mirror of
https://github.com/FFmpeg/FFmpeg.git
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96 lines
2.9 KiB
C
96 lines
2.9 KiB
C
/*
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* MSF demuxer
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* Copyright (c) 2015 Paul B Mahol
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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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static int msf_probe(AVProbeData *p)
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{
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if (memcmp(p->buf, "MSF", 3))
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return 0;
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if (AV_RB32(p->buf+8) <= 0)
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return 0;
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if (AV_RB32(p->buf+16) <= 0)
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return 0;
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return AVPROBE_SCORE_MAX / 3 * 2;
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}
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static int msf_read_header(AVFormatContext *s)
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{
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unsigned codec, align, size;
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AVStream *st;
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avio_skip(s->pb, 4);
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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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codec = avio_rb32(s->pb);
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st->codec->channels = avio_rb32(s->pb);
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if (st->codec->channels <= 0)
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return AVERROR_INVALIDDATA;
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size = avio_rb32(s->pb);
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st->codec->sample_rate = avio_rb32(s->pb);
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if (st->codec->sample_rate <= 0)
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return AVERROR_INVALIDDATA;
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align = avio_rb32(s->pb) ;
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if (align > INT_MAX / st->codec->channels)
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return AVERROR_INVALIDDATA;
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st->codec->block_align = align;
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switch (codec) {
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case 0: st->codec->codec_id = AV_CODEC_ID_PCM_S16BE; break;
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case 3: st->codec->block_align = 16 * st->codec->channels;
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st->codec->codec_id = AV_CODEC_ID_ADPCM_PSX; break;
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case 7: st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
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st->codec->codec_id = AV_CODEC_ID_MP3; break;
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default:
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avpriv_request_sample(s, "Codec %d", codec);
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return AVERROR_PATCHWELCOME;
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}
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st->duration = av_get_audio_frame_duration(st->codec, size);
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avio_skip(s->pb, 0x40 - avio_tell(s->pb));
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avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
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return 0;
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}
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static int msf_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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AVCodecContext *codec = s->streams[0]->codec;
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return av_get_packet(s->pb, pkt, codec->block_align ? codec->block_align : 1024 * codec->channels);
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}
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AVInputFormat ff_msf_demuxer = {
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.name = "msf",
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.long_name = NULL_IF_CONFIG_SMALL("MSF"),
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.read_probe = msf_probe,
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.read_header = msf_read_header,
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.read_packet = msf_read_packet,
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.extensions = "msf",
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};
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