This is possible now that the next-API is gone. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com> Signed-off-by: James Almer <jamrial@gmail.com>
		
			
				
	
	
		
			111 lines
		
	
	
		
			3.3 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			111 lines
		
	
	
		
			3.3 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 "libavutil/channel_layout.h"
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "pcm.h"
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#define AT1_SU_SIZE     212
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static int aea_read_probe(const 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 ch, i;
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        ch = p->buf[264];
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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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        for (i = 2048; i + 211 < p->buf_size; i+= 212) {
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            int bsm_s, bsm_e, inb_s, inb_e;
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            bsm_s = p->buf[0];
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            inb_s = p->buf[1];
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            inb_e = p->buf[210];
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            bsm_e = p->buf[211];
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            if (bsm_s != bsm_e || inb_s != inb_e)
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                return 0;
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        }
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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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{
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    AVStream *st = avformat_new_stream(s, NULL);
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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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    avio_skip(s->pb, 264);
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    st->codecpar->channels = avio_r8(s->pb);
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    avio_skip(s->pb, 1783);
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    st->codecpar->codec_type     = AVMEDIA_TYPE_AUDIO;
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    st->codecpar->codec_id       = AV_CODEC_ID_ATRAC1;
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    st->codecpar->sample_rate    = 44100;
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    st->codecpar->bit_rate       = 292000;
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    if (st->codecpar->channels != 1 && st->codecpar->channels != 2) {
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        av_log(s, AV_LOG_ERROR, "Channels %d not supported!\n", st->codecpar->channels);
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        return AVERROR_INVALIDDATA;
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    }
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    st->codecpar->channel_layout = (st->codecpar->channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
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    st->codecpar->block_align = AT1_SU_SIZE * st->codecpar->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]->codecpar->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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const AVInputFormat ff_aea_demuxer = {
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    .name           = "aea",
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    .long_name      = NULL_IF_CONFIG_SMALL("MD STUDIO audio"),
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    .read_probe     = aea_read_probe,
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    .read_header    = aea_read_header,
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    .read_packet    = aea_read_packet,
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    .read_seek      = ff_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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