d586fa1e7c
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24981 a1c6a512-1295-4272-9138-f99709370657
122 lines
3.6 KiB
C
122 lines
3.6 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2010 Yoshihisa Uchida
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include <stdio.h>
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#include <string.h>
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#include <stdlib.h>
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#include <ctype.h>
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#include <inttypes.h>
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#include "system.h"
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#include "metadata.h"
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#include "metadata_common.h"
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#include "metadata_parsers.h"
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#include "logf.h"
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static const unsigned char bitspersamples[28] = {
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0,
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8, /* G.711 MULAW */
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8, /* 8bit */
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16, /* 16bit */
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24, /* 24bit */
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32, /* 32bit */
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32, /* 32bit */
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64, /* 64bit */
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0, /* Fragmented sample data */
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0, /* DSP program */
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0, /* 8bit fixed point */
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0, /* 16bit fixed point */
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0, /* 24bit fixed point */
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0, /* 32bit fixed point */
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0,
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0,
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0,
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0,
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0, /* 16bit linear with emphasis */
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0, /* 16bit linear compressed */
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0, /* 16bit linear with emphasis and compression */
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0, /* Music kit DSP commands */
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0,
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0, /* G.721 MULAW */
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0, /* G.722 */
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0, /* G.723 3bit */
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0, /* G.723 5bit */
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8, /* G.711 ALAW */
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};
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static int get_au_bitspersample(unsigned int encoding)
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{
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if (encoding > 27)
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return 0;
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return bitspersamples[encoding];
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}
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bool get_au_metadata(int fd, struct mp3entry* id3)
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{
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/* Use the trackname part of the id3 structure as a temporary buffer */
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unsigned char* buf = (unsigned char *)id3->path;
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unsigned long totalsamples = 0;
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unsigned long channels = 0;
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unsigned long bitspersample = 0;
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unsigned long numbytes = 0;
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int read_bytes;
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int offset;
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id3->vbr = false; /* All Sun audio files are CBR */
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id3->filesize = filesize(fd);
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if ((lseek(fd, 0, SEEK_SET) < 0) || ((read_bytes = read(fd, buf, 24)) < 0))
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return false;
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if (read_bytes < 24 || (memcmp(buf, ".snd", 4) != 0))
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{
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/* no header */
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numbytes = id3->filesize;
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bitspersample = 8;
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id3->frequency = 8000;
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channels = 1;
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}
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else
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{
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/* data offset */
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offset = get_long_be(buf + 4);
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if (offset < 24)
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{
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DEBUGF("CODEC_ERROR: sun audio offset size is small: %d\n", offset);
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return false;
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}
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/* data size */
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numbytes = get_long_be(buf + 8);
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if (numbytes == (uint32_t)0xffffffff)
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numbytes = id3->filesize - offset;
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/* bitspersample */
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bitspersample = get_au_bitspersample(get_long_be(buf + 12));
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/* sample rate */
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id3->frequency = get_long_be(buf + 16);
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channels = get_long_be(buf + 20);
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}
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totalsamples = numbytes / ((((bitspersample - 1) / 8) + 1) * channels);
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/* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
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id3->length = ((int64_t) totalsamples * 1000) / id3->frequency;
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return true;
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}
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