||||||
|
#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<<LFO_AMPL_BITS) + 1) // lfo amplitude
|
||||||
|
#define LFO_READ_OFFSET (LFO_AMPL>>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<<LFO_FRAC_BITS)-1) // mask for the above
|
||||||
|
|
||||||
|
#define LFO1_FREQ_HZ (1.37f) // LFO1 frequency in Hz
|
||||||
|
#define LFO2_FREQ_HZ (1.52f) // LFO2 frequency in Hz
|
||||||
|
|
||||||
|
#define RV_MASTER_LOWPASS_F (0.6f) // master lowpass scaled frequency coeff.
|
||||||
|
|
||||||
|
const int16_t AudioWaveformSine[257] = { |
||||||
|
0, 804, 1608, 2410, 3212, 4011, 4808, 5602, 6393, 7179, |
||||||
|
7962, 8739, 9512, 10278, 11039, 11793, 12539, 13279, 14010, 14732, |
||||||
|
15446, 16151, 16846, 17530, 18204, 18868, 19519, 20159, 20787, 21403, |
||||||
|
22005, 22594, 23170, 23731, 24279, 24811, 25329, 25832, 26319, 26790, |
||||||
|
27245, 27683, 28105, 28510, 28898, 29268, 29621, 29956, 30273, 30571, |
||||||
|
30852, 31113, 31356, 31580, 31785, 31971, 32137, 32285, 32412, 32521, |
||||||
|
32609, 32678, 32728, 32757, 32767, 32757, 32728, 32678, 32609, 32521, |
||||||
|
32412, 32285, 32137, 31971, 31785, 31580, 31356, 31113, 30852, 30571, |
||||||
|
30273, 29956, 29621, 29268, 28898, 28510, 28105, 27683, 27245, 26790, |
||||||
|
26319, 25832, 25329, 24811, 24279, 23731, 23170, 22594, 22005, 21403, |
||||||
|
20787, 20159, 19519, 18868, 18204, 17530, 16846, 16151, 15446, 14732, |
||||||
|
14010, 13279, 12539, 11793, 11039, 10278, 9512, 8739, 7962, 7179, |
||||||
|
6393, 5602, 4808, 4011, 3212, 2410, 1608, 804, 0, -804, |
||||||
|
-1608, -2410, -3212, -4011, -4808, -5602, -6393, -7179, -7962, -8739, |
||||||
|
-9512,-10278,-11039,-11793,-12539,-13279,-14010,-14732,-15446,-16151, |
||||||
|
-16846,-17530,-18204,-18868,-19519,-20159,-20787,-21403,-22005,-22594, |
||||||
|
-23170,-23731,-24279,-24811,-25329,-25832,-26319,-26790,-27245,-27683, |
||||||
|
-28105,-28510,-28898,-29268,-29621,-29956,-30273,-30571,-30852,-31113, |
||||||
|
-31356,-31580,-31785,-31971,-32137,-32285,-32412,-32521,-32609,-32678, |
||||||
|
-32728,-32757,-32767,-32757,-32728,-32678,-32609,-32521,-32412,-32285, |
||||||
|
-32137,-31971,-31785,-31580,-31356,-31113,-30852,-30571,-30273,-29956, |
||||||
|
-29621,-29268,-28898,-28510,-28105,-27683,-27245,-26790,-26319,-25832, |
||||||
|
-25329,-24811,-24279,-23731,-23170,-22594,-22005,-21403,-20787,-20159, |
||||||
|
-19519,-18868,-18204,-17530,-16846,-16151,-15446,-14732,-14010,-13279, |
||||||
|
-12539,-11793,-11039,-10278, -9512, -8739, -7962, -7179, -6393, -5602, |
||||||
|
-4808, -4011, -3212, -2410, -1608, -804, 0 |
||||||
|
}; |
||||||
|
|
||||||
|
AudioEffectPlateReverb::AudioEffectPlateReverb(float32_t samplerate) |
||||||
|
{ |
||||||
|
input_attn = 0.5f; |
||||||
|
in_allp_k = INP_ALLP_COEFF; |
||||||
|
|
||||||
|
memset(in_allp1_bufL, 0, sizeof(in_allp1_bufL)); |
||||||
|
memset(in_allp2_bufL, 0, sizeof(in_allp2_bufL)); |
||||||
|
memset(in_allp3_bufL, 0, sizeof(in_allp3_bufL)); |
||||||
|
memset(in_allp4_bufL, 0, sizeof(in_allp4_bufL)); |
||||||
|
in_allp1_idxL = 0; |
||||||
|
in_allp2_idxL = 0; |
||||||
|
in_allp3_idxL = 0; |
||||||
|
in_allp4_idxL = 0; |
||||||
|
|
||||||
|
memset(in_allp1_bufR, 0, sizeof(in_allp1_bufR)); |
||||||
|
memset(in_allp2_bufR, 0, sizeof(in_allp2_bufR)); |
||||||
|
memset(in_allp3_bufR, 0, sizeof(in_allp3_bufR)); |
||||||
|
memset(in_allp4_bufR, 0, sizeof(in_allp4_bufR)); |
||||||
|
in_allp1_idxR = 0; |
||||||
|
in_allp2_idxR = 0; |
||||||
|
in_allp3_idxR = 0; |
||||||
|
in_allp4_idxR = 0; |
||||||
|
|
||||||
|
in_allp_out_R = 0.0f; |
||||||
|
|
||||||
|
memset(lp_allp1_buf, 0, sizeof(lp_allp1_buf)); |
||||||
|
memset(lp_allp2_buf, 0, sizeof(lp_allp2_buf)); |
||||||
|
memset(lp_allp3_buf, 0, sizeof(lp_allp3_buf)); |
||||||
|
memset(lp_allp4_buf, 0, sizeof(lp_allp4_buf)); |
||||||
|
lp_allp1_idx = 0; |
||||||
|
lp_allp2_idx = 0; |
||||||
|
lp_allp3_idx = 0; |
||||||
|
lp_allp4_idx = 0; |
||||||
|
loop_allp_k = LOOP_ALLOP_COEFF; |
||||||
|
lp_allp_out = 0.0f; |
||||||
|
|
||||||
|
memset(lp_dly1_buf, 0, sizeof(lp_dly1_buf)); |
||||||
|
memset(lp_dly2_buf, 0, sizeof(lp_dly2_buf)); |
||||||
|
memset(lp_dly3_buf, 0, sizeof(lp_dly3_buf)); |
||||||
|
memset(lp_dly4_buf, 0, sizeof(lp_dly4_buf)); |
||||||
|
lp_dly1_idx = 0; |
||||||
|
lp_dly2_idx = 0; |
||||||
|
lp_dly3_idx = 0; |
||||||
|
lp_dly4_idx = 0; |
||||||
|
|
||||||
|
lp_hidamp_k = 1.0f; |
||||||
|
lp_lodamp_k = 0.0f; |
||||||
|
|
||||||
|
lp_lowpass_f = HI_LOSS_FREQ; |
||||||
|
lp_hipass_f = LO_LOSS_FREQ; |
||||||
|
|
||||||
|
lpf1 = 0.0f; |
||||||
|
lpf2 = 0.0f; |
||||||
|
lpf3 = 0.0f; |
||||||
|
lpf4 = 0.0f; |
||||||
|
|
||||||
|
hpf1 = 0.0f; |
||||||
|
hpf2 = 0.0f; |
||||||
|
hpf3 = 0.0f; |
||||||
|
hpf4 = 0.0f; |
||||||
|
|
||||||
|
master_lowpass_f = RV_MASTER_LOWPASS_F; |
||||||
|
master_lowpass_l = 0.0f; |
||||||
|
master_lowpass_r = 0.0f; |
||||||
|
|
||||||
|
lfo1_phase_acc = 0; |
||||||
|
lfo1_adder = (UINT32_MAX + 1)/(samplerate * LFO1_FREQ_HZ); |
||||||
|
lfo2_phase_acc = 0; |
||||||
|
lfo2_adder = (UINT32_MAX + 1)/(samplerate * LFO2_FREQ_HZ);
|
||||||
|
|
||||||
|
send_level = 0.0; |
||||||
|
} |
||||||
|
|
||||||
|
// #define sat16(n, rshift) signed_saturate_rshift((n), 16, (rshift))
|
||||||
|
|
||||||
|
void AudioEffectPlateReverb::doReverb(uint16_t len, int16_t audioblock[][2]) |
||||||
|
{ |
||||||
|
int i; |
||||||
|
float32_t input, acc, temp1, temp2; |
||||||
|
uint16_t temp16; |
||||||
|
float32_t rv_time; |
||||||
|
|
||||||
|
// for LFOs:
|
||||||
|
int16_t lfo1_out_sin, lfo1_out_cos, lfo2_out_sin, lfo2_out_cos; |
||||||
|
int32_t y0, y1; |
||||||
|
int64_t y; |
||||||
|
uint32_t idx; |
||||||
|
static bool cleanup_done = false; |
||||||
|
|
||||||
|
// handle bypass, 1st call will clean the buffers to avoid continuing the previous reverb tail
|
||||||
|
if (bypass) |
||||||
|
{ |
||||||
|
if (!cleanup_done) |
||||||
|
{ |
||||||
|
memset(in_allp1_bufL, 0, sizeof(in_allp1_bufL)); |
||||||
|
memset(in_allp2_bufL, 0, sizeof(in_allp2_bufL)); |
||||||
|
memset(in_allp3_bufL, 0, sizeof(in_allp3_bufL)); |
||||||
|
memset(in_allp4_bufL, 0, sizeof(in_allp4_bufL)); |
||||||
|
memset(in_allp1_bufR, 0, sizeof(in_allp1_bufR)); |
||||||
|
memset(in_allp2_bufR, 0, sizeof(in_allp2_bufR)); |
||||||
|
memset(in_allp3_bufR, 0, sizeof(in_allp3_bufR)); |
||||||
|
memset(in_allp4_bufR, 0, sizeof(in_allp4_bufR)); |
||||||
|
memset(lp_allp1_buf, 0, sizeof(lp_allp1_buf)); |
||||||
|
memset(lp_allp2_buf, 0, sizeof(lp_allp2_buf)); |
||||||
|
memset(lp_allp3_buf, 0, sizeof(lp_allp3_buf)); |
||||||
|
memset(lp_allp4_buf, 0, sizeof(lp_allp4_buf)); |
||||||
|
memset(lp_dly1_buf, 0, sizeof(lp_dly1_buf)); |
||||||
|
memset(lp_dly2_buf, 0, sizeof(lp_dly2_buf)); |
||||||
|
memset(lp_dly3_buf, 0, sizeof(lp_dly3_buf)); |
||||||
|
memset(lp_dly4_buf, 0, sizeof(lp_dly4_buf)); |
||||||
|
|
||||||
|
cleanup_done = true; |
||||||
|
} |
||||||
|
|
||||||
|
return; |
||||||
|
} |
||||||
|
cleanup_done = false; |
||||||
|
|
||||||
|
rv_time = rv_time_k; |
||||||
|
|
||||||
|
for (i=0; i < len; i++)
|
||||||
|
{ |
||||||
|
// do the LFOs
|
||||||
|
lfo1_phase_acc += lfo1_adder; |
||||||
|
idx = lfo1_phase_acc >> 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 = (float32_t(audioblock[i][0])/32767.0f) * 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 = (float32_t(audioblock[i][1])/32767.0f) * 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; |
||||||
|
|
||||||
|
int32_t out = audioblock[i][0] + int16_t(master_lowpass_l * 32767.0f * send_level); |
||||||
|
if(out > INT16_MAX) |
||||||
|
audioblock[i][0] = INT16_MAX; |
||||||
|
else if(out < INT16_MIN) |
||||||
|
audioblock[i][0] = INT16_MIN; |
||||||
|
else |
||||||
|
audioblock[i][0] = out; |
||||||
|
|
||||||
|
// 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; |
||||||
|
|
||||||
|
out = audioblock[i][1] + int16_t(master_lowpass_l * 32767.0f * send_level); |
||||||
|
if(out > INT16_MAX) |
||||||
|
audioblock[i][1] = INT16_MAX; |
||||||
|
else if(out < INT16_MIN) |
||||||
|
audioblock[i][1] = INT16_MIN; |
||||||
|
else |
||||||
|
audioblock[i][1] = out; |
||||||
|
} |
||||||
|
} |
||||||
@ -0,0 +1,232 @@ |
|||||||
|
/* Stereo plate reverb for Teensy 4
|
||||||
|
* |
||||||
|
* Adapted for use in MiniDexed (Holger Wirtz <wirtz@parasitstudio.de>) |
||||||
|
* |
||||||
|
* 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 <stdint.h> |
||||||
|
|
||||||
|
#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<typename T> |
||||||
|
inline static T min(const T& a, const T& b) { |
||||||
|
return a < b ? a : b; |
||||||
|
} |
||||||
|
|
||||||
|
template<typename T> |
||||||
|
inline static T max(const T& a, const T& b) { |
||||||
|
return a > b ? a : b; |
||||||
|
} |
||||||
|
|
||||||
|
inline long maplong(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; |
||||||
|
} |
||||||
|
*/ |
||||||
|
|
||||||
|
inline float32_t mapfloat(float32_t val, float32_t in_min, float32_t in_max, float32_t out_min, float32_t out_max) |
||||||
|
{ |
||||||
|
return (val - 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 audioblock[][2]); |
||||||
|
|
||||||
|
void size(float n) |
||||||
|
{ |
||||||
|
n = constrain(n, 0.0f, 1.0f); |
||||||
|
n = mapfloat(n, 0.0f, 1.0f, 0.2f, rv_time_k_max); |
||||||
|
float32_t attn = mapfloat(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 = mapfloat(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 = mapfloat(n, 0.0f, 1.0f, 0.005f, 0.65f); |
||||||
|
//__disable_irq();
|
||||||
|
in_allp_k = n; |
||||||
|
loop_allp_k = n; |
||||||
|
//__enable_irq();
|
||||||
|
} |
||||||
|
|
||||||
|
void send(float n) |
||||||
|
{ |
||||||
|
send_level = constrain(n, 0.0f, 1.0f); |
||||||
|
} |
||||||
|
|
||||||
|
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 send_level; |
||||||
|
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
|
||||||
Loading…
Reference in new issue