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Add Sun Audio codec. (FS#10433)

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24955 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Yoshihisa Uchida 2010-02-28 07:22:20 +00:00
parent 1fefb48e87
commit 4e3c807466
9 changed files with 459 additions and 0 deletions
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1
apps/SOURCES
View file

@ -188,6 +188,7 @@ metadata/rm.c
metadata/nsf.c
metadata/oma.c
metadata/smaf.c
metadata/au.c
#endif
#ifdef HAVE_TAGCACHE
tagcache.c

1
apps/codecs/SOURCES
View file

@ -28,6 +28,7 @@ aiff.c
speex.c
adx.c
smaf.c
au.c
#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
/* encoders */
aiff_enc.c

319
apps/codecs/au.c Normal file
View file

@ -0,0 +1,319 @@
/***************************************************************************
* __________ __ ___.
* 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
/* Sun Audio file (Au file format) codec
*
* References
* [1] Sun Microsystems, Inc., Header file for Audio, .au, 1992
* URL http://www.opengroup.org/public/pubs/external/auformat.html
* [2] Wikipedia, Au file format, URL: http://en.wikipedia.org/wiki/Sun_Audio
*/
#define PCM_SAMPLE_SIZE (1024*2)
static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
enum
{
AU_FORMAT_UNSUPPORT = 0, /* unsupported format */
AU_FORMAT_MULAW, /* G.711 MULAW */
AU_FORMAT_PCM, /* Linear PCM */
AU_FORMAT_IEEE_FLOAT, /* IEEE float */
AU_FORMAT_ALAW, /* G.711 ALAW */
};
static int support_formats[28][2] = {
{ AU_FORMAT_UNSUPPORT, 0 },
{ AU_FORMAT_MULAW, 8 }, /* G.711 MULAW */
{ AU_FORMAT_PCM, 8 }, /* Linear PCM 8bit (signed) */
{ AU_FORMAT_PCM, 16 }, /* Linear PCM 16bit (signed, big endian) */
{ AU_FORMAT_PCM, 24 }, /* Linear PCM 24bit (signed, big endian) */
{ AU_FORMAT_PCM, 32 }, /* Linear PCM 32bit (signed, big endian) */
{ AU_FORMAT_IEEE_FLOAT, 32 }, /* Linear PCM float 32bit (signed, big endian) */
{ AU_FORMAT_IEEE_FLOAT, 64 }, /* Linear PCM float 64bit (signed, big endian) */
{ AU_FORMAT_UNSUPPORT, 0 }, /* Fragmented sample data */
{ AU_FORMAT_UNSUPPORT, 0 }, /* DSP program */
{ AU_FORMAT_UNSUPPORT, 0 }, /* 8bit fixed point */
{ AU_FORMAT_UNSUPPORT, 0 }, /* 16bit fixed point */
{ AU_FORMAT_UNSUPPORT, 0 }, /* 24bit fixed point */
{ AU_FORMAT_UNSUPPORT, 0 }, /* 32bit fixed point */
{ AU_FORMAT_UNSUPPORT, 0 },
{ AU_FORMAT_UNSUPPORT, 0 },
{ AU_FORMAT_UNSUPPORT, 0 },
{ AU_FORMAT_UNSUPPORT, 0 },
{ AU_FORMAT_UNSUPPORT, 0 }, /* 16bit linear with emphasis */
{ AU_FORMAT_UNSUPPORT, 0 }, /* 16bit linear compressed */
{ AU_FORMAT_UNSUPPORT, 0 }, /* 16bit linear with emphasis and compression */
{ AU_FORMAT_UNSUPPORT, 0 }, /* Music kit DSP commands */
{ AU_FORMAT_UNSUPPORT, 0 },
{ AU_FORMAT_UNSUPPORT, 0 }, /* G.721 MULAW */
{ AU_FORMAT_UNSUPPORT, 0 }, /* G.722 */
{ AU_FORMAT_UNSUPPORT, 0 }, /* G.723 3bit */
{ AU_FORMAT_UNSUPPORT, 0 }, /* G.723 5bit */
{ AU_FORMAT_ALAW, 8 }, /* G.711 ALAW */
};
const struct pcm_entry au_codecs[] = {
{ AU_FORMAT_MULAW, get_itut_g711_mulaw_codec },
{ AU_FORMAT_PCM, get_linear_pcm_codec },
{ AU_FORMAT_IEEE_FLOAT, get_ieee_float_codec },
{ AU_FORMAT_ALAW, get_itut_g711_alaw_codec },
};
#define NUM_FORMATS 4
static const struct pcm_codec *get_au_codec(uint32_t formattag)
{
int i;
for (i = 0; i < NUM_FORMATS; i++)
{
if (au_codecs[i].format_tag == formattag)
{
if (au_codecs[i].get_codec)
return au_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_au_format(unsigned int encoding, struct pcm_format *fmt)
{
if (encoding > 27)
{
fmt->formattag = AU_FORMAT_UNSUPPORT;
fmt->bitspersample = 0;
}
else
{
fmt->formattag = support_formats[encoding][0];
fmt->bitspersample = support_formats[encoding][1];
}
return fmt->formattag;
}
/* this is the codec entry point */
enum codec_status codec_main(void)
{
int status = CODEC_OK;
struct pcm_format format;
uint32_t bytesdone, decodedsamples;
size_t n;
int bufcount;
int endofstream;
unsigned char *buf;
uint8_t *aubuf;
off_t firstblockposn; /* position of the first block in file */
const struct pcm_codec *codec;
int offset = 0;
/* Generic codec initialisation */
ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
next_track:
if (codec_init()) {
DEBUGF("codec_init() error\n");
status = 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;
/* set format */
buf = ci->request_buffer(&n, 24);
if (n < 24 || (memcmp(buf, ".snd", 4) != 0))
{
/*
* headerless sun audio file
* It is decoded under conditions.
* format: G.711 mu-law
* channel: mono
* frequency: 8000 kHz
*/
offset = 0;
format.formattag = AU_FORMAT_MULAW;
format.channels = 1;
format.bitspersample = 8;
format.numbytes = ci->id3->filesize;
}
else
{
/* parse header */
/* data offset */
offset = get_be32(buf + 4);
if (offset < 24)
{
DEBUGF("CODEC_ERROR: sun audio offset size is small: %d\n", offset);
status = CODEC_ERROR;
goto done;
}
/* data size */
format.numbytes = get_be32(buf + 8);
if (format.numbytes == (uint32_t)0xffffffff)
format.numbytes = ci->id3->filesize - offset;
/* encoding */
format.formattag = convert_au_format(get_be32(buf + 12), &format);
if (format.formattag == AU_FORMAT_UNSUPPORT)
{
DEBUGF("CODEC_ERROR: sun audio unsupport format: %d\n", get_be32(buf + 12));
status = CODEC_ERROR;
goto done;
}
/* skip sample rate */
format.channels = get_be32(buf + 20);
if (format.channels == 0) {
DEBUGF("CODEC_ERROR: sun audio 0-channels file\n");
status = CODEC_ERROR;
goto done;
}
}
/* advance to first WAVE chunk */
ci->advance_buffer(offset);
firstblockposn = offset;
decodedsamples = 0;
codec = 0;
bytesdone = 0;
/* blockalign = 1 sample */
format.blockalign = format.bitspersample * format.channels >> 3;
/* get codec */
codec = get_au_codec(format.formattag);
if (!codec)
{
DEBUGF("CODEC_ERROR: unsupport sun audio format: %lx\n", format.formattag);
status = CODEC_ERROR;
goto done;
}
if (!codec->set_format(&format))
{
status = CODEC_ERROR;
goto done;
}
if (format.numbytes == 0) {
DEBUGF("CODEC_ERROR: data size is 0\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");
status = 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\n");
status = CODEC_ERROR;
goto done;
}
/* The main decoder loop */
endofstream = 0;
while (!endofstream) {
ci->yield();
if (ci->stop_codec || ci->new_track) {
break;
}
if (ci->seek_time) {
/* 2nd args(read_buffer) is unnecessary in the format which Sun Audio supports. */
struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, NULL);
decodedsamples = newpos->samples;
if (newpos->pos > format.numbytes)
break;
if (ci->seek_buffer(firstblockposn + newpos->pos))
{
bytesdone = newpos->pos;
}
ci->seek_complete();
}
aubuf = (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(aubuf, 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);
}
status = CODEC_OK;
done:
if (ci->request_next_track())
goto next_track;
exit:
return status;
}

1
apps/codecs/codecs.make
View file

@ -91,6 +91,7 @@ $(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
$(CODECDIR)/au.codec : $(CODECDIR)/libpcm.a
$(CODECS): $(CODECLIB) # this must be last in codec dependency list

2
apps/filetypes.c
View file

@ -102,6 +102,8 @@ static const struct filetype inbuilt_filetypes[] = {
{ "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 },
{ "au", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
{ "snd", 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 },

11
apps/metadata.c
View file

@ -168,6 +168,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
/* SMAF (Synthetic music Mobile Application Format) */
[AFMT_SMAF] =
AFMT_ENTRY("SMAF", "smaf", NULL, "mmf\0" ),
/* Sun Audio file */
[AFMT_AU] =
AFMT_ENTRY("AU", "au", NULL, "au\0snd\0" ),
#endif
};
@ -459,6 +462,14 @@ bool get_metadata(struct mp3entry* id3, int fd, const char* trackname)
}
break;
case AFMT_AU:
if (!get_au_metadata(fd, id3))
{
DEBUGF("get_au_metadata error\n");
return false;
}
break;
#endif /* CONFIG_CODEC == SWCODEC */
default:

1
apps/metadata.h
View file

@ -79,6 +79,7 @@ enum
AFMT_TM2, /* Atari 8bit tm2 format */
AFMT_OMA_ATRAC3, /* Atrac3 in Sony OMA container */
AFMT_SMAF, /* SMAF */
AFMT_AU, /* Sun Audio file */
#endif
/* add new formats at any index above this line to have a sensible order -

122
apps/metadata/au.c Normal file
View file

@ -0,0 +1,122 @@
/***************************************************************************
* __________ __ ___.
* 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 int 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;
}

1
apps/metadata/metadata_parsers.h
View file

@ -43,3 +43,4 @@ 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);
bool get_au_metadata(int fd, struct mp3entry* id3);