rockbox/apps/metadata/au.c
Nils Wallménius d586fa1e7c Mark array as const
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24981 a1c6a512-1295-4272-9138-f99709370657
2010-03-01 08:41:57 +00:00

122 lines
3.6 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2010 Yoshihisa Uchida
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <ctype.h>
#include <inttypes.h>
#include "system.h"
#include "metadata.h"
#include "metadata_common.h"
#include "metadata_parsers.h"
#include "logf.h"
static const unsigned char bitspersamples[28] = {
0,
8, /* G.711 MULAW */
8, /* 8bit */
16, /* 16bit */
24, /* 24bit */
32, /* 32bit */
32, /* 32bit */
64, /* 64bit */
0, /* Fragmented sample data */
0, /* DSP program */
0, /* 8bit fixed point */
0, /* 16bit fixed point */
0, /* 24bit fixed point */
0, /* 32bit fixed point */
0,
0,
0,
0,
0, /* 16bit linear with emphasis */
0, /* 16bit linear compressed */
0, /* 16bit linear with emphasis and compression */
0, /* Music kit DSP commands */
0,
0, /* G.721 MULAW */
0, /* G.722 */
0, /* G.723 3bit */
0, /* G.723 5bit */
8, /* G.711 ALAW */
};
static int get_au_bitspersample(unsigned int encoding)
{
if (encoding > 27)
return 0;
return bitspersamples[encoding];
}
bool get_au_metadata(int fd, struct mp3entry* id3)
{
/* Use the trackname part of the id3 structure as a temporary buffer */
unsigned char* buf = (unsigned char *)id3->path;
unsigned long totalsamples = 0;
unsigned long channels = 0;
unsigned long bitspersample = 0;
unsigned long numbytes = 0;
int read_bytes;
int offset;
id3->vbr = false; /* All Sun audio files are CBR */
id3->filesize = filesize(fd);
if ((lseek(fd, 0, SEEK_SET) < 0) || ((read_bytes = read(fd, buf, 24)) < 0))
return false;
if (read_bytes < 24 || (memcmp(buf, ".snd", 4) != 0))
{
/* no header */
numbytes = id3->filesize;
bitspersample = 8;
id3->frequency = 8000;
channels = 1;
}
else
{
/* data offset */
offset = get_long_be(buf + 4);
if (offset < 24)
{
DEBUGF("CODEC_ERROR: sun audio offset size is small: %d\n", offset);
return false;
}
/* data size */
numbytes = get_long_be(buf + 8);
if (numbytes == (uint32_t)0xffffffff)
numbytes = id3->filesize - offset;
/* bitspersample */
bitspersample = get_au_bitspersample(get_long_be(buf + 12));
/* sample rate */
id3->frequency = get_long_be(buf + 16);
channels = get_long_be(buf + 20);
}
totalsamples = numbytes / ((((bitspersample - 1) / 8) + 1) * channels);
/* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
id3->length = ((int64_t) totalsamples * 1000) / id3->frequency;
return true;
}