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authorNicolas George <nicolas.george@normalesup.org>2013-03-10 16:44:46 +0100
committerNicolas George <nicolas.george@normalesup.org>2013-03-10 19:39:25 +0100
commitde54a96aa8f5959ebd87516d514c77055b369fed (patch)
tree491fb7fae2bd8178c0ff42e9777596716ff3d509 /libavfilter
parentb71db3f38eafeb05e068804e002984b0d232e5f6 (diff)
downloadffmpeg-de54a96aa8f5959ebd87516d514c77055b369fed.tar.gz
lavfi/buffersink: implement av_buffersink_get_samples().
Note: the implementation could be more efficient, but at the cost of more diff. Most of the code from the following commit: commit a2cd9be212fca02dd3d6ee65cb6ab9f84c5e28e5 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.
Diffstat (limited to 'libavfilter')
-rw-r--r--libavfilter/buffersink.c72
1 files changed, 70 insertions, 2 deletions
diff --git a/libavfilter/buffersink.c b/libavfilter/buffersink.c
index 9f92051c2b..35607850f0 100644
--- a/libavfilter/buffersink.c
+++ b/libavfilter/buffersink.c
@@ -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)