mirror of https://github.com/FFmpeg/FFmpeg.git
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/*
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* LOAS AudioSyncStream demuxer |
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* Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at> |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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#include "libavutil/intreadwrite.h" |
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#include "libavutil/internal.h" |
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#include "avformat.h" |
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#include "rawdec.h" |
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static int loas_probe(AVProbeData *p) |
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{ |
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int max_frames = 0, first_frames = 0; |
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int fsize, frames; |
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uint8_t *buf0 = p->buf; |
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uint8_t *buf2; |
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uint8_t *buf; |
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uint8_t *end = buf0 + p->buf_size - 3; |
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buf = buf0; |
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for(; buf < end; buf= buf2+1) { |
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buf2 = buf; |
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for(frames = 0; buf2 < end; frames++) { |
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uint32_t header = AV_RB24(buf2); |
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if((header >> 13) != 0x2B7) |
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break; |
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fsize = (header & 0x1FFF) + 3; |
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if(fsize < 7) |
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break; |
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fsize = FFMIN(fsize, end - buf2); |
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buf2 += fsize; |
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} |
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max_frames = FFMAX(max_frames, frames); |
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if(buf == buf0) |
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first_frames= frames; |
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} |
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if (first_frames>=3) return AVPROBE_SCORE_MAX/2+1; |
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else if(max_frames>100)return AVPROBE_SCORE_MAX/2; |
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else if(max_frames>=3) return AVPROBE_SCORE_MAX/4; |
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else if(max_frames>=1) return 1; |
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else return 0; |
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} |
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static int loas_read_header(AVFormatContext *s, |
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AVFormatParameters *ap) |
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{ |
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AVStream *st; |
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st = av_new_stream(s, 0); |
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if (!st) |
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return AVERROR(ENOMEM); |
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO; |
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st->codec->codec_id = s->iformat->value; |
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st->need_parsing = AVSTREAM_PARSE_FULL; |
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//LCM of all possible AAC sample rates
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av_set_pts_info(st, 64, 1, 28224000); |
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return 0; |
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} |
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AVInputFormat ff_loas_demuxer = { |
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.name = "loas", |
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.long_name = NULL_IF_CONFIG_SMALL("LOAS AudioSyncStream"), |
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.read_probe = loas_probe, |
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.read_header = loas_read_header, |
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.read_packet = ff_raw_read_partial_packet, |
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.flags= AVFMT_GENERIC_INDEX, |
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.value = CODEC_ID_AAC_LATM, |
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}; |
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