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- /*
- * 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;
- uint8_t *buf0 = p->buf;
- uint8_t *buf2;
- uint8_t *buf;
- 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_MAX/2+1;
- else if(max_frames>100)return AVPROBE_SCORE_MAX/2;
- else if(max_frames>=3) return AVPROBE_SCORE_MAX/4;
- else return 0;
- }
- static int loas_read_header(AVFormatContext *s,
- AVFormatParameters *ap)
- {
- 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->value;
- st->need_parsing = AVSTREAM_PARSE_FULL;
- //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,
- .value = CODEC_ID_AAC_LATM,
- };
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