*
* Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/fifo.h"
+#include "libavutil/mathematics.h"
#include "avformat.h"
#include "audiointerleave.h"
#include "internal.h"
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
- if (st->codec->codec_type == CODEC_TYPE_AUDIO)
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
av_fifo_free(aic->fifo);
}
}
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
- if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
aic->sample_size = (st->codec->channels *
av_get_bits_per_sample(st->codec->codec_id)) / 8;
if (!aic->sample_size) {
return 0;
}
-static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
- int stream_index, int flush)
+static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
+ int stream_index, int flush)
{
AVStream *st = s->streams[stream_index];
AudioInterleaveContext *aic = st->priv_data;
-
+ int ret;
int size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
if (!size || (!flush && size == av_fifo_size(aic->fifo)))
return 0;
- av_new_packet(pkt, size);
- av_fifo_read(aic->fifo, pkt->data, size);
+ ret = av_new_packet(pkt, size);
+ if (ret < 0)
+ return ret;
+ av_fifo_generic_read(aic->fifo, pkt->data, size, NULL);
pkt->dts = pkt->pts = aic->dts;
pkt->duration = av_rescale_q(*aic->samples, st->time_base, aic->time_base);
int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
{
- int i;
+ int i, ret;
if (pkt) {
AVStream *st = s->streams[pkt->stream_index];
AudioInterleaveContext *aic = st->priv_data;
- if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
unsigned new_size = av_fifo_size(aic->fifo) + pkt->size;
if (new_size > aic->fifo_size) {
if (av_fifo_realloc2(aic->fifo, new_size) < 0)
// rewrite pts and dts to be decoded time line position
pkt->pts = pkt->dts = aic->dts;
aic->dts += pkt->duration;
- ff_interleave_add_packet(s, pkt, compare_ts);
+ if ((ret = ff_interleave_add_packet(s, pkt, compare_ts)) < 0)
+ return ret;
}
pkt = NULL;
}
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
- if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
AVPacket new_pkt;
- while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
- ff_interleave_add_packet(s, &new_pkt, compare_ts);
+ while (interleave_new_audio_packet(s, &new_pkt, i, flush))
+ if ((ret = ff_interleave_add_packet(s, &new_pkt, compare_ts)) < 0)
+ return ret;
}
}
- return get_packet(s, out, pkt, flush);
+ return get_packet(s, out, NULL, flush);
}