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FFmpeg/libavfilter/buffersink.c
Michael Niedermayer f8911b987d Merge remote-tracking branch 'qatar/master'
* qatar/master:
  mss3: use standard zigzag table
  mss3: split DSP functions that are used in MTS2(MSS4) into separate file
  motion-test: do not use getopt()
  tcp: add initial timeout limit for incoming connections
  configure: Change the rdtsc check to a linker check
  avconv: propagate fatal errors from lavfi.
  lavfi: add error handling to filter_samples().
  fate-run: make avconv() properly deal with multiple inputs.
  asplit: don't leak the input buffer.
  af_resample: fix request_frame() behavior.
  af_asyncts: fix request_frame() behavior.
  libx264: support aspect ratio switching
  matroskadec: honor error_recognition when encountering unknown elements.
  lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
  lavr: resampling: add filter type and Kaiser window beta to AVOptions
  lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
  lavr: mix: validate internal sample format in ff_audio_mix_init()

Conflicts:
	ffmpeg.c
	ffplay.c
	libavcodec/libx264.c
	libavfilter/audio.c
	libavfilter/split.c
	libavformat/tcp.c
	tests/fate-run.sh

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-07-09 22:40:12 +02:00

175 lines
5.5 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/avassert.h"
#include "libavutil/mathematics.h"
#include "audio.h"
#include "avfilter.h"
#include "buffersink.h"
#include "internal.h"
typedef struct {
AVFilterBufferRef *cur_buf; ///< last buffer delivered on the sink
AVAudioFifo *audio_fifo; ///< FIFO for audio samples
int64_t next_pts; ///< interpolating audio pts
} BufferSinkContext;
static av_cold void uninit(AVFilterContext *ctx)
{
BufferSinkContext *sink = ctx->priv;
if (sink->audio_fifo)
av_audio_fifo_free(sink->audio_fifo);
}
static void start_frame(AVFilterLink *link, AVFilterBufferRef *buf)
{
BufferSinkContext *s = link->dst->priv;
// av_assert0(!s->cur_buf);
s->cur_buf = buf;
link->cur_buf = NULL;
};
static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
{
start_frame(link, buf);
return 0;
}
int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)
{
BufferSinkContext *s = ctx->priv;
AVFilterLink *link = ctx->inputs[0];
int ret;
if (!buf)
return ff_poll_frame(ctx->inputs[0]);
if ((ret = ff_request_frame(link)) < 0)
return ret;
if (!s->cur_buf)
return AVERROR(EINVAL);
*buf = s->cur_buf;
s->cur_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),
.uninit = uninit,
.inputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_VIDEO,
.start_frame = start_frame,
.min_perms = AV_PERM_READ,
.needs_fifo = 1 },
{ .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),
.uninit = uninit,
.inputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ,
.needs_fifo = 1 },
{ .name = NULL }},
.outputs = (AVFilterPad[]) {{ .name = NULL }},
};