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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavfilter/buffersink.c
Michael Niedermayer 1cbf7fb434 Merge remote-tracking branch 'qatar/master'
* qatar/master: (26 commits)
  fate: use diff -b in oneline comparison
  Add missing version bumps and APIchanges/Changelog entries.
  lavfi: move buffer management function to a separate file.
  lavfi: move formats-related functions from default.c to formats.c
  lavfi: move video-related functions to a separate file.
  fate: make smjpeg a demux test
  fate: separate sierra-vmd audio and video tests
  fate: separate smacker audio and video tests
  libmp3lame: set supported channel layouts.
  avconv: automatically insert asyncts when -async is used.
  avconv: add support for audio filters.
  lavfi: add asyncts filter.
  lavfi: add aformat filter
  lavfi: add an audio buffer sink.
  lavfi: add an audio buffer source.
  buffersrc: add av_buffersrc_write_frame().
  buffersrc: fix invalid read in uninit if the fifo hasn't been allocated
  lavfi: rename vsrc_buffer.c to buffersrc.c
  avfiltergraph: reindent
  lavfi: add channel layout/sample rate negotiation.
  ...

Conflicts:
	Changelog
	doc/APIchanges
	doc/filters.texi
	ffmpeg.c
	ffprobe.c
	libavcodec/libmp3lame.c
	libavfilter/Makefile
	libavfilter/af_aformat.c
	libavfilter/allfilters.c
	libavfilter/avfilter.c
	libavfilter/avfilter.h
	libavfilter/avfiltergraph.c
	libavfilter/buffersrc.c
	libavfilter/defaults.c
	libavfilter/formats.c
	libavfilter/src_buffer.c
	libavfilter/version.h
	libavfilter/vf_yadif.c
	libavfilter/vsrc_buffer.c
	libavfilter/vsrc_buffer.h
	libavutil/avutil.h
	tests/fate/audio.mak
	tests/fate/demux.mak
	tests/fate/video.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-16 02:27:31 +02:00

209 lines
6.5 KiB
C

/*
* Copyright (c) 2011 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
* buffer sink
*/
#include "libavutil/audio_fifo.h"
#include "libavutil/audioconvert.h"
#include "libavutil/fifo.h"
#include "libavutil/mathematics.h"
#include "audio.h"
#include "avfilter.h"
#include "buffersink.h"
typedef struct {
AVFifoBuffer *fifo; ///< FIFO buffer of frame references
AVAudioFifo *audio_fifo; ///< FIFO for audio samples
int64_t next_pts; ///< interpolating audio pts
} BufferSinkContext;
#define FIFO_INIT_SIZE 8
static av_cold void uninit(AVFilterContext *ctx)
{
BufferSinkContext *sink = ctx->priv;
while (sink->fifo && av_fifo_size(sink->fifo)) {
AVFilterBufferRef *buf;
av_fifo_generic_read(sink->fifo, &buf, sizeof(buf), NULL);
avfilter_unref_buffer(buf);
}
av_fifo_free(sink->fifo);
if (sink->audio_fifo)
av_audio_fifo_free(sink->audio_fifo);
}
static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
{
BufferSinkContext *sink = ctx->priv;
if (!(sink->fifo = av_fifo_alloc(FIFO_INIT_SIZE*sizeof(AVFilterBufferRef*)))) {
av_log(ctx, AV_LOG_ERROR, "Failed to allocate fifo\n");
return AVERROR(ENOMEM);
}
return 0;
}
static void write_buf(AVFilterContext *ctx, AVFilterBufferRef *buf)
{
BufferSinkContext *sink = ctx->priv;
if (av_fifo_space(sink->fifo) < sizeof(AVFilterBufferRef *) &&
(av_fifo_realloc2(sink->fifo, av_fifo_size(sink->fifo) * 2) < 0)) {
av_log(ctx, AV_LOG_ERROR, "Error reallocating the FIFO.\n");
return;
}
av_fifo_generic_write(sink->fifo, &buf, sizeof(buf), NULL);
}
static void end_frame(AVFilterLink *link)
{
write_buf(link->dst, link->cur_buf);
link->cur_buf = NULL;
}
static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
{
write_buf(link->dst, buf);
}
int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)
{
BufferSinkContext *sink = ctx->priv;
AVFilterLink *link = ctx->inputs[0];
int ret;
if (!buf) {
if (av_fifo_size(sink->fifo))
return av_fifo_size(sink->fifo)/sizeof(*buf);
else
return avfilter_poll_frame(ctx->inputs[0]);
}
if (!av_fifo_size(sink->fifo) &&
(ret = avfilter_request_frame(link)) < 0)
return ret;
if (!av_fifo_size(sink->fifo))
return AVERROR(EINVAL);
av_fifo_generic_read(sink->fifo, buf, sizeof(*buf), NULL);
return 0;
}
static int read_from_fifo(AVFilterContext *ctx, AVFilterBufferRef **pbuf,
int nb_samples)
{
BufferSinkContext *s = ctx->priv;
AVFilterLink *link = ctx->inputs[0];
AVFilterBufferRef *buf;
if (!(buf = ff_get_audio_buffer(link, AV_PERM_WRITE, nb_samples)))
return AVERROR(ENOMEM);
av_audio_fifo_read(s->audio_fifo, (void**)buf->extended_data, nb_samples);
buf->pts = s->next_pts;
s->next_pts += av_rescale_q(nb_samples, (AVRational){1, link->sample_rate},
link->time_base);
*pbuf = buf;
return 0;
}
int av_buffersink_read_samples(AVFilterContext *ctx, AVFilterBufferRef **pbuf,
int nb_samples)
{
BufferSinkContext *s = ctx->priv;
AVFilterLink *link = ctx->inputs[0];
int ret = 0;
if (!s->audio_fifo) {
int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
if (!(s->audio_fifo = av_audio_fifo_alloc(link->format, nb_channels, nb_samples)))
return AVERROR(ENOMEM);
}
while (ret >= 0) {
AVFilterBufferRef *buf;
if (av_audio_fifo_size(s->audio_fifo) >= nb_samples)
return read_from_fifo(ctx, pbuf, nb_samples);
ret = av_buffersink_read(ctx, &buf);
if (ret == AVERROR_EOF && av_audio_fifo_size(s->audio_fifo))
return read_from_fifo(ctx, pbuf, av_audio_fifo_size(s->audio_fifo));
else if (ret < 0)
return ret;
if (buf->pts != AV_NOPTS_VALUE) {
s->next_pts = buf->pts -
av_rescale_q(av_audio_fifo_size(s->audio_fifo),
(AVRational){ 1, link->sample_rate },
link->time_base);
}
ret = av_audio_fifo_write(s->audio_fifo, (void**)buf->extended_data,
buf->audio->nb_samples);
avfilter_unref_buffer(buf);
}
return ret;
}
AVFilter avfilter_vsink_buffer = {
.name = "buffersink_old",
.description = NULL_IF_CONFIG_SMALL("Buffer video frames, and make them available to the end of the filter graph."),
.priv_size = sizeof(BufferSinkContext),
.init = init,
.uninit = uninit,
.inputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_VIDEO,
.end_frame = end_frame,
.min_perms = AV_PERM_READ, },
{ .name = NULL }},
.outputs = (AVFilterPad[]) {{ .name = NULL }},
};
AVFilter avfilter_asink_abuffer = {
.name = "abuffersink_old",
.description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them available to the end of the filter graph."),
.priv_size = sizeof(BufferSinkContext),
.init = init,
.uninit = uninit,
.inputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ, },
{ .name = NULL }},
.outputs = (AVFilterPad[]) {{ .name = NULL }},
};