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ffmpeg/libavfilter/af_aap.c

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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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