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a-Reverb: Using new algorithm based on FreeVerb

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This commit is contained in:
Damien Zammit 2016-07-13 18:07:02 +10:00
parent 065d9434b9
commit 5965fedc51
2 changed files with 224 additions and 122 deletions
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293
libs/plugins/a-reverb.lv2/a-reverb.c
View File

@ -1,8 +1,9 @@
/* a-reverb -- based on b_reverb (setBfree)
/* a-reverb -- based on b_reverb (setBfree) and FreeVerb
*
* Copyright (C) 2003-2004 Fredrik Kilander <fk@dsv.su.se>
* Copyright (C) 2008-2016 Robin Gareus <robin@gareus.org>
* Copyright (C) 2012 Will Panther <pantherb@setbfree.org>
* Copyright (C) 2016 Damien Zammit <damien@zamaudio.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
@ -24,21 +25,24 @@
#include <math.h>
#include <string.h>
#define RV_NZ 7
#define RV_NZ (8+4)
#define DENORMAL_PROTECT (1e-14)
typedef struct {
float* delays[RV_NZ]; /**< delay line buffer */
float* idx0[RV_NZ]; /**< Reset pointer ref delays[]*/
float* idxp[RV_NZ]; /**< Index pointer ref delays[]*/
float* endp[RV_NZ]; /**< End pointer ref delays[]*/
typedef struct {
float* delays[2][RV_NZ]; /**< delay line buffer */
float* idx0[2][RV_NZ]; /**< Reset pointer ref delays[]*/
float* idxp[2][RV_NZ]; /**< Index pointer ref delays[]*/
float* endp[2][RV_NZ]; /**< End pointer ref delays[]*/
float gain[RV_NZ]; /**< feedback gains */
float yy1; /**< Previous output sample */
float y_1; /**< Feedback sample */
float yy1_0; /**< Previous output sample */
float y_1_0; /**< Feedback sample */
float yy1_1; /**< Previous output sample */
float y_1_1; /**< Feedback sample */
int end[RV_NZ];
int end[2][RV_NZ];
float inputGain; /**< Input gain value */
float fbk; /**< Feedback gain */
@ -47,18 +51,18 @@ typedef struct {
} b_reverb;
static int
setReverbPointers (b_reverb *r, int i, const double rate)
setReverbPointers (b_reverb *r, int i, int c, const double rate)
{
int e = (r->end[i] * rate / 25000.0);
int e = (r->end[c][i] * rate / 44100.0);
e = e | 1;
r->delays[i] = (float*)realloc ((void*)r->delays[i], (e + 2) * sizeof (float));
if (!r->delays[i]) {
r->delays[c][i] = (float*)realloc ((void*)r->delays[c][i], (e + 2) * sizeof (float));
if (!r->delays[c][i]) {
return -1;
} else {
memset (r->delays[i], 0 , (e + 2) * sizeof (float));
memset (r->delays[c][i], 0 , (e + 2) * sizeof (float));
}
r->endp[i] = r->delays[i] + e + 1;
r->idx0[i] = r->idxp[i] = &(r->delays[i][0]);
r->endp[c][i] = r->delays[c][i] + e + 1;
r->idx0[c][i] = r->idxp[c][i] = &(r->delays[c][i][0]);
return 0;
}
@ -67,103 +71,171 @@ static int
initReverb (b_reverb *r, const double rate)
{
int err = 0;
r->inputGain = 0.1; /* Input gain value */
int stereowidth = 7;
r->inputGain = powf (10.0, .05 * -20.0); // -20dB
r->fbk = -0.015; /* Feedback gain */
r->wet = 0.1; /* Output dry gain */
r->dry = 0.9; /* Output wet gain */
r->wet = 0.3;
r->dry = 0.7;
/* feedback combfilter */
r->gain[0] = 0.773;
r->gain[1] = 0.802;
r->gain[2] = 0.753;
r->gain[3] = 0.733;
r->gain[0] = 0.75;
r->gain[1] = 0.75;
r->gain[2] = 0.75;
r->gain[3] = 0.75;
r->gain[4] = 0.75;
r->gain[5] = 0.75;
r->gain[6] = 0.75;
r->gain[7] = 0.75;
/* all-pass filter */
r->gain[4] = sqrtf (0.5);
r->gain[5] = sqrtf (0.5);
r->gain[6] = sqrtf (0.5);
r->gain[8] = 0.5;
r->gain[9] = 0.5;
r->gain[10] = 0.5;
r->gain[11] = 0.5;
/* delay lines */
r->end[0] = 1687;
r->end[1] = 1601;
r->end[2] = 2053;
r->end[3] = 2251;
/* delay lines left */
r->end[0][0] = 1116;
r->end[0][1] = 1188;
r->end[0][2] = 1277;
r->end[0][3] = 1356;
r->end[0][4] = 1422;
r->end[0][5] = 1491;
r->end[0][6] = 1557;
r->end[0][7] = 1617;
/* all pass filters left */
r->end[0][8] = 556;
r->end[0][9] = 441;
r->end[0][10] = 341;
r->end[0][11] = 225;
/* delay lines right */
r->end[1][0] = 1116 + stereowidth;
r->end[1][1] = 1188 + stereowidth;
r->end[1][2] = 1277 + stereowidth;
r->end[1][3] = 1356 + stereowidth;
r->end[1][4] = 1422 + stereowidth;
r->end[1][5] = 1491 + stereowidth;
r->end[1][6] = 1557 + stereowidth;
r->end[1][7] = 1617 + stereowidth;
/* all pass filters */
r->end[4] = 347;
r->end[5] = 113;
r->end[6] = 37;
r->end[1][8] = 556 + stereowidth;
r->end[1][9] = 441 + stereowidth;
r->end[1][10] = 341 + stereowidth;
r->end[1][11] = 225 + stereowidth;
for (int i = 0; i < RV_NZ; ++i) {
r->delays[i]= NULL;
r->delays[0][i] = NULL;
r->delays[1][i] = NULL;
}
r->yy1 = 0.0;
r->y_1 = 0.0;
r->yy1_0 = 0.0;
r->y_1_0 = 0.0;
r->yy1_1 = 0.0;
r->y_1_1 = 0.0;
for (int i = 0; i < RV_NZ; i++) {
err |= setReverbPointers (r, i, rate);
err |= setReverbPointers (r, i, 0, rate);
err |= setReverbPointers (r, i, 1, rate);
}
return err;
}
static void
reverb (b_reverb* r,
const float* inbuf,
float* outbuf,
const float* inbuf0,
const float* inbuf1,
float* outbuf0,
float* outbuf1,
size_t n_samples)
{
float** const idxp = r->idxp;
float* const* const endp = r->endp;
float* const* const idx0 = r->idx0;
float** const idxp0 = r->idxp[0];
float** const idxp1 = r->idxp[1];
float* const* const endp0 = r->endp[0];
float* const* const endp1 = r->endp[1];
float* const* const idx00 = r->idx0[0];
float* const* const idx01 = r->idx0[1];
const float* const gain = r->gain;
const float inputGain = r->inputGain;
const float fbk = r->fbk;
const float wet = r->wet;
const float dry = r->dry;
const float* xp = inbuf;
float* yp = outbuf;
const float* xp0 = inbuf0;
const float* xp1 = inbuf1;
float* yp0 = outbuf0;
float* yp1 = outbuf1;
float y_1 = r->y_1;
float yy1 = r->yy1;
float y_1_0 = r->y_1_0;
float yy1_0 = r->yy1_0;
float y_1_1 = r->y_1_1;
float yy1_1 = r->yy1_1;
for (size_t i = 0; i < n_samples; ++i) {
int j;
float y;
const float xo = *xp++;
const float x = y_1 + (inputGain * xo);
const float xo0 = *xp0++;
const float xo1 = *xp1++;
const float x0 = y_1_0 + (inputGain * xo0);
const float x1 = y_1_1 + (inputGain * xo1);
float xa = 0.0;
float xb = 0.0;
/* First we do four feedback comb filters (ie parallel delay lines,
* each with a single tap at the end that feeds back at the start) */
for (j = 0; j < 4; ++j) {
y = *idxp[j];
*idxp[j] = x + (gain[j] * y);
if (endp[j] <= ++(idxp[j])) {
idxp[j] = idx0[j];
for (j = 0; j < 8; ++j) {
y = *idxp0[j];
*idxp0[j] = x0 + (gain[j] * y);
if (endp0[j] <= ++(idxp0[j])) {
idxp0[j] = idx00[j];
}
xa += y;
}
for (; j < 7; ++j) {
y = *idxp[j];
*idxp[j] = gain[j] * (xa + y);
if (endp[j] <= ++(idxp[j])) {
idxp[j] = idx0[j];
for (; j < 12; ++j) {
y = *idxp0[j];
*idxp0[j] = gain[j] * (xa + y);
if (endp0[j] <= ++(idxp0[j])) {
idxp0[j] = idx00[j];
}
xa = y - xa;
}
y = 0.5f * (xa + yy1);
yy1 = y;
y_1 = fbk * xa;
y = 0.5f * (xa + yy1_0);
yy1_0 = y;
y_1_0 = fbk * xa;
*yp++ = ((wet * y) + (dry * xo));
*yp0++ = ((wet * y) + (dry * xo0));
for (j = 0; j < 8; ++j) {
y = *idxp1[j];
*idxp1[j] = x1 + (gain[j] * y);
if (endp1[j] <= ++(idxp1[j])) {
idxp1[j] = idx01[j];
}
xb += y;
}
for (; j < 12; ++j) {
y = *idxp1[j];
*idxp1[j] = gain[j] * (xb + y);
if (endp1[j] <= ++(idxp1[j])) {
idxp1[j] = idx01[j];
}
xb = y - xb;
}
y = 0.5f * (xb + yy1_1);
yy1_1 = y;
y_1_1 = fbk * xb;
*yp1++ = ((wet * y) + (dry * xo1));
}
r->y_1 = y_1 + DENORMAL_PROTECT;
r->yy1 = yy1 + DENORMAL_PROTECT;
r->y_1_0 = y_1_0 + DENORMAL_PROTECT;
r->yy1_0 = yy1_0 + DENORMAL_PROTECT;
r->y_1_1 = y_1_1 + DENORMAL_PROTECT;
r->yy1_1 = yy1_1 + DENORMAL_PROTECT;
}
/******************************************************************************
@ -173,23 +245,25 @@ reverb (b_reverb* r,
#include "lv2/lv2plug.in/ns/lv2core/lv2.h"
typedef enum {
AR_INPUT = 0,
AR_OUTPUT = 1,
AR_MIX = 2,
AR_GAIN_IN = 3,
AR_GAIN_OUT = 4,
AR_INPUT0 = 0,
AR_INPUT1 = 1,
AR_OUTPUT0 = 2,
AR_OUTPUT1 = 3,
AR_MIX = 4,
AR_ROOMSZ = 5,
} PortIndex;
typedef struct {
float* input;
float* output;
float* input0;
float* input1;
float* output0;
float* output1;
float* mix;
float* gain_in;
float* gain_out; // unused
float* roomsz;
float v_mix;
float v_gain_in;
float v_roomsz;
b_reverb r;
} AReverb;
@ -209,7 +283,7 @@ instantiate (const LV2_Descriptor* descriptor,
}
// these are set in initReverb()
self->v_gain_in = -40; // [dB]
self->v_roomsz = 0.75;
self->v_mix = 0.1;
return (LV2_Handle)self;
@ -223,20 +297,23 @@ connect_port (LV2_Handle instance,
AReverb* self = (AReverb*)instance;
switch ((PortIndex)port) {
case AR_INPUT:
self->input = (float*)data;
case AR_INPUT0:
self->input0 = (float*)data;
break;
case AR_OUTPUT:
self->output = (float*)data;
case AR_INPUT1:
self->input1 = (float*)data;
break;
case AR_OUTPUT0:
self->output0 = (float*)data;
break;
case AR_OUTPUT1:
self->output1 = (float*)data;
break;
case AR_MIX:
self->mix = (float*)data;
break;
case AR_GAIN_IN:
self->gain_in = (float*)data;
break;
case AR_GAIN_OUT:
self->gain_out = (float*)data;
case AR_ROOMSZ:
self->roomsz = (float*)data;
break;
}
}
@ -246,8 +323,10 @@ run (LV2_Handle instance, uint32_t n_samples)
{
AReverb* self = (AReverb*)instance;
const float* const input = self->input;
float* const output = self->output;
const float* const input0 = self->input0;
const float* const input1 = self->input1;
float* const output0 = self->output0;
float* const output1 = self->output1;
// TODO interpolate
if (*self->mix != self->v_mix) {
@ -256,31 +335,40 @@ run (LV2_Handle instance, uint32_t n_samples)
self->r.wet = self->v_mix * u;
self->r.dry = u - (self->v_mix * u);
}
if (*self->gain_in != self->v_gain_in) {
self->v_gain_in = *self->gain_in;
self->r.inputGain = powf (10.0, .05 * self->v_gain_in);
}
if (self->gain_out) { // unused
const float g = *self->gain_out;
const float u = self->r.wet + self->r.dry;
self->r.wet = g * (self->r.wet / u);
self->r.dry = g * (self->r.dry / u);
if (*self->roomsz != self->v_roomsz) {
self->v_roomsz = *self->roomsz;
self->r.gain[0] = self->v_roomsz;
self->r.gain[1] = self->v_roomsz;
self->r.gain[2] = self->v_roomsz;
self->r.gain[3] = self->v_roomsz;
self->r.gain[4] = self->v_roomsz;
self->r.gain[5] = self->v_roomsz;
self->r.gain[6] = self->v_roomsz;
self->r.gain[7] = self->v_roomsz;
}
reverb (&self->r, input, output, n_samples);
reverb (&self->r, input0, input1, output0, output1, n_samples);
}
static void
activate (LV2_Handle instance)
{
AReverb* self = (AReverb*)instance;
self->r.y_1 = 0;
self->r.yy1 = 0;
self->r.y_1_0 = 0;
self->r.yy1_0 = 0;
self->r.y_1_1 = 0;
self->r.yy1_1 = 0;
for (int i = 0; i < RV_NZ; ++i) {
self->r.delays[0][i] = NULL;
self->r.delays[1][i] = NULL;
}
}
static void
deactivate (LV2_Handle instance)
{
activate(instance);
}
static void
@ -288,7 +376,8 @@ cleanup (LV2_Handle instance)
{
AReverb* self = (AReverb*)instance;
for (int i = 0; i < RV_NZ; ++i) {
free (self->r.delays[i]);
free (self->r.delays[0][i]);
free (self->r.delays[1][i]);
}
free (instance);
}

53
libs/plugins/a-reverb.lv2/a-reverb.ttl.in
View File

@ -28,34 +28,47 @@
a lv2:AudioPort ,
lv2:InputPort ;
lv2:index 0 ;
lv2:symbol "in" ;
lv2:name "In" ;
lv2:symbol "in0" ;
lv2:name "In 0" ;
],
[
a lv2:AudioPort ,
lv2:InputPort ;
lv2:index 1 ;
lv2:symbol "in1" ;
lv2:name "In 1" ;
],
[
a lv2:AudioPort ,
lv2:OutputPort ;
lv2:index 1 ;
lv2:symbol "out" ;
lv2:name "Out" ;
],
[
a lv2:InputPort ,
lv2:ControlPort ;
lv2:index 3 ;
lv2:symbol "gain_in" ;
lv2:name "Input Gain";
lv2:default -30;
lv2:minimum -80;
lv2:maximum -20;
unit:unit unit:db ;
],
[
a lv2:InputPort ,
lv2:ControlPort ;
lv2:index 2 ;
lv2:symbol "out0" ;
lv2:name "Out 0" ;
],
[
a lv2:AudioPort ,
lv2:OutputPort ;
lv2:index 3 ;
lv2:symbol "out1" ;
lv2:name "Out 1" ;
],
[
a lv2:InputPort ,
lv2:ControlPort ;
lv2:index 4 ;
lv2:symbol "mix" ;
lv2:name "Dry/Wet";
lv2:default 0.3;
lv2:minimum 0.0 ;
lv2:maximum 1.0 ;
],
[
a lv2:InputPort ,
lv2:ControlPort ;
lv2:index 5 ;
lv2:symbol "roomsz" ;
lv2:name "Room Size";
lv2:default 0.5;
lv2:minimum 0.5 ;
lv2:maximum 0.8 ;
] .