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FFmpeg/libavformat/xa.c
Vitor Sessak cdfc38f43b Fix memory leak for truncated frames
Originally committed as revision 21901 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-02-19 20:20:17 +00:00

129 lines
3.7 KiB
C

/*
* Maxis XA (.xa) File Demuxer
* Copyright (c) 2008 Robert Marston
*
* 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
*/
/**
* @file libavformat/xa.c
* Maxis XA File Demuxer
* by Robert Marston (rmarston@gmail.com)
* for more information on the XA audio format see
* http://wiki.multimedia.cx/index.php?title=Maxis_XA
*/
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#define XA00_TAG MKTAG('X', 'A', 0, 0)
#define XAI0_TAG MKTAG('X', 'A', 'I', 0)
#define XAJ0_TAG MKTAG('X', 'A', 'J', 0)
typedef struct MaxisXADemuxContext {
uint32_t out_size;
uint32_t sent_bytes;
uint32_t audio_frame_counter;
} MaxisXADemuxContext;
static int xa_probe(AVProbeData *p)
{
int channels, srate, bits_per_sample;
if (p->buf_size < 24)
return 0;
switch(AV_RL32(p->buf)) {
case XA00_TAG:
case XAI0_TAG:
case XAJ0_TAG:
break;
default:
return 0;
}
channels = AV_RL16(p->buf + 10);
srate = AV_RL32(p->buf + 12);
bits_per_sample = AV_RL16(p->buf + 22);
if (!channels || channels > 8 || !srate || srate > 192000 ||
bits_per_sample < 4 || bits_per_sample > 32)
return 0;
return AVPROBE_SCORE_MAX/2;
}
static int xa_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
MaxisXADemuxContext *xa = s->priv_data;
ByteIOContext *pb = s->pb;
AVStream *st;
/*Set up the XA Audio Decoder*/
st = av_new_stream(s, 0);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = CODEC_TYPE_AUDIO;
st->codec->codec_id = CODEC_ID_ADPCM_EA_MAXIS_XA;
url_fskip(pb, 4); /* Skip the XA ID */
xa->out_size = get_le32(pb);
url_fskip(pb, 2); /* Skip the tag */
st->codec->channels = get_le16(pb);
st->codec->sample_rate = get_le32(pb);
/* Value in file is average byte rate*/
st->codec->bit_rate = get_le32(pb) * 8;
st->codec->block_align = get_le16(pb);
st->codec->bits_per_coded_sample = get_le16(pb);
av_set_pts_info(st, 64, 1, st->codec->sample_rate);
return 0;
}
static int xa_read_packet(AVFormatContext *s,
AVPacket *pkt)
{
MaxisXADemuxContext *xa = s->priv_data;
AVStream *st = s->streams[0];
ByteIOContext *pb = s->pb;
unsigned int packet_size;
int ret;
if(xa->sent_bytes > xa->out_size)
return AVERROR(EIO);
/* 1 byte header and 14 bytes worth of samples * number channels per block */
packet_size = 15*st->codec->channels;
ret = av_get_packet(pb, pkt, packet_size);
if(ret < 0)
return ret;
pkt->stream_index = st->index;
xa->sent_bytes += packet_size;
pkt->pts = xa->audio_frame_counter;
/* 14 bytes Samples per channel with 2 samples per byte */
xa->audio_frame_counter += 28 * st->codec->channels;
return ret;
}
AVInputFormat xa_demuxer = {
"xa",
NULL_IF_CONFIG_SMALL("Maxis XA File Format"),
sizeof(MaxisXADemuxContext),
xa_probe,
xa_read_header,
xa_read_packet,
};