/* * 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" 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) 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_MAX/2+1; else if(max_frames>500)return AVPROBE_SCORE_MAX/2; else if(max_frames>=3) return AVPROBE_SCORE_MAX/4; 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); //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", .raw_codec_id = AV_CODEC_ID_AAC, };