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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavformat/aea.c
Michael Niedermayer 78accb876c Merge remote-tracking branch 'qatar/master'
* qatar/master:
  ffmpeg: fix some indentation
  ffmpeg: fix operation with --disable-avfilter
  simple_idct: remove disabled code
  motion_est: remove disabled code
  vc1: remove disabled code
  fate: separate lavf-mxf_d10 test from lavf-mxf
  cabac: Move code only used in the cabac test program to cabac.c.
  ffplay: warn that -pix_fmt is no longer working, suggest alternative
  ffplay: warn that -s is no longer working, suggest alternative
  lavf: rename enc variable in utils.c:has_codec_parameters()
  lavf: use designated initialisers for all (de)muxers.
  wav: remove a use of deprecated AV_METADATA_ macro
  rmdec: remove useless ap parameter from rm_read_header_old()
  dct-test: remove write-only variable
  des: fix #if conditional around P_shuffle
  Use LOCAL_ALIGNED in ff_check_alignment()

Conflicts:
	ffmpeg.c
	libavformat/avidec.c
	libavformat/matroskaenc.c
	libavformat/mp3enc.c
	libavformat/oggenc.c
	libavformat/utils.c
	tests/ref/lavf/mxf

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-07-17 20:12:02 +02:00

108 lines
3.1 KiB
C

/*
* MD STUDIO audio demuxer
*
* Copyright (c) 2009 Benjamin Larsson
*
* 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
*/
#include "avformat.h"
#include "pcm.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/audioconvert.h"
#define AT1_SU_SIZE 212
static int aea_read_probe(AVProbeData *p)
{
if (p->buf_size <= 2048+212)
return 0;
/* Magic is '00 08 00 00' in Little Endian*/
if (AV_RL32(p->buf)==0x800) {
int bsm_s, bsm_e, inb_s, inb_e, ch;
ch = p->buf[264];
bsm_s = p->buf[2048];
inb_s = p->buf[2048+1];
inb_e = p->buf[2048+210];
bsm_e = p->buf[2048+211];
if (ch != 1 && ch != 2)
return 0;
/* Check so that the redundant bsm bytes and info bytes are valid
* the block size mode bytes have to be the same
* the info bytes have to be the same
*/
if (bsm_s == bsm_e && inb_s == inb_e)
return AVPROBE_SCORE_MAX / 4 + 1;
}
return 0;
}
static int aea_read_header(AVFormatContext *s,
AVFormatParameters *ap)
{
AVStream *st = av_new_stream(s, 0);
if (!st)
return AVERROR(ENOMEM);
/* Parse the amount of channels and skip to pos 2048(0x800) */
avio_skip(s->pb, 264);
st->codec->channels = avio_r8(s->pb);
avio_skip(s->pb, 1783);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = CODEC_ID_ATRAC1;
st->codec->sample_rate = 44100;
st->codec->bit_rate = 292000;
if (st->codec->channels != 1 && st->codec->channels != 2) {
av_log(s,AV_LOG_ERROR,"Channels %d not supported!\n",st->codec->channels);
return -1;
}
st->codec->channel_layout = (st->codec->channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
st->codec->block_align = AT1_SU_SIZE * st->codec->channels;
return 0;
}
static int aea_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret = av_get_packet(s->pb, pkt, s->streams[0]->codec->block_align);
pkt->stream_index = 0;
if (ret <= 0)
return AVERROR(EIO);
return ret;
}
AVInputFormat ff_aea_demuxer = {
.name = "aea",
.long_name = NULL_IF_CONFIG_SMALL("MD STUDIO audio"),
.read_probe = aea_read_probe,
.read_header = aea_read_header,
.read_packet = aea_read_packet,
.read_seek = pcm_read_seek,
.flags= AVFMT_GENERIC_INDEX,
.extensions = "aea",
};