9d76cf0b18
* qatar/master: rtpdec: Templatize the code for different g726 bitrate variants rv40: move loop filter to rv34dsp context lavf: make av_set_pts_info private. rtpdec: Add support for G726 audio rtpdec: Add an init function that can do custom codec context initialization avconv: make copy_tb on by default. matroskadec: don't set codec timebase. rmdec: don't set codec timebase. avconv: compute next_pts from input packet duration when possible. lavf: estimate frame duration from r_frame_rate. avconv: update InputStream.pts in the streamcopy case. Conflicts: avconv.c libavdevice/alsa-audio-dec.c libavdevice/bktr.c libavdevice/fbdev.c libavdevice/libdc1394.c libavdevice/oss_audio.c libavdevice/v4l.c libavdevice/v4l2.c libavdevice/vfwcap.c libavdevice/x11grab.c libavformat/au.c libavformat/eacdata.c libavformat/flvdec.c libavformat/mpegts.c libavformat/mxfenc.c libavformat/rtpdec_g726.c libavformat/wtv.c libavformat/xmv.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
96 lines
2.8 KiB
C
96 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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AVFormatParameters *ap)
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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->value;
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st->need_parsing = AVSTREAM_PARSE_FULL;
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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"),
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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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.value = CODEC_ID_AAC,
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};
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