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FFmpeg/libavfilter/af_aap.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

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/*
* Copyright (c) 2023 Paul B Mahol
*
* 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/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "filters.h"
#include "internal.h"
enum OutModes {
IN_MODE,
DESIRED_MODE,
OUT_MODE,
NOISE_MODE,
ERROR_MODE,
NB_OMODES
};
typedef struct AudioAPContext {
const AVClass *class;
int order;
int projection;
float mu;
float delta;
int output_mode;
int precision;
int kernel_size;
AVFrame *offset;
AVFrame *delay;
AVFrame *coeffs;
AVFrame *e;
AVFrame *p;
AVFrame *x;
AVFrame *w;
AVFrame *dcoeffs;
AVFrame *tmp;
AVFrame *tmpm;
AVFrame *itmpm;
void **tmpmp;
void **itmpmp;
AVFrame *frame[2];
int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
AVFloatDSPContext *fdsp;
} AudioAPContext;
#define OFFSET(x) offsetof(AudioAPContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption aap_options[] = {
{ "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=16}, 1, INT16_MAX, A },
{ "projection", "set the filter projection", OFFSET(projection), AV_OPT_TYPE_INT, {.i64=2}, 1, 256, A },
{ "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.0001},0,1, AT },
{ "delta", "set the filter delta", OFFSET(delta), AV_OPT_TYPE_FLOAT, {.dbl=0.001},0, 1, AT },
{ "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, .unit = "mode" },
{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, .unit = "mode" },
{ "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, .unit = "mode" },
{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, .unit = "mode" },
{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, .unit = "mode" },
{ "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, .unit = "mode" },
{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, .unit = "precision" },
{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "precision" },
{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "precision" },
{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, .unit = "precision" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(aap);
static int query_formats(AVFilterContext *ctx)
{
AudioAPContext *s = ctx->priv;
static const enum AVSampleFormat sample_fmts[3][3] = {
{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
{ AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
};
int ret;
if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
return ret;
if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static int activate(AVFilterContext *ctx)
{
AudioAPContext *s = ctx->priv;
int i, ret, status;
int nb_samples;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]),
ff_inlink_queued_samples(ctx->inputs[1]));
for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) {
if (s->frame[i])
continue;
if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) {
ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]);
if (ret < 0)
return ret;
}
}
if (s->frame[0] && s->frame[1]) {
AVFrame *out;
out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples);
if (!out) {
av_frame_free(&s->frame[0]);
av_frame_free(&s->frame[1]);
return AVERROR(ENOMEM);
}
ff_filter_execute(ctx, s->filter_channels, out, NULL,
FFMIN(ctx->outputs[0]->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
out->pts = s->frame[0]->pts;
out->duration = s->frame[0]->duration;
av_frame_free(&s->frame[0]);
av_frame_free(&s->frame[1]);
ret = ff_filter_frame(ctx->outputs[0], out);
if (ret < 0)
return ret;
}
if (!nb_samples) {
for (i = 0; i < 2; i++) {
if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
ff_outlink_set_status(ctx->outputs[0], status, pts);
return 0;
}
}
}
if (ff_outlink_frame_wanted(ctx->outputs[0])) {
for (i = 0; i < 2; i++) {
if (s->frame[i] || ff_inlink_queued_samples(ctx->inputs[i]) > 0)
continue;
ff_inlink_request_frame(ctx->inputs[i]);
return 0;
}
}
return 0;
}
#define DEPTH 32
#include "aap_template.c"
#undef DEPTH
#define DEPTH 64
#include "aap_template.c"
static int config_output(AVFilterLink *outlink)
{
const int channels = outlink->ch_layout.nb_channels;
AVFilterContext *ctx = outlink->src;
AudioAPContext *s = ctx->priv;
s->kernel_size = FFALIGN(s->order, 16);
if (!s->offset)
s->offset = ff_get_audio_buffer(outlink, 3);
if (!s->delay)
s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
if (!s->dcoeffs)
s->dcoeffs = ff_get_audio_buffer(outlink, s->kernel_size);
if (!s->coeffs)
s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
if (!s->e)
s->e = ff_get_audio_buffer(outlink, 2 * s->projection);
if (!s->p)
s->p = ff_get_audio_buffer(outlink, s->projection + 1);
if (!s->x)
s->x = ff_get_audio_buffer(outlink, 2 * (s->projection + s->order));
if (!s->w)
s->w = ff_get_audio_buffer(outlink, s->projection);
if (!s->tmp)
s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
if (!s->tmpm)
s->tmpm = ff_get_audio_buffer(outlink, s->projection * s->projection);
if (!s->itmpm)
s->itmpm = ff_get_audio_buffer(outlink, s->projection * s->projection);
if (!s->tmpmp)
s->tmpmp = av_calloc(s->projection * channels, sizeof(*s->tmpmp));
if (!s->itmpmp)
s->itmpmp = av_calloc(s->projection * channels, sizeof(*s->itmpmp));
if (!s->offset || !s->delay || !s->dcoeffs || !s->coeffs || !s->tmpmp || !s->itmpmp ||
!s->e || !s->p || !s->x || !s->w || !s->tmp || !s->tmpm || !s->itmpm)
return AVERROR(ENOMEM);
switch (outlink->format) {
case AV_SAMPLE_FMT_DBLP:
for (int ch = 0; ch < channels; ch++) {
double *itmpm = (double *)s->itmpm->extended_data[ch];
double *tmpm = (double *)s->tmpm->extended_data[ch];
double **itmpmp = (double **)&s->itmpmp[s->projection * ch];
double **tmpmp = (double **)&s->tmpmp[s->projection * ch];
for (int i = 0; i < s->projection; i++) {
itmpmp[i] = &itmpm[i * s->projection];
tmpmp[i] = &tmpm[i * s->projection];
}
}
s->filter_channels = filter_channels_double;
break;
case AV_SAMPLE_FMT_FLTP:
for (int ch = 0; ch < channels; ch++) {
float *itmpm = (float *)s->itmpm->extended_data[ch];
float *tmpm = (float *)s->tmpm->extended_data[ch];
float **itmpmp = (float **)&s->itmpmp[s->projection * ch];
float **tmpmp = (float **)&s->tmpmp[s->projection * ch];
for (int i = 0; i < s->projection; i++) {
itmpmp[i] = &itmpm[i * s->projection];
tmpmp[i] = &tmpm[i * s->projection];
}
}
s->filter_channels = filter_channels_float;
break;
}
return 0;
}
static av_cold int init(AVFilterContext *ctx)
{
AudioAPContext *s = ctx->priv;
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
return AVERROR(ENOMEM);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioAPContext *s = ctx->priv;
av_freep(&s->fdsp);
av_frame_free(&s->offset);
av_frame_free(&s->delay);
av_frame_free(&s->dcoeffs);
av_frame_free(&s->coeffs);
av_frame_free(&s->e);
av_frame_free(&s->p);
av_frame_free(&s->w);
av_frame_free(&s->x);
av_frame_free(&s->tmp);
av_frame_free(&s->tmpm);
av_frame_free(&s->itmpm);
av_freep(&s->tmpmp);
av_freep(&s->itmpmp);
}
static const AVFilterPad inputs[] = {
{
.name = "input",
.type = AVMEDIA_TYPE_AUDIO,
},
{
.name = "desired",
.type = AVMEDIA_TYPE_AUDIO,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_af_aap = {
.name = "aap",
.description = NULL_IF_CONFIG_SMALL("Apply Affine Projection algorithm to first audio stream."),
.priv_size = sizeof(AudioAPContext),
.priv_class = &aap_class,
.init = init,
.uninit = uninit,
.activate = activate,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
.process_command = ff_filter_process_command,
};
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