/*
* Copyright (c) Markus Schmidt and Christian Holschuh
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/opt.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"
typedef struct LFOContext {
double freq;
double offset;
int srate;
double amount;
double pwidth;
double phase;
} LFOContext;
typedef struct SRContext {
double target;
double real;
double samples;
double last;
} SRContext;
typedef struct ACrusherContext {
const AVClass *class;
double level_in;
double level_out;
double bits;
double mix;
int mode;
double dc;
double idc;
double aa;
double samples;
int is_lfo;
double lforange;
double lforate;
double sqr;
double aa1;
double coeff;
int round;
double sov;
double smin;
double sdiff;
LFOContext lfo;
SRContext *sr;
} ACrusherContext;
#define OFFSET(x) offsetof(ACrusherContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption acrusher_options[] = {
{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "level_out","set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "bits", "set bit reduction", OFFSET(bits), AV_OPT_TYPE_DOUBLE, {.dbl=8}, 1, 64, A },
{ "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, A },
{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "mode" },
{ "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" },
{ "dc", "set DC", OFFSET(dc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, .25, 4, A },
{ "aa", "set anti-aliasing", OFFSET(aa), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, A },
{ "samples", "set sample reduction", OFFSET(samples), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 250, A },
{ "lfo", "enable LFO", OFFSET(is_lfo), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
{ "lforange", "set LFO depth", OFFSET(lforange), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 250, A },
{ "lforate", "set LFO rate", OFFSET(lforate), AV_OPT_TYPE_DOUBLE, {.dbl=.3}, .01, 200, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS(acrusher);
static double samplereduction(ACrusherContext *s, SRContext *sr, double in)
{
sr->samples++;
if (sr->samples >= s->round) {
sr->target += s->samples;
sr->real += s->round;
if (sr->target + s->samples >= sr->real + 1) {
sr->last = in;
sr->target = 0;
sr->real = 0;
}
sr->samples = 0;
}
return sr->last;
}
static double add_dc(double s, double dc, double idc)
{
return s > 0 ? s * dc : s * idc;
}
static double remove_dc(double s, double dc, double idc)
{
return s > 0 ? s * idc : s * dc;
}
static inline double factor(double y, double k, double aa1, double aa)
{
return 0.5 * (sin(M_PI * (fabs(y - k) - aa1) / aa - M_PI_2) + 1);
}
static double bitreduction(ACrusherContext *s, double in)
{
const double sqr = s->sqr;
const double coeff = s->coeff;
const double aa = s->aa;
const double aa1 = s->aa1;
double y, k;
// add dc
in = add_dc(in, s->dc, s->idc);
// main rounding calculation depending on mode
// the idea for anti-aliasing:
// you need a function f which brings you to the scale, where
// you want to round and the function f_b (with f(f_b)=id) which
// brings you back to your original scale.
//
// then you can use the logic below in the following way:
// y = f(in) and k = roundf(y)
// if (y > k + aa1)
// k = f_b(k) + ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
// if (y < k + aa1)
// k = f_b(k) - ( f_b(k+1) - f_b(k) ) * 0.5 * (sin(x - PI/2) + 1)
//
// whereas x = (fabs(f(in) - k) - aa1) * PI / aa
// for both cases.
switch (s->mode) {
case 0:
default:
// linear
y = in * coeff;
k = roundf(y);
if (k - aa1 <= y && y <= k + aa1) {
k /= coeff;
} else if (y > k + aa1) {
k = k / coeff + ((k + 1) / coeff - k / coeff) *
factor(y, k, aa1, aa);
} else {
k = k / coeff - (k / coeff - (k - 1) / coeff) *
factor(y, k, aa1, aa);
}
break;
case 1:
// logarithmic
y = sqr * log(fabs(in)) + sqr * sqr;
k = roundf(y);
if(!in) {
k = 0;
} else if (k - aa1 <= y && y <= k + aa1) {
k = in / fabs(in) * exp(k / sqr - sqr);
} else if (y > k + aa1) {
double x = exp(k / sqr - sqr);
k = FFSIGN(in) * (x + (exp((k + 1) / sqr - sqr) - x) *
factor(y, k, aa1, aa));
} else {
double x = exp(k / sqr - sqr);
k = in / fabs(in) * (x - (x - exp((k - 1) / sqr - sqr)) *
factor(y, k, aa1, aa));
}
break;
}
// mix between dry and wet signal
k += (in - k) * s->mix;
// remove dc
k = remove_dc(k, s->dc, s->idc);
return k;
}
static double lfo_get(LFOContext *lfo)
{
double phs = FFMIN(100., lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
double val;
if (phs > 1)
phs = fmod(phs, 1.);
val = sin((phs * 360.) * M_PI / 180);
return val * lfo->amount;
}
static void lfo_advance(LFOContext *lfo, unsigned count)
{
lfo->phase = fabs(lfo->phase + count * lfo->freq * (1. / lfo->srate));
if (lfo->phase >= 1.)
lfo->phase = fmod(lfo->phase, 1.);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
ACrusherContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out;
const double *src = (const double *)in->data[0];
double *dst;
const double level_in = s->level_in;
const double level_out = s->level_out;
const double mix = s->mix;
int n, c;
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(inlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
dst = (double *)out->data[0];
for (n = 0; n < in->nb_samples; n++) {
if (s->is_lfo) {
s->samples = s->smin + s->sdiff * (lfo_get(&s->lfo) + 0.5);
s->round = round(s->samples);
}
for (c = 0; c < inlink->channels; c++) {
double sample = src[c] * level_in;
sample = mix * samplereduction(s, &s->sr[c], sample) + src[c] * (1. - mix) * level_in;
dst[c] = bitreduction(s, sample) * level_out;
}
src += c;
dst += c;
if (s->is_lfo)
lfo_advance(&s->lfo, 1);
}
if (in != out)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE
};
int ret;
layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
static av_cold void uninit(AVFilterContext *ctx)
{
ACrusherContext *s = ctx->priv;
av_freep(&s->sr);
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
ACrusherContext *s = ctx->priv;
double rad, sunder, smax, sover;
s->idc = 1. / s->dc;
s->coeff = exp2(s->bits) - 1;
s->sqr = sqrt(s->coeff / 2);
s->aa1 = (1. - s->aa) / 2.;
s->round = round(s->samples);
rad = s->lforange / 2.;
s->smin = FFMAX(s->samples - rad, 1.);
sunder = s->samples - rad - s->smin;
smax = FFMIN(s->samples + rad, 250.);
sover = s->samples + rad - smax;
smax -= sunder;
s->smin -= sover;
s->sdiff = smax - s->smin;
s->lfo.freq = s->lforate;
s->lfo.pwidth = 1.;
s->lfo.srate = inlink->sample_rate;
s->lfo.amount = .5;
s->sr = av_calloc(inlink->channels, sizeof(*s->sr));
if (!s->sr)
return AVERROR(ENOMEM);
return 0;
}
static const AVFilterPad avfilter_af_acrusher_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad avfilter_af_acrusher_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_acrusher = {
.name = "acrusher",
.description = NULL_IF_CONFIG_SMALL("Reduce audio bit resolution."),
.priv_size = sizeof(ACrusherContext),
.priv_class = &acrusher_class,
.uninit = uninit,
.query_formats = query_formats,
.inputs = avfilter_af_acrusher_inputs,
.outputs = avfilter_af_acrusher_outputs,
};