mirror of
https://github.com/FFmpeg/FFmpeg.git
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114 lines
3.5 KiB
C
114 lines
3.5 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 "avio_internal.h"
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#include "demux.h"
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#include "internal.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+AT1_SU_SIZE)
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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, block_size, score = 0;
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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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block_size = ch * AT1_SU_SIZE;
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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 (int i = 2048 + block_size; i + block_size <= p->buf_size; i += block_size) {
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if (AV_RN16(p->buf+i) != AV_RN16(p->buf+i+AT1_SU_SIZE))
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return 0;
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score++;
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}
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return FFMIN(AVPROBE_SCORE_MAX / 4 + score, AVPROBE_SCORE_MAX);
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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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char title[256 + 1];
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int channels, ret;
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if (!st)
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return AVERROR(ENOMEM);
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/* Read the title, parse the number of channels and skip to pos 2048(0x800) */
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avio_rl32(s->pb); // magic
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ret = ffio_read_size(s->pb, title, sizeof(title) - 1);
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if (ret < 0)
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return ret;
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title[sizeof(title) - 1] = '\0';
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if (title[0] != '\0')
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av_dict_set(&st->metadata, "title", title, 0);
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avio_rl32(s->pb); // Block count
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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 = 146000 * channels;
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if (channels != 1 && channels != 2) {
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av_log(s, AV_LOG_ERROR, "Channels %d not supported!\n", channels);
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return AVERROR_INVALIDDATA;
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}
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av_channel_layout_default(&st->codecpar->ch_layout, channels);
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st->codecpar->block_align = AT1_SU_SIZE * st->codecpar->ch_layout.nb_channels;
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avpriv_set_pts_info(st, 64, 1, 44100);
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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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return av_get_packet(s->pb, pkt, s->streams[0]->codecpar->block_align);
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}
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const FFInputFormat ff_aea_demuxer = {
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.p.name = "aea",
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.p.long_name = NULL_IF_CONFIG_SMALL("MD STUDIO audio"),
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.p.flags = AVFMT_GENERIC_INDEX,
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.p.extensions = "aea",
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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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};
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