mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-21 10:55:51 +02:00
325f6e0a97
* qatar/master: lavfi: do not export the filters from shared objects Conflicts: libavfilter/af_amix.c libavfilter/af_anull.c libavfilter/asrc_anullsrc.c libavfilter/f_select.c libavfilter/f_settb.c libavfilter/split.c libavfilter/src_movie.c libavfilter/vf_aspect.c libavfilter/vf_blackframe.c libavfilter/vf_colorbalance.c libavfilter/vf_copy.c libavfilter/vf_crop.c libavfilter/vf_cropdetect.c libavfilter/vf_drawbox.c libavfilter/vf_format.c libavfilter/vf_framestep.c libavfilter/vf_frei0r.c libavfilter/vf_hflip.c libavfilter/vf_libopencv.c libavfilter/vf_lut.c libavfilter/vf_null.c libavfilter/vf_overlay.c libavfilter/vf_scale.c libavfilter/vf_transpose.c libavfilter/vf_unsharp.c libavfilter/vf_vflip.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
197 lines
6.1 KiB
C
197 lines
6.1 KiB
C
/*
|
|
* Copyright (c) 2012 Andrey Utkin
|
|
* Copyright (c) 2012 Stefano Sabatini
|
|
*
|
|
* 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
|
|
*/
|
|
|
|
/**
|
|
* @file
|
|
* Filter that changes number of samples on single output operation
|
|
*/
|
|
|
|
#include "libavutil/audio_fifo.h"
|
|
#include "libavutil/avassert.h"
|
|
#include "libavutil/channel_layout.h"
|
|
#include "libavutil/opt.h"
|
|
#include "avfilter.h"
|
|
#include "audio.h"
|
|
#include "internal.h"
|
|
#include "formats.h"
|
|
|
|
typedef struct {
|
|
const AVClass *class;
|
|
int nb_out_samples; ///< how many samples to output
|
|
AVAudioFifo *fifo; ///< samples are queued here
|
|
int64_t next_out_pts;
|
|
int pad;
|
|
} ASNSContext;
|
|
|
|
#define OFFSET(x) offsetof(ASNSContext, x)
|
|
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
|
|
|
|
static const AVOption asetnsamples_options[] = {
|
|
{ "nb_out_samples", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
|
|
{ "n", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, FLAGS },
|
|
{ "pad", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS },
|
|
{ "p", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS },
|
|
{ NULL }
|
|
};
|
|
|
|
AVFILTER_DEFINE_CLASS(asetnsamples);
|
|
|
|
static av_cold int init(AVFilterContext *ctx)
|
|
{
|
|
ASNSContext *asns = ctx->priv;
|
|
|
|
asns->next_out_pts = AV_NOPTS_VALUE;
|
|
av_log(ctx, AV_LOG_VERBOSE, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold void uninit(AVFilterContext *ctx)
|
|
{
|
|
ASNSContext *asns = ctx->priv;
|
|
av_audio_fifo_free(asns->fifo);
|
|
}
|
|
|
|
static int config_props_output(AVFilterLink *outlink)
|
|
{
|
|
ASNSContext *asns = outlink->src->priv;
|
|
|
|
asns->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, asns->nb_out_samples);
|
|
if (!asns->fifo)
|
|
return AVERROR(ENOMEM);
|
|
outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int push_samples(AVFilterLink *outlink)
|
|
{
|
|
ASNSContext *asns = outlink->src->priv;
|
|
AVFrame *outsamples = NULL;
|
|
int ret, nb_out_samples, nb_pad_samples;
|
|
|
|
if (asns->pad) {
|
|
nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0;
|
|
nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo));
|
|
} else {
|
|
nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo));
|
|
nb_pad_samples = 0;
|
|
}
|
|
|
|
if (!nb_out_samples)
|
|
return 0;
|
|
|
|
outsamples = ff_get_audio_buffer(outlink, nb_out_samples);
|
|
if (!outsamples)
|
|
return AVERROR(ENOMEM);
|
|
|
|
av_audio_fifo_read(asns->fifo,
|
|
(void **)outsamples->extended_data, nb_out_samples);
|
|
|
|
if (nb_pad_samples)
|
|
av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples,
|
|
nb_pad_samples, outlink->channels,
|
|
outlink->format);
|
|
outsamples->nb_samples = nb_out_samples;
|
|
outsamples->channel_layout = outlink->channel_layout;
|
|
outsamples->sample_rate = outlink->sample_rate;
|
|
outsamples->pts = asns->next_out_pts;
|
|
|
|
if (asns->next_out_pts != AV_NOPTS_VALUE)
|
|
asns->next_out_pts += nb_out_samples;
|
|
|
|
ret = ff_filter_frame(outlink, outsamples);
|
|
if (ret < 0)
|
|
return ret;
|
|
return nb_out_samples;
|
|
}
|
|
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
ASNSContext *asns = ctx->priv;
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
int ret;
|
|
int nb_samples = insamples->nb_samples;
|
|
|
|
if (av_audio_fifo_space(asns->fifo) < nb_samples) {
|
|
av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples);
|
|
ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples);
|
|
if (ret < 0) {
|
|
av_log(ctx, AV_LOG_ERROR,
|
|
"Stretching audio fifo failed, discarded %d samples\n", nb_samples);
|
|
return -1;
|
|
}
|
|
}
|
|
av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
|
|
if (asns->next_out_pts == AV_NOPTS_VALUE)
|
|
asns->next_out_pts = insamples->pts;
|
|
av_frame_free(&insamples);
|
|
|
|
while (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
|
|
push_samples(outlink);
|
|
return 0;
|
|
}
|
|
|
|
static int request_frame(AVFilterLink *outlink)
|
|
{
|
|
AVFilterLink *inlink = outlink->src->inputs[0];
|
|
int ret;
|
|
|
|
ret = ff_request_frame(inlink);
|
|
if (ret == AVERROR_EOF) {
|
|
ret = push_samples(outlink);
|
|
return ret < 0 ? ret : ret > 0 ? 0 : AVERROR_EOF;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static const AVFilterPad asetnsamples_inputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.filter_frame = filter_frame,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
static const AVFilterPad asetnsamples_outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.request_frame = request_frame,
|
|
.config_props = config_props_output,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
AVFilter ff_af_asetnsamples = {
|
|
.name = "asetnsamples",
|
|
.description = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."),
|
|
.priv_size = sizeof(ASNSContext),
|
|
.priv_class = &asetnsamples_class,
|
|
.init = init,
|
|
.uninit = uninit,
|
|
.inputs = asetnsamples_inputs,
|
|
.outputs = asetnsamples_outputs,
|
|
};
|