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FFmpeg/libavformat/sol.c
Michael Niedermayer 7c1aba4f01 Merge remote-tracking branch 'qatar/master'
* 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>
2011-09-27 02:14:37 +02:00

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,
};