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lavfi/buffersink: implement av_buffersink_get_samples().

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Note: the implementation could be more efficient, but at
the cost of more diff.

Most of the code from the following commit:

commit a2cd9be212
Author: Anton Khirnov <anton@khirnov.net>
Date:   Fri May 4 19:22:38 2012 +0200

    lavfi: add an audio buffer sink.

Adapted to call av_buffersink_get_frame_flags() instead of
accessing the frame directly.
This commit is contained in:
Nicolas George 2013-03-10 16:44:46 +01:00
parent b71db3f38e
commit de54a96aa8
1 changed files with 70 additions and 2 deletions
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72
libavfilter/buffersink.c
View File

@ -23,7 +23,7 @@
* buffer sink
*/
#include "libavutil/fifo.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
@ -46,6 +46,10 @@ typedef struct {
int64_t *channel_layouts; ///< list of accepted channel layouts, terminated by -1
int all_channel_counts;
int *sample_rates; ///< list of accepted sample rates, terminated by -1
/* only used for compat API */
AVAudioFifo *audio_fifo; ///< FIFO for audio samples
int64_t next_pts; ///< interpolating audio pts
} BufferSinkContext;
static av_cold void uninit(AVFilterContext *ctx)
@ -53,6 +57,9 @@ static av_cold void uninit(AVFilterContext *ctx)
BufferSinkContext *sink = ctx->priv;
AVFrame *frame;
if (sink->audio_fifo)
av_audio_fifo_free(sink->audio_fifo);
if (sink->fifo) {
while (av_fifo_size(sink->fifo) >= sizeof(AVFilterBufferRef *)) {
av_fifo_generic_read(sink->fifo, &frame, sizeof(frame), NULL);
@ -140,9 +147,70 @@ int av_buffersink_get_frame_flags(AVFilterContext *ctx, AVFrame *frame, int flag
return 0;
}
static int read_from_fifo(AVFilterContext *ctx, AVFrame *frame,
int nb_samples)
{
BufferSinkContext *s = ctx->priv;
AVFilterLink *link = ctx->inputs[0];
AVFrame *tmp;
if (!(tmp = ff_get_audio_buffer(link, nb_samples)))
return AVERROR(ENOMEM);
av_audio_fifo_read(s->audio_fifo, (void**)tmp->extended_data, nb_samples);
tmp->pts = s->next_pts;
s->next_pts += av_rescale_q(nb_samples, (AVRational){1, link->sample_rate},
link->time_base);
av_frame_move_ref(frame, tmp);
av_frame_free(&tmp);
return 0;
}
int av_buffersink_get_samples(AVFilterContext *ctx, AVFrame *frame, int nb_samples)
{
av_assert0(!"TODO");
BufferSinkContext *s = ctx->priv;
AVFilterLink *link = ctx->inputs[0];
AVFrame *cur_frame;
int ret = 0;
if (!s->audio_fifo) {
int nb_channels = link->channels;
if (!(s->audio_fifo = av_audio_fifo_alloc(link->format, nb_channels, nb_samples)))
return AVERROR(ENOMEM);
}
while (ret >= 0) {
if (av_audio_fifo_size(s->audio_fifo) >= nb_samples)
return read_from_fifo(ctx, frame, nb_samples);
if (!(cur_frame = av_frame_alloc()))
return AVERROR(ENOMEM);
ret = av_buffersink_get_frame_flags(ctx, cur_frame, 0);
if (ret == AVERROR_EOF && av_audio_fifo_size(s->audio_fifo)) {
av_frame_free(&cur_frame);
return read_from_fifo(ctx, frame, av_audio_fifo_size(s->audio_fifo));
} else if (ret < 0) {
av_frame_free(&cur_frame);
return ret;
}
if (cur_frame->pts != AV_NOPTS_VALUE) {
s->next_pts = cur_frame->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**)cur_frame->extended_data,
cur_frame->nb_samples);
av_frame_free(&cur_frame);
}
return ret;
}
AVBufferSinkParams *av_buffersink_params_alloc(void)