third_party_ffmpeg/libavformat/gsmdec.c
Hendrik Leppkes 7f5af80ba4 Merge commit 'ce70f28a1732c74a9cd7fec2d56178750bd6e457'
* commit 'ce70f28a1732c74a9cd7fec2d56178750bd6e457':
  avpacket: Replace av_free_packet with av_packet_unref

Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
2015-10-27 14:28:56 +01:00

101 lines
2.9 KiB
C

/*
* RAW GSM demuxer
* Copyright (c) 2011 Justin Ruggles
*
* 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/channel_layout.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "avformat.h"
#include "internal.h"
#define GSM_BLOCK_SIZE 33
#define GSM_BLOCK_SAMPLES 160
#define GSM_SAMPLE_RATE 8000
typedef struct GSMDemuxerContext {
AVClass *class;
int sample_rate;
} GSMDemuxerContext;
static int gsm_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret, size;
size = GSM_BLOCK_SIZE;
pkt->pos = avio_tell(s->pb);
pkt->stream_index = 0;
ret = av_get_packet(s->pb, pkt, size);
if (ret < GSM_BLOCK_SIZE) {
av_packet_unref(pkt);
return ret < 0 ? ret : AVERROR(EIO);
}
pkt->duration = 1;
pkt->pts = pkt->pos / GSM_BLOCK_SIZE;
return 0;
}
static int gsm_read_header(AVFormatContext *s)
{
GSMDemuxerContext *c = s->priv_data;
AVStream *st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = s->iformat->raw_codec_id;
st->codec->channels = 1;
st->codec->channel_layout = AV_CH_LAYOUT_MONO;
st->codec->sample_rate = c->sample_rate;
st->codec->bit_rate = GSM_BLOCK_SIZE * 8 * c->sample_rate / GSM_BLOCK_SAMPLES;
avpriv_set_pts_info(st, 64, GSM_BLOCK_SAMPLES, GSM_SAMPLE_RATE);
return 0;
}
static const AVOption options[] = {
{ "sample_rate", "", offsetof(GSMDemuxerContext, sample_rate),
AV_OPT_TYPE_INT, {.i64 = GSM_SAMPLE_RATE}, 1, INT_MAX / GSM_BLOCK_SIZE,
AV_OPT_FLAG_DECODING_PARAM },
{ NULL },
};
static const AVClass gsm_class = {
.class_name = "gsm demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_gsm_demuxer = {
.name = "gsm",
.long_name = NULL_IF_CONFIG_SMALL("raw GSM"),
.priv_data_size = sizeof(GSMDemuxerContext),
.read_header = gsm_read_header,
.read_packet = gsm_read_packet,
.flags = AVFMT_GENERIC_INDEX,
.extensions = "gsm",
.raw_codec_id = AV_CODEC_ID_GSM,
.priv_class = &gsm_class,
};