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https://github.com/FFmpeg/FFmpeg.git
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40bdd8cc05
Do this by allocating AVStream together with the data that is currently in AVStreamInternal; or rather: Put AVStream at the beginning of a new structure called FFStream (which encompasses more than just the internal fields and is a proper context in its own right, hence the name) and remove AVStreamInternal altogether. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
97 lines
2.8 KiB
C
97 lines
2.8 KiB
C
/*
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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 "internal.h"
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#include "rawdec.h"
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#define LOAS_SYNC_WORD 0x2b7
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static int loas_probe(const 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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const uint8_t *buf0 = p->buf;
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const uint8_t *buf2;
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const uint8_t *buf;
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const 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) != LOAS_SYNC_WORD)
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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)
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return AVPROBE_SCORE_EXTENSION + 1;
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else if (max_frames > 100)
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return AVPROBE_SCORE_EXTENSION;
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else if (max_frames >= 3)
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return AVPROBE_SCORE_EXTENSION / 2;
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else
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return 0;
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}
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static int loas_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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->codec_id = s->iformat->raw_codec_id;
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ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
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//LCM of all possible AAC 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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const 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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.raw_codec_id = AV_CODEC_ID_AAC_LATM,
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.priv_data_size = sizeof(FFRawDemuxerContext),
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.priv_class = &ff_raw_demuxer_class,
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};
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