mirror of
https://github.com/xenia-project/FFmpeg.git
synced 2024-11-24 20:19:55 +00:00
799d749c77
* qatar/master: (24 commits) yop: set channel layout wtv: set channel layout for mpeg audio westwood_aud: set channel layout wc3movie: set channel layout tmv: set channel layout tiertexseq: set channel layout swfdec: set channel layout sol: set channel layout smacker: set channel layout siff: set channel layout sierravmd: set channel layout rtpdec_amr: set channel layout rsodec: set channel layout rmdec: set channel layout for RA version 3 qcp: set channel layout psxstr: set channel layout omadec: set channel layout oggparsespeex: validate channel count and set channel layout nuv: set channel layout mxg: set channel layout ... Conflicts: libavformat/swfdec.c libavformat/wtv.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
152 lines
4.1 KiB
C
152 lines
4.1 KiB
C
/*
|
|
* Sierra SOL demuxer
|
|
* Copyright Konstantin Shishkov
|
|
*
|
|
* 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
|
|
*/
|
|
|
|
/*
|
|
* Based on documents from Game Audio Player and own research
|
|
*/
|
|
|
|
#include "libavutil/channel_layout.h"
|
|
#include "libavutil/intreadwrite.h"
|
|
#include "avformat.h"
|
|
#include "internal.h"
|
|
#include "pcm.h"
|
|
|
|
/* if we don't know the size in advance */
|
|
#define AU_UNKNOWN_SIZE ((uint32_t)(~0))
|
|
|
|
static int sol_probe(AVProbeData *p)
|
|
{
|
|
/* check file header */
|
|
uint16_t magic = AV_RL32(p->buf);
|
|
if ((magic == 0x0B8D || magic == 0x0C0D || magic == 0x0C8D) &&
|
|
p->buf[2] == 'S' && p->buf[3] == 'O' &&
|
|
p->buf[4] == 'L' && p->buf[5] == 0)
|
|
return AVPROBE_SCORE_MAX;
|
|
else
|
|
return 0;
|
|
}
|
|
|
|
#define SOL_DPCM 1
|
|
#define SOL_16BIT 4
|
|
#define SOL_STEREO 16
|
|
|
|
static enum AVCodecID sol_codec_id(int magic, int type)
|
|
{
|
|
if (magic == 0x0B8D)
|
|
{
|
|
if (type & SOL_DPCM) return AV_CODEC_ID_SOL_DPCM;
|
|
else return AV_CODEC_ID_PCM_U8;
|
|
}
|
|
if (type & SOL_DPCM)
|
|
{
|
|
if (type & SOL_16BIT) return AV_CODEC_ID_SOL_DPCM;
|
|
else if (magic == 0x0C8D) return AV_CODEC_ID_SOL_DPCM;
|
|
else return AV_CODEC_ID_SOL_DPCM;
|
|
}
|
|
if (type & SOL_16BIT) return AV_CODEC_ID_PCM_S16LE;
|
|
return AV_CODEC_ID_PCM_U8;
|
|
}
|
|
|
|
static int sol_codec_type(int magic, int type)
|
|
{
|
|
if (magic == 0x0B8D) return 1;//SOL_DPCM_OLD;
|
|
if (type & SOL_DPCM)
|
|
{
|
|
if (type & SOL_16BIT) return 3;//SOL_DPCM_NEW16;
|
|
else if (magic == 0x0C8D) return 1;//SOL_DPCM_OLD;
|
|
else return 2;//SOL_DPCM_NEW8;
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
static int sol_channels(int magic, int type)
|
|
{
|
|
if (magic == 0x0B8D || !(type & SOL_STEREO)) return 1;
|
|
return 2;
|
|
}
|
|
|
|
static int sol_read_header(AVFormatContext *s)
|
|
{
|
|
unsigned int magic,tag;
|
|
AVIOContext *pb = s->pb;
|
|
unsigned int id, channels, rate, type;
|
|
enum AVCodecID codec;
|
|
AVStream *st;
|
|
|
|
/* check ".snd" header */
|
|
magic = avio_rl16(pb);
|
|
tag = avio_rl32(pb);
|
|
if (tag != MKTAG('S', 'O', 'L', 0))
|
|
return -1;
|
|
rate = avio_rl16(pb);
|
|
type = avio_r8(pb);
|
|
avio_skip(pb, 4); /* size */
|
|
if (magic != 0x0B8D)
|
|
avio_r8(pb); /* newer SOLs contain padding byte */
|
|
|
|
codec = sol_codec_id(magic, type);
|
|
channels = sol_channels(magic, type);
|
|
|
|
if (codec == AV_CODEC_ID_SOL_DPCM)
|
|
id = sol_codec_type(magic, type);
|
|
else id = 0;
|
|
|
|
/* now we are ready: build format streams */
|
|
st = avformat_new_stream(s, NULL);
|
|
if (!st)
|
|
return -1;
|
|
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
|
|
st->codec->codec_tag = id;
|
|
st->codec->codec_id = codec;
|
|
st->codec->channels = channels;
|
|
st->codec->channel_layout = channels == 1 ? AV_CH_LAYOUT_MONO :
|
|
AV_CH_LAYOUT_STEREO;
|
|
st->codec->sample_rate = rate;
|
|
avpriv_set_pts_info(st, 64, 1, rate);
|
|
return 0;
|
|
}
|
|
|
|
#define MAX_SIZE 4096
|
|
|
|
static int sol_read_packet(AVFormatContext *s,
|
|
AVPacket *pkt)
|
|
{
|
|
int ret;
|
|
|
|
if (url_feof(s->pb))
|
|
return AVERROR(EIO);
|
|
ret= av_get_packet(s->pb, pkt, MAX_SIZE);
|
|
if (ret < 0)
|
|
return ret;
|
|
pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
|
|
pkt->stream_index = 0;
|
|
return 0;
|
|
}
|
|
|
|
AVInputFormat ff_sol_demuxer = {
|
|
.name = "sol",
|
|
.long_name = NULL_IF_CONFIG_SMALL("Sierra SOL"),
|
|
.read_probe = sol_probe,
|
|
.read_header = sol_read_header,
|
|
.read_packet = sol_read_packet,
|
|
.read_seek = ff_pcm_read_seek,
|
|
};
|