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FFmpeg/libavformat/aea.c
Derek Buitenhuis 6f69f7a8bf Merge commit '9200514ad8717c63f82101dc394f4378854325bf'
* commit '9200514ad8717c63f82101dc394f4378854325bf':
  lavf: replace AVStream.codec with AVStream.codecpar

This has been a HUGE effort from:
    - Derek Buitenhuis <derek.buitenhuis@gmail.com>
    - Hendrik Leppkes <h.leppkes@gmail.com>
    - wm4 <nfxjfg@googlemail.com>
    - Clément Bœsch <clement@stupeflix.com>
    - James Almer <jamrial@gmail.com>
    - Michael Niedermayer <michael@niedermayer.cc>
    - Rostislav Pehlivanov <atomnuker@gmail.com>

Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
2016-04-10 20:59:55 +01:00

111 lines
3.3 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 "libavutil/channel_layout.h"
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "pcm.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 ch, i;
ch = p->buf[264];
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
*/
for (i = 2048; i + 211 < p->buf_size; i+= 212) {
int bsm_s, bsm_e, inb_s, inb_e;
bsm_s = p->buf[0];
inb_s = p->buf[1];
inb_e = p->buf[210];
bsm_e = p->buf[211];
if (bsm_s != bsm_e || inb_s != inb_e)
return 0;
}
return AVPROBE_SCORE_MAX / 4 + 1;
}
return 0;
}
static int aea_read_header(AVFormatContext *s)
{
AVStream *st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
/* Parse the amount of channels and skip to pos 2048(0x800) */
avio_skip(s->pb, 264);
st->codecpar->channels = avio_r8(s->pb);
avio_skip(s->pb, 1783);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = AV_CODEC_ID_ATRAC1;
st->codecpar->sample_rate = 44100;
st->codecpar->bit_rate = 292000;
if (st->codecpar->channels != 1 && st->codecpar->channels != 2) {
av_log(s, AV_LOG_ERROR, "Channels %d not supported!\n", st->codecpar->channels);
return AVERROR_INVALIDDATA;
}
st->codecpar->channel_layout = (st->codecpar->channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
st->codecpar->block_align = AT1_SU_SIZE * st->codecpar->channels;
return 0;
}
static int aea_read_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret = av_get_packet(s->pb, pkt, s->streams[0]->codecpar->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 = ff_pcm_read_seek,
.flags = AVFMT_GENERIC_INDEX,
.extensions = "aea",
};