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https://github.com/FFmpeg/FFmpeg.git
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f6d1b18b3d
This allows testing parsers with a wider range of input packet sizes. Which is important and usefull for regression testing, some of our parsers in fact to not work if the packet size is changed from 1024 Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
98 lines
2.9 KiB
C
98 lines
2.9 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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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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FF_RAW_DEMUXER_CLASS(loas)
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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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.raw_codec_id = AV_CODEC_ID_AAC_LATM,
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.priv_data_size = sizeof(FFRawDemuxerContext),
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.priv_class = &loas_demuxer_class,
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};
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