/*
  * LOAS AudioSyncStream demuxer
  * Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
  *
  * 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/intreadwrite.h"
 #include "libavutil/internal.h"
 #include "avformat.h"

 #include "internal.h"

 #include "rawdec.h"
 
 static int loas_probe(AVProbeData *p)
 {
     int max_frames = 0, first_frames = 0;
     int fsize, frames;

     const uint8_t *buf0 = p->buf;
     const uint8_t *buf2;
     const uint8_t *buf;
     const uint8_t *end = buf0 + p->buf_size - 3;

     buf = buf0;
 
     for(; buf < end; buf= buf2+1) {
         buf2 = buf;
 
         for(frames = 0; buf2 < end; frames++) {
             uint32_t header = AV_RB24(buf2);
             if((header >> 13) != 0x2B7)
                 break;
             fsize = (header & 0x1FFF) + 3;
             if(fsize < 7)
                 break;
             fsize = FFMIN(fsize, end - buf2);
             buf2 += fsize;
         }
         max_frames = FFMAX(max_frames, frames);
         if(buf == buf0)
             first_frames= frames;
     }

     if   (first_frames>=3) return AVPROBE_SCORE_EXTENSION+1;
     else if(max_frames>100)return AVPROBE_SCORE_EXTENSION;
     else if(max_frames>=3) return AVPROBE_SCORE_EXTENSION / 2;

     else                   return 0;
 }
 

 static int loas_read_header(AVFormatContext *s)

 {
     AVStream *st;
 

     st = avformat_new_stream(s, NULL);

     if (!st)
         return AVERROR(ENOMEM);
 
     st->codec->codec_type = AVMEDIA_TYPE_AUDIO;

     st->codec->codec_id = s->iformat->raw_codec_id;

     st->need_parsing = AVSTREAM_PARSE_FULL_RAW;

 
     //LCM of all possible AAC sample rates

     avpriv_set_pts_info(st, 64, 1, 28224000);

 
     return 0;
 }
 
 AVInputFormat ff_loas_demuxer = {
     .name           = "loas",
     .long_name      = NULL_IF_CONFIG_SMALL("LOAS AudioSyncStream"),
     .read_probe     = loas_probe,
     .read_header    = loas_read_header,
     .read_packet    = ff_raw_read_partial_packet,
     .flags= AVFMT_GENERIC_INDEX,

     .raw_codec_id = AV_CODEC_ID_AAC_LATM,

 };