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https://github.com/FFmpeg/FFmpeg.git
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737eb5976f
It is pretty hopeless that other considerable projects will adopt libavutil alone in other projects. Projects that need small footprint are better off with more specialized libraries such as gnulib or rather just copy the necessary parts that they need. With this in mind, nobody is helped by having libavutil and libavcore split. In order to ease maintenance inside and around FFmpeg and to reduce confusion where to put common code, avcore's functionality is merged (back) to avutil. Signed-off-by: Reinhard Tartler <siretart@tauware.de>
110 lines
3.0 KiB
C
110 lines
3.0 KiB
C
/*
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* MD STUDIO audio demuxer
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*
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* Copyright (c) 2009 Benjamin Larsson
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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 "avformat.h"
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#include "pcm.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/audioconvert.h"
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#define AT1_SU_SIZE 212
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static int aea_read_probe(AVProbeData *p)
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{
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if (p->buf_size <= 2048+212)
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return 0;
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/* Magic is '00 08 00 00' in Little Endian*/
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if (AV_RL32(p->buf)==0x800) {
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int bsm_s, bsm_e, inb_s, inb_e, ch;
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ch = p->buf[264];
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bsm_s = p->buf[2048];
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inb_s = p->buf[2048+1];
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inb_e = p->buf[2048+210];
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bsm_e = p->buf[2048+211];
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if (ch != 1 && ch != 2)
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return 0;
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/* Check so that the redundant bsm bytes and info bytes are valid
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* the block size mode bytes have to be the same
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* the info bytes have to be the same
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*/
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if (bsm_s == bsm_e && inb_s == inb_e)
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return AVPROBE_SCORE_MAX / 4 + 1;
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}
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return 0;
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}
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static int aea_read_header(AVFormatContext *s,
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AVFormatParameters *ap)
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{
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AVStream *st = av_new_stream(s, 0);
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if (!st)
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return AVERROR(ENOMEM);
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/* Parse the amount of channels and skip to pos 2048(0x800) */
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url_fskip(s->pb, 264);
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st->codec->channels = get_byte(s->pb);
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url_fskip(s->pb, 1783);
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = CODEC_ID_ATRAC1;
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st->codec->sample_rate = 44100;
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st->codec->bit_rate = 292000;
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if (st->codec->channels != 1 && st->codec->channels != 2) {
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av_log(s,AV_LOG_ERROR,"Channels %d not supported!\n",st->codec->channels);
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return -1;
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}
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st->codec->channel_layout = (st->codec->channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
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st->codec->block_align = AT1_SU_SIZE * st->codec->channels;
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return 0;
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}
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static int aea_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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int ret = av_get_packet(s->pb, pkt, s->streams[0]->codec->block_align);
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pkt->stream_index = 0;
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if (ret <= 0)
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return AVERROR(EIO);
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return ret;
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}
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AVInputFormat ff_aea_demuxer = {
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"aea",
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NULL_IF_CONFIG_SMALL("MD STUDIO audio"),
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0,
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aea_read_probe,
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aea_read_header,
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aea_read_packet,
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0,
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pcm_read_seek,
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.flags= AVFMT_GENERIC_INDEX,
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.extensions = "aea",
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};
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