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https://github.com/xenia-project/FFmpeg.git
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d1dad7c824
* qatar/master: mpc8: return more meaningful error codes. mpc: return more meaningful error codes. wv,mpc8: don't return apetag data in packets. rtmp: do not warn about receiving metadata packets x86: h264dsp: Adjust YASM #ifdefs x86: yadif: Mark mmxext optimizations as such h264: convert loop filter strength dsp function to yasm. Improve descriptiveness of a number of codec and container long names Conflicts: libavcodec/flvdec.c libavcodec/libopenjpegdec.c libavformat/apetag.c libavformat/mp3dec.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
95 lines
2.8 KiB
C
95 lines
2.8 KiB
C
/*
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* raw ADTS AAC demuxer
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* Copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at>
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* Copyright (c) 2009 Robert Swain ( rob opendot cl )
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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 "avformat.h"
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#include "internal.h"
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#include "rawdec.h"
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#include "id3v1.h"
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static int adts_aac_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 - 7;
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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_RB16(buf2);
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if((header&0xFFF6) != 0xFFF0)
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break;
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fsize = (AV_RB32(buf2 + 3) >> 13) & 0x1FFF;
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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>500)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 adts_aac_read_header(AVFormatContext *s)
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{
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AVStream *st;
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st = avformat_new_stream(s, NULL);
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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->raw_codec_id;
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st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
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ff_id3v1_read(s);
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//LCM of all possible ADTS sample rates
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avpriv_set_pts_info(st, 64, 1, 28224000);
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return 0;
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}
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AVInputFormat ff_aac_demuxer = {
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.name = "aac",
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.long_name = NULL_IF_CONFIG_SMALL("raw ADTS AAC (Advanced Audio Coding)"),
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.read_probe = adts_aac_probe,
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.read_header = adts_aac_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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.extensions = "aac",
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.raw_codec_id = CODEC_ID_AAC,
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};
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