ffmpeg/libavfilter/af_aderivative.c
Andreas Rheinhardt 8be701d9f7 avfilter/avfilter: Add numbers of (in|out)pads directly to AVFilter
Up until now, an AVFilter's lists of input and output AVFilterPads
were terminated by a sentinel and the only way to get the length
of these lists was by using avfilter_pad_count(). This has two
drawbacks: first, sizeof(AVFilterPad) is not negligible
(i.e. 64B on 64bit systems); second, getting the size involves
a function call instead of just reading the data.

This commit therefore changes this. The sentinels are removed and new
private fields nb_inputs and nb_outputs are added to AVFilter that
contain the number of elements of the respective AVFilterPad array.

Given that AVFilter.(in|out)puts are the only arrays of zero-terminated
AVFilterPads an API user has access to (AVFilterContext.(in|out)put_pads
are not zero-terminated and they already have a size field) the argument
to avfilter_pad_count() is always one of these lists, so it just has to
find the filter the list belongs to and read said number. This is slower
than before, but a replacement function that just reads the internal numbers
that users are expected to switch to will be added soon; and furthermore,
avfilter_pad_count() is probably never called in hot loops anyway.

This saves about 49KiB from the binary; notice that these sentinels are
not in .bss despite being zeroed: they are in .data.rel.ro due to the
non-sentinels.

Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-08-20 12:53:58 +02:00

194 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 "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;
static int query_formats(AVFilterContext *ctx)
{
static const enum AVSampleFormat derivative_sample_fmts[] = {
AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
static const enum AVSampleFormat integral_sample_fmts[] = {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret = ff_set_common_formats_from_list(ctx, strcmp(ctx->filter->name, "aintegral") ?
derivative_sample_fmts : integral_sample_fmts);
if (ret < 0)
return ret;
ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
#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 = 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->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 AVFilterPad aderivative_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
const AVFilter ff_af_aderivative = {
.name = "aderivative",
.description = NULL_IF_CONFIG_SMALL("Compute derivative of input audio."),
.query_formats = query_formats,
.priv_size = sizeof(ADerivativeContext),
.uninit = uninit,
FILTER_INPUTS(aderivative_inputs),
FILTER_OUTPUTS(aderivative_outputs),
};
const AVFilter ff_af_aintegral = {
.name = "aintegral",
.description = NULL_IF_CONFIG_SMALL("Compute integral of input audio."),
.query_formats = query_formats,
.priv_size = sizeof(ADerivativeContext),
.uninit = uninit,
FILTER_INPUTS(aderivative_inputs),
FILTER_OUTPUTS(aderivative_outputs),
};