/* * 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" #define LOAS_SYNC_WORD 0x2b7 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) != LOAS_SYNC_WORD) 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, };