FFmpeg/libavformat/aeadec.c
Andreas Rheinhardt 7e41a658f5 avformat/aeadec: Use sample rate as time base
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-19 02:54:24 +01:00

114 lines
3.5 KiB
C

/*
* MD STUDIO audio demuxer
*
* Copyright (c) 2009 Benjamin Larsson
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/channel_layout.h"
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "avio_internal.h"
#include "demux.h"
#include "internal.h"
#include "pcm.h"
#define AT1_SU_SIZE 212
static int aea_read_probe(const AVProbeData *p)
{
if (p->buf_size <= 2048+AT1_SU_SIZE)
return 0;
/* Magic is '00 08 00 00' in little-endian*/
if (AV_RL32(p->buf)==0x800) {
int ch, block_size, score = 0;
ch = p->buf[264];
if (ch != 1 && ch != 2)
return 0;
block_size = ch * AT1_SU_SIZE;
/* Check so that the redundant bsm bytes and info bytes are valid
* the block size mode bytes have to be the same
* the info bytes have to be the same
*/
for (int i = 2048 + block_size; i + block_size <= p->buf_size; i += block_size) {
if (AV_RN16(p->buf+i) != AV_RN16(p->buf+i+AT1_SU_SIZE))
return 0;
score++;
}
return FFMIN(AVPROBE_SCORE_MAX / 4 + score, AVPROBE_SCORE_MAX);
}
return 0;
}
static int aea_read_header(AVFormatContext *s)
{
AVStream *st = avformat_new_stream(s, NULL);
char title[256 + 1];
int channels, ret;
if (!st)
return AVERROR(ENOMEM);
/* Read the title, parse the number of channels and skip to pos 2048(0x800) */
avio_rl32(s->pb); // magic
ret = ffio_read_size(s->pb, title, sizeof(title) - 1);
if (ret < 0)
return ret;
title[sizeof(title) - 1] = '\0';
if (title[0] != '\0')
av_dict_set(&st->metadata, "title", title, 0);
avio_rl32(s->pb); // Block count
channels = avio_r8(s->pb);
avio_skip(s->pb, 1783);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = AV_CODEC_ID_ATRAC1;
st->codecpar->sample_rate = 44100;
st->codecpar->bit_rate = 146000 * channels;
if (channels != 1 && channels != 2) {
av_log(s, AV_LOG_ERROR, "Channels %d not supported!\n", channels);
return AVERROR_INVALIDDATA;
}
av_channel_layout_default(&st->codecpar->ch_layout, channels);
st->codecpar->block_align = AT1_SU_SIZE * st->codecpar->ch_layout.nb_channels;
avpriv_set_pts_info(st, 64, 1, 44100);
return 0;
}
static int aea_read_packet(AVFormatContext *s, AVPacket *pkt)
{
return av_get_packet(s->pb, pkt, s->streams[0]->codecpar->block_align);
}
const FFInputFormat ff_aea_demuxer = {
.p.name = "aea",
.p.long_name = NULL_IF_CONFIG_SMALL("MD STUDIO audio"),
.p.flags = AVFMT_GENERIC_INDEX,
.p.extensions = "aea",
.read_probe = aea_read_probe,
.read_header = aea_read_header,
.read_packet = aea_read_packet,
.read_seek = ff_pcm_read_seek,
};