virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-02 03:06:28 +02:00
Code Issues Releases Activity
49e018d6fe
FFmpeg/libavfilter/af_acrusher.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h 
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

341 lines
10 KiB
C
Raw Blame History

/*
* 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/mem.h"
#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|AV_OPT_FLAG_RUNTIME_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, .unit = "mode" },
{ "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "mode" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "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->ch_layout.nb_channels; c++) {
double sample = src[c] * level_in;
sample = mix * samplereduction(s, &s->sr[c], sample) + src[c] * (1. - mix) * level_in;
dst[c] = ctx->is_disabled ? src[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 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;
if (!s->sr)
s->sr = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->sr));
if (!s->sr)
return AVERROR(ENOMEM);
return 0;
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
AVFilterLink *inlink = ctx->inputs[0];
int ret;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
return config_input(inlink);
}
static const AVFilterPad avfilter_af_acrusher_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
};
const 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,
FILTER_INPUTS(avfilter_af_acrusher_inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBL),
.process_command = process_command,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};
Reference in New Issue View Git Blame Copy Permalink