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/*
* Audio Interleaving functions
*
* Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
*
* 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
*/
#include "libavutil/fifo.h"
#include "libavutil/mathematics.h"
#include "avformat.h"
#include "audiointerleave.h"
#include "internal.h"
void ff_audio_interleave_close(AVFormatContext *s)
{
int i;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
av_fifo_free(aic->fifo);
}
}
int ff_audio_interleave_init(AVFormatContext *s,
const int *samples_per_frame,
AVRational time_base)
{
int i;
if (!samples_per_frame)
return -1;
if (!time_base.num) {
av_log(s, AV_LOG_ERROR, "timebase not set for audio interleave\n");
return -1;
}
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
AudioInterleaveContext *aic = st->priv_data;
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) {
av_log(s, AV_LOG_ERROR, "could not compute sample size\n");
return -1;
}
aic->samples_per_frame = samples_per_frame;
aic->samples = aic->samples_per_frame;
aic->time_base = time_base;
aic->fifo_size = 100* *aic->samples;
aic->fifo= av_fifo_alloc(100 * *aic->samples);
}
}
return 0;
}
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 size = FFMIN(av_fifo_size(aic->fifo), *aic->samples * aic->sample_size);
if (!size || (!flush && size == av_fifo_size(aic->fifo)))
return 0;
if (av_new_packet(pkt, size) < 0)
return AVERROR(ENOMEM);
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);
pkt->stream_index = stream_index;
aic->dts += pkt->duration;
aic->samples++;
if (!*aic->samples)
aic->samples = aic->samples_per_frame;
return size;
}
int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt, int flush,
int (*get_packet)(AVFormatContext *, AVPacket *, AVPacket *, int),
int (*compare_ts)(AVFormatContext *, AVPacket *, AVPacket *))
{
int i;
if (pkt) {
AVStream *st = s->streams[pkt->stream_index];
AudioInterleaveContext *aic = st->priv_data;
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)
return -1;
aic->fifo_size = new_size;
}
av_fifo_generic_write(aic->fifo, pkt->data, pkt->size, NULL);
} else {
int ret;
// rewrite pts and dts to be decoded time line position
pkt->pts = pkt->dts = aic->dts;
aic->dts += pkt->duration;
ret = ff_interleave_add_packet(s, pkt, compare_ts);
if (ret < 0)
return ret;
}
pkt = NULL;
}
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
AVPacket new_pkt;
int ret;
while ((ret = interleave_new_audio_packet(s, &new_pkt, i, flush)) > 0) {
ret = ff_interleave_add_packet(s, &new_pkt, compare_ts);
if (ret < 0)
return ret;
}
if (ret < 0)
return ret;
}
}
return get_packet(s, out, NULL, flush);
}
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