1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
FFmpeg/libavformat/sol.c
Michael Niedermayer 99eb31e263 Merge remote-tracking branch 'qatar/master'
* qatar/master: (25 commits)
  Replace custom DEBUG preprocessor trickery by the standard one.
  vorbis: Remove non-compiling debug statement.
  vorbis: Remove pointless DEBUG #ifdef around debug output macros.
  cook: Remove non-compiling debug output.
  Remove pointless #ifdefs around function declarations in a header.
  Replace #ifdef + av_log() combinations by av_dlog().
  Replace custom debug output functions by av_dlog().
  cook: Remove unused debug functions.
  Remove stray extra arguments from av_dlog() invocations.
  targa: fix big-endian build
  v4l2: remove one forgotten use of AVFormatParameters.pix_fmt.
  vfwcap: add a framerate private option.
  v4l2: add a framerate private option.
  libdc1394: add a framerate private option.
  fbdev: add a framerate private option.
  bktr: add a framerate private option.
  oma: check avio_read() return value
  nutdec: remove unused variable
  Remove unused variables
  swscale: allocate larger buffer to handle altivec overreads.
  ...

Conflicts:
	ffmpeg.c
	libavcodec/dca.c
	libavcodec/dirac.c
	libavcodec/error_resilience.c
	libavcodec/h264.c
	libavcodec/mpeg12.c
	libavcodec/mpeg4videodec.c
	libavcodec/mpegvideo.c
	libavcodec/mpegvideo_enc.c
	libavcodec/pthread.c
	libavcodec/rv10.c
	libavcodec/s302m.c
	libavcodec/shorten.c
	libavcodec/truemotion2.c
	libavcodec/utils.c
	libavdevice/dv1394.c
	libavdevice/fbdev.c
	libavdevice/libdc1394.c
	libavdevice/v4l2.c
	libavformat/4xm.c
	libavformat/apetag.c
	libavformat/asfdec.c
	libavformat/avidec.c
	libavformat/mmf.c
	libavformat/mpeg.c
	libavformat/mpegenc.c
	libavformat/mpegts.c
	libavformat/oggdec.c
	libavformat/oggparseogm.c
	libavformat/rl2.c
	libavformat/rmdec.c
	libavformat/rpl.c
	libavformat/rtpdec_latm.c
	libavformat/sauce.c
	libavformat/sol.c
	libswscale/utils.c
	tests/ref/vsynth1/error
	tests/ref/vsynth2/error

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-06-03 05:19:30 +02:00

153 lines
3.9 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);
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 = {
"sol",
NULL_IF_CONFIG_SMALL("Sierra SOL format"),
0,
sol_probe,
sol_read_header,
sol_read_packet,
NULL,
pcm_read_seek,
};