/*
  * raw ADTS AAC demuxer
  * Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
  * Copyright (c) 2009 Robert Swain ( rob opendot cl )
  *
  * 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 "avformat.h"

 #include "internal.h"

 #include "rawdec.h"

 #include "id3v1.h"

 #include "apetag.h"

 
 static int adts_aac_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 - 7;

 
     buf = buf0;
 

     for (; buf < end; buf = buf2 + 1) {

         buf2 = buf;
 

         for (frames = 0; buf2 < end; frames++) {

             uint32_t header = AV_RB16(buf2);

             if ((header & 0xFFF6) != 0xFFF0) {
                 if (buf != buf0) {
                     // Found something that isn't an ADTS header, starting
                     // from a position other than the start of the buffer.
                     // Discard the count we've accumulated so far since it
                     // probably was a false positive.
                     frames = 0;
                 }

                 break;

             }

             fsize = (AV_RB32(buf2 + 3) >> 13) & 0x1FFF;

             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 if (max_frames >= 1)
         return 1;
     else
         return 0;

 }
 

 static int adts_aac_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;

 
     ff_id3v1_read(s);

     if (s->pb->seekable &&
         !av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX)) {
         int64_t cur = avio_tell(s->pb);
         ff_ape_parse_tag(s);
         avio_seek(s->pb, cur, SEEK_SET);
     }

     // LCM of all possible ADTS sample rates

     avpriv_set_pts_info(st, 64, 1, 28224000);

     return 0;
 }
 

 AVInputFormat ff_aac_demuxer = {

     .name         = "aac",
     .long_name    = NULL_IF_CONFIG_SMALL("raw ADTS AAC (Advanced Audio Coding)"),
     .read_probe   = adts_aac_probe,
     .read_header  = adts_aac_read_header,
     .read_packet  = ff_raw_read_partial_packet,
     .flags        = AVFMT_GENERIC_INDEX,
     .extensions   = "aac",

     .mime_type    = "audio/aac,audio/aacp,audio/x-aac",

     .raw_codec_id = AV_CODEC_ID_AAC,

 };