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  1. /*
  2.  * MD STUDIO audio demuxer
  3.  *
  4.  * Copyright (c) 2009 Benjamin Larsson
  5.  *
  6.  * This file is part of FFmpeg.
  7.  *
  8.  * FFmpeg is free software; you can redistribute it and/or
  9.  * modify it under the terms of the GNU Lesser General Public
  10.  * License as published by the Free Software Foundation; either
  11.  * version 2.1 of the License, or (at your option) any later version.
  12.  *
  13.  * FFmpeg is distributed in the hope that it will be useful,
  14.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  16.  * Lesser General Public License for more details.
  17.  *
  18.  * You should have received a copy of the GNU Lesser General Public
  19.  * License along with FFmpeg; if not, write to the Free Software
  20.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21.  */
  22.  
  23. #include "libavutil/channel_layout.h"
  24. #include "libavutil/intreadwrite.h"
  25. #include "avformat.h"
  26. #include "pcm.h"
  27.  
  28. #define AT1_SU_SIZE     212
  29.  
  30. static int aea_read_probe(AVProbeData *p)
  31. {
  32.     if (p->buf_size <= 2048+212)
  33.         return 0;
  34.  
  35.     /* Magic is '00 08 00 00' in Little Endian*/
  36.     if (AV_RL32(p->buf)==0x800) {
  37.         int ch, i;
  38.         ch = p->buf[264];
  39.  
  40.         if (ch != 1 && ch != 2)
  41.             return 0;
  42.  
  43.         /* Check so that the redundant bsm bytes and info bytes are valid
  44.          * the block size mode bytes have to be the same
  45.          * the info bytes have to be the same
  46.          */
  47.         for (i = 2048; i + 211 < p->buf_size; i+= 212) {
  48.             int bsm_s, bsm_e, inb_s, inb_e;
  49.             bsm_s = p->buf[0];
  50.             inb_s = p->buf[1];
  51.             inb_e = p->buf[210];
  52.             bsm_e = p->buf[211];
  53.  
  54.             if (bsm_s != bsm_e || inb_s != inb_e)
  55.                 return 0;
  56.         }
  57.         return AVPROBE_SCORE_MAX / 4 + 1;
  58.     }
  59.     return 0;
  60. }
  61.  
  62. static int aea_read_header(AVFormatContext *s)
  63. {
  64.     AVStream *st = avformat_new_stream(s, NULL);
  65.     if (!st)
  66.         return AVERROR(ENOMEM);
  67.  
  68.     /* Parse the amount of channels and skip to pos 2048(0x800) */
  69.     avio_skip(s->pb, 264);
  70.     st->codec->channels = avio_r8(s->pb);
  71.     avio_skip(s->pb, 1783);
  72.  
  73.  
  74.     st->codec->codec_type     = AVMEDIA_TYPE_AUDIO;
  75.     st->codec->codec_id       = AV_CODEC_ID_ATRAC1;
  76.     st->codec->sample_rate    = 44100;
  77.     st->codec->bit_rate       = 292000;
  78.  
  79.     if (st->codec->channels != 1 && st->codec->channels != 2) {
  80.         av_log(s,AV_LOG_ERROR,"Channels %d not supported!\n",st->codec->channels);
  81.         return AVERROR_INVALIDDATA;
  82.     }
  83.  
  84.     st->codec->channel_layout = (st->codec->channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
  85.  
  86.     st->codec->block_align = AT1_SU_SIZE * st->codec->channels;
  87.     return 0;
  88. }
  89.  
  90. static int aea_read_packet(AVFormatContext *s, AVPacket *pkt)
  91. {
  92.     int ret = av_get_packet(s->pb, pkt, s->streams[0]->codec->block_align);
  93.  
  94.     pkt->stream_index = 0;
  95.     if (ret <= 0)
  96.         return AVERROR(EIO);
  97.  
  98.     return ret;
  99. }
  100.  
  101. AVInputFormat ff_aea_demuxer = {
  102.     .name           = "aea",
  103.     .long_name      = NULL_IF_CONFIG_SMALL("MD STUDIO audio"),
  104.     .read_probe     = aea_read_probe,
  105.     .read_header    = aea_read_header,
  106.     .read_packet    = aea_read_packet,
  107.     .read_seek      = ff_pcm_read_seek,
  108.     .flags          = AVFMT_GENERIC_INDEX,
  109.     .extensions     = "aea",
  110. };
  111.