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FFmpeg/libavfilter/af_aderivative.c
Andreas Rheinhardt 50ea7389ec avfilter: Deduplicate default audio inputs/outputs
Lots of audio filters use very simple inputs or outputs:
An array with a single AVFilterPad whose name is "default"
and whose type is AVMEDIA_TYPE_AUDIO; everything else is unset.

Given that we never use pointer equality for inputs or outputs*,
we can simply use a single AVFilterPad instead of dozens; this
even saves .data.rel.ro (4784B here) as well as relocations.

*: In fact, several filters (like the filters in af_biquads.c)
already use the same inputs; furthermore, ff_filter_alloc()
duplicates the input and output pads so that we do not even
work with the pads directly.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2023-08-07 09:21:13 +02:00

186 lines
7.0 KiB
C

/*
* 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 "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct ADerivativeContext {
const AVClass *class;
AVFrame *prev;
void (*filter)(void **dst, void **prv, const void **src,
int nb_samples, int channels);
} ADerivativeContext;
#define DERIVATIVE(name, type) \
static void aderivative_## name ##p(void **d, void **p, const void **s, \
int nb_samples, int channels) \
{ \
int n, c; \
\
for (c = 0; c < channels; c++) { \
const type *src = s[c]; \
type *dst = d[c]; \
type *prv = p[c]; \
\
for (n = 0; n < nb_samples; n++) { \
const type current = src[n]; \
\
dst[n] = current - prv[0]; \
prv[0] = current; \
} \
} \
}
DERIVATIVE(flt, float)
DERIVATIVE(dbl, double)
DERIVATIVE(s16, int16_t)
DERIVATIVE(s32, int32_t)
#define INTEGRAL(name, type) \
static void aintegral_## name ##p(void **d, void **p, const void **s, \
int nb_samples, int channels) \
{ \
int n, c; \
\
for (c = 0; c < channels; c++) { \
const type *src = s[c]; \
type *dst = d[c]; \
type *prv = p[c]; \
\
for (n = 0; n < nb_samples; n++) { \
const type current = src[n]; \
\
dst[n] = current + prv[0]; \
prv[0] = dst[n]; \
} \
} \
}
INTEGRAL(flt, float)
INTEGRAL(dbl, double)
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
ADerivativeContext *s = ctx->priv;
switch (inlink->format) {
case AV_SAMPLE_FMT_FLTP: s->filter = aderivative_fltp; break;
case AV_SAMPLE_FMT_DBLP: s->filter = aderivative_dblp; break;
case AV_SAMPLE_FMT_S32P: s->filter = aderivative_s32p; break;
case AV_SAMPLE_FMT_S16P: s->filter = aderivative_s16p; break;
}
if (strcmp(ctx->filter->name, "aintegral"))
return 0;
switch (inlink->format) {
case AV_SAMPLE_FMT_FLTP: s->filter = aintegral_fltp; break;
case AV_SAMPLE_FMT_DBLP: s->filter = aintegral_dblp; break;
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
ADerivativeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out;
if (ctx->is_disabled) {
if (s->prev)
av_samples_set_silence(s->prev->extended_data, 0, 1,
s->prev->ch_layout.nb_channels,
s->prev->format);
return ff_filter_frame(outlink, in);
}
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
if (!s->prev) {
s->prev = ff_get_audio_buffer(inlink, 1);
if (!s->prev) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
}
s->filter((void **)out->extended_data, (void **)s->prev->extended_data, (const void **)in->extended_data,
in->nb_samples, in->ch_layout.nb_channels);
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
ADerivativeContext *s = ctx->priv;
av_frame_free(&s->prev);
}
static const AVFilterPad aderivative_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
static const AVOption aderivative_options[] = {
{ NULL }
};
AVFILTER_DEFINE_CLASS_EXT(aderivative, "aderivative/aintegral", aderivative_options);
const AVFilter ff_af_aderivative = {
.name = "aderivative",
.description = NULL_IF_CONFIG_SMALL("Compute derivative of input audio."),
.priv_size = sizeof(ADerivativeContext),
.priv_class = &aderivative_class,
.uninit = uninit,
FILTER_INPUTS(aderivative_inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_DBLP),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};
const AVFilter ff_af_aintegral = {
.name = "aintegral",
.description = NULL_IF_CONFIG_SMALL("Compute integral of input audio."),
.priv_size = sizeof(ADerivativeContext),
.priv_class = &aderivative_class,
.uninit = uninit,
FILTER_INPUTS(aderivative_inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};