mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-12 19:18:44 +02:00
7c1aba4f01
* qatar/master: (21 commits) fate: allow testing with libavfilter disabled x86: XOP/FMA4 CPU detection support ws_snd: misc cosmetic clean-ups ws_snd: remove the 2-bit ADPCM table and just subtract 2 instead. ws_snd: use memcpy() and memset() instead of loops ws_snd: use samples pointer for loop termination instead of a separate iterator variable. ws_snd: make sure number of channels is 1 ws_snd: add some checks to prevent buffer overread or overwrite. ws_snd: decode to AV_SAMPLE_FMT_U8 instead of S16. flacdec: fix buffer size checking in get_metadata_size() rtp: Simplify ff_rtp_get_payload_type rtpenc: Add a payload type private option rtp: Correct ff_rtp_get_payload_type documentation avconv: replace all fprintf() by av_log(). avconv: change av_log verbosity from ERROR to FATAL for fatal errors. cmdutils: replace fprintf() by av_log() avtools: parse loglevel before all the other options. oggdec: add support for Xiph's CELT codec sol: return error if av_get_packet() fails. cosmetics: reindent and pretty-print ... Conflicts: avconv.c cmdutils.c libavcodec/avcodec.h libavcodec/version.h libavformat/oggparsecelt.c libavformat/utils.c libavutil/avutil.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
153 lines
4.0 KiB
C
153 lines
4.0 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/bswap.h"
|
|
#include "avformat.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;
|
|
magic=av_le2ne16(*((uint16_t*)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 CodecID sol_codec_id(int magic, int type)
|
|
{
|
|
if (magic == 0x0B8D)
|
|
{
|
|
if (type & SOL_DPCM) return CODEC_ID_SOL_DPCM;
|
|
else return CODEC_ID_PCM_U8;
|
|
}
|
|
if (type & SOL_DPCM)
|
|
{
|
|
if (type & SOL_16BIT) return CODEC_ID_SOL_DPCM;
|
|
else if (magic == 0x0C8D) return CODEC_ID_SOL_DPCM;
|
|
else return CODEC_ID_SOL_DPCM;
|
|
}
|
|
if (type & SOL_16BIT) return CODEC_ID_PCM_S16LE;
|
|
return 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,
|
|
AVFormatParameters *ap)
|
|
{
|
|
unsigned int magic,tag;
|
|
AVIOContext *pb = s->pb;
|
|
unsigned int id, channels, rate, type;
|
|
enum CodecID 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 == CODEC_ID_SOL_DPCM)
|
|
id = sol_codec_type(magic, type);
|
|
else id = 0;
|
|
|
|
/* now we are ready: build format streams */
|
|
st = av_new_stream(s, 0);
|
|
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->sample_rate = rate;
|
|
av_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->stream_index = 0;
|
|
|
|
/* note: we need to modify the packet size here to handle the last
|
|
packet */
|
|
pkt->size = ret;
|
|
return 0;
|
|
}
|
|
|
|
AVInputFormat ff_sol_demuxer = {
|
|
.name = "sol",
|
|
.long_name = NULL_IF_CONFIG_SMALL("Sierra SOL format"),
|
|
.read_probe = sol_probe,
|
|
.read_header = sol_read_header,
|
|
.read_packet = sol_read_packet,
|
|
.read_seek = pcm_read_seek,
|
|
};
|