add SMAF codec (.mmf extension)(FS#10432)

This codec supports only wave data (ADPCM and PCM).
It does not support MIDI, picture, and movie.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24878 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Yoshihisa Uchida 2010-02-24 11:46:29 +00:00
parent aa58715a54
commit 45e009a364
9 changed files with 821 additions and 1 deletions

View File

@ -186,6 +186,7 @@ metadata/asap.c
metadata/rm.c
metadata/nsf.c
metadata/oma.c
metadata/smaf.c
#endif
#ifdef HAVE_TAGCACHE
tagcache.c

View File

@ -27,6 +27,7 @@ shorten.c
aiff.c
speex.c
adx.c
smaf.c
#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
/* encoders */
aiff_enc.c

View File

@ -90,6 +90,7 @@ $(CODECDIR)/atrac3_rm.codec : $(CODECDIR)/libatrac.a $(CODECDIR)/librm.a
$(CODECDIR)/atrac3_oma.codec : $(CODECDIR)/libatrac.a
$(CODECDIR)/aiff.codec : $(CODECDIR)/libpcm.a
$(CODECDIR)/wav.codec : $(CODECDIR)/libpcm.a
$(CODECDIR)/smaf.codec : $(CODECDIR)/libpcm.a
$(CODECS): $(CODECLIB) # this must be last in codec dependency list

435
apps/codecs/smaf.c Normal file
View File

@ -0,0 +1,435 @@
/***************************************************************************
* __________ __ ___.
* 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 "codeclib.h"
#include "codecs/libpcm/support_formats.h"
CODEC_HEADER
/*
* SMAF (Synthetic music Mobile Application Format)
*
* References
* [1] YAMAHA Corporation, Synthetic music Mobile Application Format Ver.3.05, 2002
*/
enum {
SMAF_TRACK_CHUNK_SCORE = 0, /* Score Track */
SMAF_TRACK_CHUNK_AUDIO, /* PCM Audio Track */
};
/* SMAF supported codec formats */
enum {
SMAF_FORMAT_UNSUPPORT = 0, /* unsupported format */
SMAF_FORMAT_SIGNED_PCM, /* 2's complement PCM */
SMAF_FORMAT_UNSIGNED_PCM, /* Offset Binary PCM */
SMAF_FORMAT_ADPCM, /* YAMAHA ADPCM */
};
static int support_formats[2][3] = {
{SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_UNSIGNED_PCM, SMAF_FORMAT_ADPCM },
{SMAF_FORMAT_SIGNED_PCM, SMAF_FORMAT_ADPCM, SMAF_FORMAT_UNSUPPORT },
};
static const struct pcm_entry pcm_codecs[] = {
{ SMAF_FORMAT_SIGNED_PCM, get_linear_pcm_codec },
{ SMAF_FORMAT_UNSIGNED_PCM, get_linear_pcm_codec },
{ SMAF_FORMAT_ADPCM, get_yamaha_adpcm_codec },
};
#define NUM_FORMATS 3
static int basebits[4] = { 4, 8, 12, 16 };
#define PCM_SAMPLE_SIZE (2048*2)
static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
static const struct pcm_codec *get_codec(uint32_t formattag)
{
int i;
for (i = 0; i < NUM_FORMATS; i++)
{
if (pcm_codecs[i].format_tag == formattag)
{
if (pcm_codecs[i].get_codec)
return pcm_codecs[i].get_codec();
return 0;
}
}
return 0;
}
static unsigned int get_be32(uint8_t *buf)
{
return (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3];
}
static int convert_smaf_audio_format(int track_chunk, unsigned int audio_format)
{
if (audio_format > 3)
return SMAF_FORMAT_UNSUPPORT;
return support_formats[track_chunk][audio_format];
}
static int convert_smaf_audio_basebit(unsigned int basebit)
{
if (basebit > 4)
return 0;
return basebits[basebit];
}
static bool parse_audio_track(struct pcm_format *fmt,
unsigned char **stbuf, unsigned char *endbuf)
{
unsigned char *buf = *stbuf;
int chunksize;
buf += 8;
fmt->channels = ((buf[2] & 0x80) >> 7) + 1;
fmt->formattag = convert_smaf_audio_format(SMAF_TRACK_CHUNK_AUDIO,
(buf[2] >> 4) & 0x07);
if (fmt->formattag == SMAF_FORMAT_UNSUPPORT)
{
DEBUGF("CODEC_ERROR: unsupport pcm data format : %d\n", (buf[2] >> 4) & 0x07);
return false;
}
fmt->bitspersample = convert_smaf_audio_basebit(buf[3] >> 4);
if (fmt->bitspersample == 0)
{
DEBUGF("CODEC_ERROR: unsupport pcm data basebit : %d\n", buf[3] >> 4);
return false;
}
buf += 6;
while (buf < endbuf)
{
chunksize = get_be32(buf + 4) + 8;
if (memcmp(buf, "Awa", 3) == 0)
{
fmt->numbytes = get_be32(buf + 4);
buf += 8;
return true;
}
buf += chunksize;
}
DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n");
return false;
}
static bool parse_score_track(struct pcm_format *fmt,
unsigned char **stbuf, unsigned char *endbuf)
{
unsigned char *buf = *stbuf;
int chunksize;
if (buf[9] != 0x00)
{
DEBUGF("CODEC_ERROR: score track chunk unsupport sequence type %d\n", buf[9]);
return false;
}
/*
* skip to the next chunk.
* MA-2/MA-3/MA-5: padding 16 bytes
* MA-7: padding 32 bytes
*/
if (buf[3] < 7)
buf += 28;
else
buf += 44;
while (buf < endbuf)
{
chunksize = get_be32(buf + 4) + 8;
if (memcmp(buf, "Mtsp", 4) == 0)
{
buf += 8;
if (memcmp(buf, "Mwa", 3) != 0)
{
DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n");
return false;
}
fmt->numbytes = get_be32(buf + 4) - 3;
fmt->channels = ((buf[8] & 0x80) >> 7) + 1;
fmt->formattag = convert_smaf_audio_format(SMAF_TRACK_CHUNK_SCORE,
(buf[8] >> 4) & 0x07);
if (fmt->formattag == SMAF_FORMAT_UNSUPPORT)
{
DEBUGF("CODEC_ERROR: unsupport pcm data format : %d\n",
(buf[8] >> 4) & 0x07);
return false;
}
fmt->bitspersample = convert_smaf_audio_basebit(buf[8] & 0x0f);
if (fmt->bitspersample == 0)
{
DEBUGF("CODEC_ERROR: unsupport pcm data basebit : %d\n",
buf[8] & 0x0f);
return false;
}
buf += 11;
return true;
}
buf += chunksize;
}
DEBUGF("CODEC_ERROR: smaf does not include stream pcm data\n");
return false;
}
static bool parse_header(struct pcm_format *fmt, size_t *pos)
{
unsigned char *buf, *stbuf, *endbuf;
size_t chunksize;
ci->memset(fmt, 0, sizeof(struct pcm_format));
/* assume the SMAF pcm data position is less than 1024 bytes */
stbuf = ci->request_buffer(&chunksize, 1024);
if (chunksize < 1024)
return false;
buf = stbuf;
endbuf = stbuf + chunksize;
if (memcmp(buf, "MMMD", 4) != 0)
{
DEBUGF("CODEC_ERROR: does not smaf format %c%c%c%c\n",
buf[0], buf[1], buf[2], buf[3]);
return false;
}
buf += 8;
while (buf < endbuf)
{
chunksize = get_be32(buf + 4) + 8;
if (memcmp(buf, "ATR", 3) == 0)
{
if (!parse_audio_track(fmt, &buf, endbuf))
return false;
break;
}
if (memcmp(buf, "MTR", 3) == 0)
{
if (!parse_score_track(fmt, &buf, endbuf))
return false;
break;
}
buf += chunksize;
}
if (buf >= endbuf)
{
DEBUGF("CODEC_ERROR: unsupported smaf format\n");
return false;
}
/* blockalign */
if (fmt->formattag == SMAF_FORMAT_SIGNED_PCM ||
fmt->formattag == SMAF_FORMAT_UNSIGNED_PCM)
fmt->blockalign = fmt->channels * fmt->bitspersample >> 3;
/* data signess (default signed) */
fmt->is_signed = (fmt->formattag != SMAF_FORMAT_UNSIGNED_PCM);
fmt->is_little_endian = false;
/* sets pcm data position */
*pos = buf - stbuf;
return true;
}
static struct pcm_format format;
static uint32_t bytesdone;
static uint8_t *read_buffer(size_t *realsize)
{
uint8_t *buffer = (uint8_t *)ci->request_buffer(realsize, format.chunksize);
if (bytesdone + (*realsize) > format.numbytes)
*realsize = format.numbytes - bytesdone;
bytesdone += *realsize;
ci->advance_buffer(*realsize);
return buffer;
}
enum codec_status codec_main(void)
{
int status = CODEC_OK;
uint32_t decodedsamples;
uint32_t i = CODEC_OK;
size_t n;
int bufcount;
int endofstream;
uint8_t *smafbuf;
off_t firstblockposn; /* position of the first block in file */
const struct pcm_codec *codec;
/* Generic codec initialisation */
ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
next_track:
if (codec_init()) {
i = CODEC_ERROR;
goto exit;
}
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
codec_set_replaygain(ci->id3);
ci->memset(&format, 0, sizeof(struct pcm_format));
format.is_signed = true;
format.is_little_endian = false;
decodedsamples = 0;
codec = 0;
if (!parse_header(&format, &n))
{
i = CODEC_ERROR;
goto done;
}
codec = get_codec(format.formattag);
if (codec == 0)
{
DEBUGF("CODEC_ERROR: unsupport audio format: 0x%lx\n", format.formattag);
i = CODEC_ERROR;
goto done;
}
if (!codec->set_format(&format))
{
i = CODEC_ERROR;
goto done;
}
/* common format check */
if (format.channels == 0) {
DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-channels file\n");
status = CODEC_ERROR;
goto done;
}
if (format.samplesperblock == 0) {
DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-wSamplesPerBlock file\n");
status = CODEC_ERROR;
goto done;
}
if (format.blockalign == 0)
{
DEBUGF("CODEC_ERROR: 'fmt ' chunk not found or 0-blockalign file\n");
i = CODEC_ERROR;
goto done;
}
if (format.numbytes == 0) {
DEBUGF("CODEC_ERROR: 'data' chunk not found or has zero-length\n");
status = CODEC_ERROR;
goto done;
}
/* check chunksize */
if ((format.chunksize / format.blockalign) * format.samplesperblock * format.channels
> PCM_SAMPLE_SIZE)
format.chunksize = (PCM_SAMPLE_SIZE / format.blockalign) * format.blockalign;
if (format.chunksize == 0)
{
DEBUGF("CODEC_ERROR: chunksize is 0\n");
i = CODEC_ERROR;
goto done;
}
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
if (format.channels == 2) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
} else if (format.channels == 1) {
ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
} else {
DEBUGF("CODEC_ERROR: more than 2 channels unsupported\n");
i = CODEC_ERROR;
goto done;
}
firstblockposn = 1024 - n;
ci->advance_buffer(firstblockposn);
/* The main decoder loop */
bytesdone = 0;
ci->set_elapsed(0);
endofstream = 0;
while (!endofstream) {
ci->yield();
if (ci->stop_codec || ci->new_track)
break;
if (ci->seek_time) {
struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, &read_buffer);
decodedsamples = newpos->samples;
if (newpos->pos > format.numbytes)
break;
if (ci->seek_buffer(firstblockposn + newpos->pos))
{
bytesdone = newpos->pos;
}
ci->seek_complete();
}
smafbuf = (uint8_t *)ci->request_buffer(&n, format.chunksize);
if (n == 0)
break; /* End of stream */
if (bytesdone + n > format.numbytes) {
n = format.numbytes - bytesdone;
endofstream = 1;
}
status = codec->decode(smafbuf, n, samples, &bufcount);
if (status == CODEC_ERROR)
{
DEBUGF("codec error\n");
goto done;
}
ci->pcmbuf_insert(samples, NULL, bufcount);
ci->advance_buffer(n);
bytesdone += n;
decodedsamples += bufcount;
if (bytesdone >= format.numbytes)
endofstream = 1;
ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
}
i = CODEC_OK;
done:
if (ci->request_next_track())
goto next_track;
exit:
return i;
}

View File

@ -101,6 +101,7 @@ static const struct filetype inbuilt_filetypes[] = {
{ "oma", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
{ "aa3", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
{ "at3", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
{ "mmf", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
#endif
{ "m3u", FILE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST },
{ "m3u8",FILE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST },

View File

@ -165,6 +165,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
/* Atrac3 in Sony OMA Container */
[AFMT_OMA_ATRAC3] =
AFMT_ENTRY("ATRAC3", "atrac3_oma", NULL, "oma\0aa3\0" ),
/* SMAF (Synthetic music Mobile Application Format) */
[AFMT_SMAF] =
AFMT_ENTRY("SMAF", "smaf", NULL, "mmf\0" ),
#endif
};
@ -447,6 +450,14 @@ bool get_metadata(struct mp3entry* id3, int fd, const char* trackname)
return false;
}
break;
case AFMT_SMAF:
if (!get_smaf_metadata(fd, id3))
{
DEBUGF("get_smaf_metadata error\n");
return false;
}
break;
#endif /* CONFIG_CODEC == SWCODEC */

View File

@ -78,6 +78,7 @@ enum
AFMT_TM8, /* Atari 8bit tm8 format */
AFMT_TM2, /* Atari 8bit tm2 format */
AFMT_OMA_ATRAC3, /* Atrac3 in Sony OMA container */
AFMT_SMAF, /* SMAF */
#endif
/* add new formats at any index above this line to have a sensible order -

View File

@ -42,4 +42,4 @@ bool get_asap_metadata(int fd, struct mp3entry* id3);
bool get_rm_metadata(int fd, struct mp3entry* id3);
bool get_nsf_metadata(int fd, struct mp3entry* id3);
bool get_oma_metadata(int fd, struct mp3entry* id3);
bool get_smaf_metadata(int fd, struct mp3entry* id3);

369
apps/metadata/smaf.c Normal file
View File

@ -0,0 +1,369 @@
/***************************************************************************
* __________ __ ___.
* 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 "rbunicode.h"
#include "logf.h"
static int basebits[4] = { 4, 8, 12, 16 };
static int frequency[5] = { 4000, 8000, 11025, 22050, 44100 };
static int support_codepages[7] = {
SJIS, ISO_8859_1, -1, GB_2312, BIG_5, -1, -1,
};
/* extra codepage */
#define UCS_2 (NUM_CODEPAGES + 1)
#define UTF_16 (NUM_CODEPAGES + 2)
/* support id3 tag */
#define TAG_TITLE (('S'<<8)|'T')
#define TAG_ARTIST (('A'<<8)|'N')
#define TAG_COMPOSER (('S'<<8)|'W')
static unsigned char smafbuf[1024];
static bool read_datachunk(unsigned char *src, int size, unsigned short tag,
int codepage, unsigned char *dst)
{
int datasize = 0;
unsigned char *utf8;
while(size > datasize + 4)
{
datasize = (src[2] << 8) | src[3];
if (tag == ((src[0] << 8) | src[1]))
{
src += 4;
if (codepage < NUM_CODEPAGES)
utf8 = iso_decode(src, dst, codepage, datasize);
else /* codepage == UTF_16, UCS_2 */
utf8 = utf16BEdecode(src, dst, datasize);
*utf8 = '\0';
return true;
}
src += (datasize + 4);
}
return false;
}
static bool read_option(unsigned char *src, int size, unsigned short tag,
int codepage, unsigned char *dst)
{
int datasize = 0;
unsigned char *endsrc = src + size;
unsigned char *utf8;
while(src < endsrc)
{
utf8 = src;
src += 3;
datasize = 0;
while (*src != ',' || *(src-1) == '\\')
{
datasize++;
src++;
}
if (tag == ((utf8[0] << 8) | utf8[1]) && utf8[2] == ':')
{
utf8 += 3;
if (codepage < NUM_CODEPAGES)
utf8 = iso_decode(utf8, dst, codepage, datasize);
else /* codepage == UTF_16, UCS_2 */
utf8 = utf16BEdecode(utf8, dst, datasize);
*utf8 = '\0';
return true;
}
src++;
}
return false;
}
static int convert_smaf_audio_frequency(unsigned int freq)
{
if (freq > 4)
return 0;
return frequency[freq];
}
static int convert_smaf_audio_basebit(unsigned int basebit)
{
if (basebit > 4)
return 0;
return basebits[basebit];
}
static int convert_smaf_codetype(unsigned int codetype)
{
if (codetype < 7)
return support_codepages[codetype];
else if (codetype < 0x20)
return -1;
else if (codetype == 0x20)
return UCS_2;
else if (codetype == 0x23)
return UTF_8;
else if (codetype == 0x24)
return UTF_16;
else if (codetype == 0xff)
return ISO_8859_1;
return -1;
}
static bool get_smaf_metadata_audio_track(struct mp3entry *id3,
unsigned char* buf, unsigned char *endbuf)
{
int bitspersample;
int channels;
int chunksize;
long numbytes;
unsigned long totalsamples;
channels = ((buf[10] & 0x80) >> 7) + 1;
bitspersample = convert_smaf_audio_basebit(buf[11] >> 4);
if (bitspersample == 0)
{
DEBUGF("metada error: smaf unsupport basebit %d\n", buf[11] >> 4);
return false;
}
id3->frequency = convert_smaf_audio_frequency(buf[10] & 0x0f);
buf += 14;
while (buf < endbuf)
{
chunksize = get_long_be(buf + 4) + 8;
if (memcmp(buf, "Awa", 3) == 0)
{
numbytes = get_long_be(buf + 4) - 3;
totalsamples = (numbytes << 3) / (bitspersample * channels);
/* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
id3->length = ((int64_t)totalsamples * 1000LL) / id3->frequency;
return true;
}
buf += chunksize;
}
DEBUGF("metada error: smaf does not include pcm audio data\n");
return false;
}
static bool get_smaf_metadata_score_track(struct mp3entry *id3,
unsigned char* buf, unsigned char *endbuf)
{
int bitspersample;
int channels;
int chunksize;
long numbytes;
unsigned long totalsamples;
/*
* skip to the next chunk.
* MA-2/MA-3/MA-5: padding 16 bytes
* MA-7: padding 32 bytes
*/
if (buf[3] < 7)
buf += 28;
else
buf += 44;
while (buf + 10 < endbuf)
{
chunksize = get_long_be(buf + 4) + 8;
if (memcmp(buf, "Mtsp", 4) == 0)
{
buf += 8;
if (memcmp(buf, "Mwa", 3) != 0)
{
DEBUGF("metada error: smaf unsupport format: %c%c%c%c\n",
buf[0], buf[1], buf[2], buf[3]);
return false;
}
channels = ((buf[8] & 0x80) >> 7) + 1;
bitspersample = convert_smaf_audio_basebit(buf[8] & 0x0f);
if (bitspersample == 0)
{
DEBUGF("metada error: smaf unsupport basebit %d\n", buf[8] & 0x0f);
return false;
}
numbytes = get_long_be(buf + 4) - 3;
totalsamples = numbytes * 8 / (bitspersample * channels);
id3->frequency = (buf[9] << 8) | buf[10];
/* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */
id3->length = ((int64_t) totalsamples * 1000) / id3->frequency;
return true;
}
buf += chunksize;
}
DEBUGF("metada error: smaf does not include pcm audio data\n");
return false;
}
bool get_smaf_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 char *endbuf = smafbuf + 1024;
int i;
int contents_size;
int codepage = ISO_8859_1;
id3->title = NULL;
id3->artist = NULL;
id3->composer = NULL;
id3->vbr = false; /* All SMAF files are CBR */
id3->filesize = filesize(fd);
/* get RIFF chunk header */
if ((lseek(fd, 0, SEEK_SET) < 0) || (read(fd, buf, 21) < 21))
{
return false;
}
if ((memcmp(buf, "MMMD", 4) != 0) || (memcmp(&buf[8], "CNTI", 4) != 0))
{
DEBUGF("metada error: does not smaf format\n");
return false;
}
contents_size = get_long_be(buf + 12);
if (contents_size < 5)
{
DEBUGF("metada error: CNTI chunk size is small %d\n", contents_size);
return false;
}
contents_size -= 5;
i = contents_size;
if (i == 0)
{
read(fd, buf, 16);
if (memcmp(buf, "OPDA", 4) != 0)
{
DEBUGF("metada error: smaf does not include OPDA chunk\n");
return false;
}
contents_size = get_long_be(buf + 4) - 8;
if (memcmp(buf + 8, "Dch", 3) != 0)
{
DEBUGF("metada error: smaf does not include Dch chunk\n");
return false;
}
codepage = convert_smaf_codetype(buf[11]);
if (codepage < 0)
{
DEBUGF("metada error: smaf unsupport codetype: %d\n", buf[11]);
return false;
}
i = get_long_be(buf + 12);
if (i > MAX_PATH)
{
DEBUGF("metada warning: smaf contents size is big %d\n", i);
i = MAX_PATH;
}
if (read(fd, buf, i) < i)
return false;
/* title */
if (read_datachunk(buf, i, TAG_TITLE, codepage, id3->id3v1buf[0]))
id3->title = id3->id3v1buf[0];
/* artist */
if (read_datachunk(buf, i, TAG_ARTIST, codepage, id3->id3v1buf[1]))
id3->artist = id3->id3v1buf[1];
/* composer */
if (read_datachunk(buf, i, TAG_COMPOSER, codepage, id3->id3v1buf[2]))
id3->composer = id3->id3v1buf[2];
}
else
{
codepage = convert_smaf_codetype(buf[14]);
if (codepage < 0)
{
DEBUGF("metada error: smaf unsupport codetype: %d\n", buf[11]);
return false;
}
if (i > MAX_PATH)
{
DEBUGF("metada warning: smaf contents size is big %d\n", i);
i = MAX_PATH;
}
if (read(fd, buf, i) < i)
return false;
/* title */
if (read_option(buf, i, TAG_TITLE, codepage, id3->id3v1buf[0]))
id3->title = id3->id3v1buf[0];
/* artist */
if (read_option(buf, i, TAG_ARTIST, codepage, id3->id3v1buf[1]))
id3->artist = id3->id3v1buf[1];
/* composer */
if (read_option(buf, i, TAG_COMPOSER, codepage, id3->id3v1buf[2]))
id3->composer = id3->id3v1buf[2];
}
if (contents_size > i)
lseek(fd, contents_size - i, SEEK_CUR);
/* assume the SMAF pcm data position is less than 1024 bytes */
if (read(fd, smafbuf, 1024) < 1024)
return false;
buf = smafbuf;
while (buf + 8 < endbuf)
{
i = get_long_be(buf + 4) + 8;
if (memcmp(buf, "ATR", 3) == 0)
return get_smaf_metadata_audio_track(id3, buf, endbuf);
else if (memcmp(buf, "MTR", 3) == 0)
return get_smaf_metadata_score_track(id3, buf, endbuf);
buf += i;
}
DEBUGF("metada error: smaf does not include track chunk\n");
return false;
}