Add Sun Audio codec. (FS#10433)
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@24955 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Yoshihisa Uchida
parent
1fefb48e87
commit
4e3c807466
9 changed files with 459 additions and 0 deletions
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@ -188,6 +188,7 @@ metadata/rm.c
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metadata/nsf.c
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metadata/oma.c
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metadata/smaf.c
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metadata/au.c
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#endif
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#ifdef HAVE_TAGCACHE
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tagcache.c
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@ -28,6 +28,7 @@ aiff.c
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speex.c
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adx.c
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smaf.c
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au.c
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#if defined(HAVE_RECORDING) && !defined(SIMULATOR)
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/* encoders */
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aiff_enc.c
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319
apps/codecs/au.c
Normal file
319
apps/codecs/au.c
Normal file
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@ -0,0 +1,319 @@
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/***************************************************************************
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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 "codeclib.h"
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#include "codecs/libpcm/support_formats.h"
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CODEC_HEADER
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/* Sun Audio file (Au file format) codec
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*
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* References
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* [1] Sun Microsystems, Inc., Header file for Audio, .au, 1992
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* URL http://www.opengroup.org/public/pubs/external/auformat.html
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* [2] Wikipedia, Au file format, URL: http://en.wikipedia.org/wiki/Sun_Audio
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*/
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#define PCM_SAMPLE_SIZE (1024*2)
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static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
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enum
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{
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AU_FORMAT_UNSUPPORT = 0, /* unsupported format */
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AU_FORMAT_MULAW, /* G.711 MULAW */
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AU_FORMAT_PCM, /* Linear PCM */
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AU_FORMAT_IEEE_FLOAT, /* IEEE float */
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AU_FORMAT_ALAW, /* G.711 ALAW */
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};
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static int support_formats[28][2] = {
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{ AU_FORMAT_UNSUPPORT, 0 },
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{ AU_FORMAT_MULAW, 8 }, /* G.711 MULAW */
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{ AU_FORMAT_PCM, 8 }, /* Linear PCM 8bit (signed) */
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{ AU_FORMAT_PCM, 16 }, /* Linear PCM 16bit (signed, big endian) */
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{ AU_FORMAT_PCM, 24 }, /* Linear PCM 24bit (signed, big endian) */
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{ AU_FORMAT_PCM, 32 }, /* Linear PCM 32bit (signed, big endian) */
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{ AU_FORMAT_IEEE_FLOAT, 32 }, /* Linear PCM float 32bit (signed, big endian) */
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{ AU_FORMAT_IEEE_FLOAT, 64 }, /* Linear PCM float 64bit (signed, big endian) */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* Fragmented sample data */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* DSP program */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* 8bit fixed point */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* 16bit fixed point */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* 24bit fixed point */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* 32bit fixed point */
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{ AU_FORMAT_UNSUPPORT, 0 },
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{ AU_FORMAT_UNSUPPORT, 0 },
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{ AU_FORMAT_UNSUPPORT, 0 },
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{ AU_FORMAT_UNSUPPORT, 0 },
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{ AU_FORMAT_UNSUPPORT, 0 }, /* 16bit linear with emphasis */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* 16bit linear compressed */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* 16bit linear with emphasis and compression */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* Music kit DSP commands */
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{ AU_FORMAT_UNSUPPORT, 0 },
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{ AU_FORMAT_UNSUPPORT, 0 }, /* G.721 MULAW */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* G.722 */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* G.723 3bit */
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{ AU_FORMAT_UNSUPPORT, 0 }, /* G.723 5bit */
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{ AU_FORMAT_ALAW, 8 }, /* G.711 ALAW */
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};
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const struct pcm_entry au_codecs[] = {
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{ AU_FORMAT_MULAW, get_itut_g711_mulaw_codec },
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{ AU_FORMAT_PCM, get_linear_pcm_codec },
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{ AU_FORMAT_IEEE_FLOAT, get_ieee_float_codec },
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{ AU_FORMAT_ALAW, get_itut_g711_alaw_codec },
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};
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#define NUM_FORMATS 4
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static const struct pcm_codec *get_au_codec(uint32_t formattag)
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{
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int i;
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for (i = 0; i < NUM_FORMATS; i++)
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{
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if (au_codecs[i].format_tag == formattag)
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{
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if (au_codecs[i].get_codec)
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return au_codecs[i].get_codec();
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return 0;
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}
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}
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return 0;
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}
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static unsigned int get_be32(uint8_t *buf)
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{
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return (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3];
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}
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static int convert_au_format(unsigned int encoding, struct pcm_format *fmt)
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{
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if (encoding > 27)
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{
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fmt->formattag = AU_FORMAT_UNSUPPORT;
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fmt->bitspersample = 0;
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}
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else
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{
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fmt->formattag = support_formats[encoding][0];
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fmt->bitspersample = support_formats[encoding][1];
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}
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return fmt->formattag;
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}
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/* this is the codec entry point */
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enum codec_status codec_main(void)
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{
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int status = CODEC_OK;
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struct pcm_format format;
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uint32_t bytesdone, decodedsamples;
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size_t n;
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int bufcount;
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int endofstream;
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unsigned char *buf;
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uint8_t *aubuf;
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off_t firstblockposn; /* position of the first block in file */
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const struct pcm_codec *codec;
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int offset = 0;
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/* Generic codec initialisation */
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ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
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next_track:
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if (codec_init()) {
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DEBUGF("codec_init() error\n");
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status = CODEC_ERROR;
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goto exit;
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}
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while (!*ci->taginfo_ready && !ci->stop_codec)
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ci->sleep(1);
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codec_set_replaygain(ci->id3);
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ci->memset(&format, 0, sizeof(struct pcm_format));
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format.is_signed = true;
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format.is_little_endian = false;
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/* set format */
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buf = ci->request_buffer(&n, 24);
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if (n < 24 || (memcmp(buf, ".snd", 4) != 0))
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{
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/*
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* headerless sun audio file
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* It is decoded under conditions.
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* format: G.711 mu-law
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* channel: mono
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* frequency: 8000 kHz
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*/
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offset = 0;
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format.formattag = AU_FORMAT_MULAW;
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format.channels = 1;
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format.bitspersample = 8;
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format.numbytes = ci->id3->filesize;
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}
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else
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{
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/* parse header */
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/* data offset */
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offset = get_be32(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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status = CODEC_ERROR;
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goto done;
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}
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/* data size */
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format.numbytes = get_be32(buf + 8);
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if (format.numbytes == (uint32_t)0xffffffff)
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format.numbytes = ci->id3->filesize - offset;
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/* encoding */
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format.formattag = convert_au_format(get_be32(buf + 12), &format);
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if (format.formattag == AU_FORMAT_UNSUPPORT)
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{
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DEBUGF("CODEC_ERROR: sun audio unsupport format: %d\n", get_be32(buf + 12));
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status = CODEC_ERROR;
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goto done;
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}
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/* skip sample rate */
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format.channels = get_be32(buf + 20);
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if (format.channels == 0) {
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DEBUGF("CODEC_ERROR: sun audio 0-channels file\n");
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status = CODEC_ERROR;
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goto done;
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}
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}
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/* advance to first WAVE chunk */
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ci->advance_buffer(offset);
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firstblockposn = offset;
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decodedsamples = 0;
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codec = 0;
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bytesdone = 0;
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/* blockalign = 1 sample */
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format.blockalign = format.bitspersample * format.channels >> 3;
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/* get codec */
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codec = get_au_codec(format.formattag);
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if (!codec)
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{
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DEBUGF("CODEC_ERROR: unsupport sun audio format: %lx\n", format.formattag);
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status = CODEC_ERROR;
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goto done;
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}
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if (!codec->set_format(&format))
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{
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status = CODEC_ERROR;
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goto done;
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}
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if (format.numbytes == 0) {
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DEBUGF("CODEC_ERROR: data size is 0\n");
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status = CODEC_ERROR;
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goto done;
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}
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/* check chunksize */
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if ((format.chunksize / format.blockalign) * format.samplesperblock * format.channels
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> PCM_SAMPLE_SIZE)
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format.chunksize = (PCM_SAMPLE_SIZE / format.blockalign) * format.blockalign;
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if (format.chunksize == 0)
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{
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DEBUGF("CODEC_ERROR: chunksize is 0\n");
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status = CODEC_ERROR;
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goto done;
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}
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ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
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if (format.channels == 2) {
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ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
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} else if (format.channels == 1) {
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ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
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} else {
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DEBUGF("CODEC_ERROR: more than 2 channels\n");
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status = CODEC_ERROR;
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goto done;
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}
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/* The main decoder loop */
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endofstream = 0;
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while (!endofstream) {
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ci->yield();
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if (ci->stop_codec || ci->new_track) {
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break;
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}
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if (ci->seek_time) {
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/* 2nd args(read_buffer) is unnecessary in the format which Sun Audio supports. */
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struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, NULL);
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decodedsamples = newpos->samples;
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if (newpos->pos > format.numbytes)
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break;
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if (ci->seek_buffer(firstblockposn + newpos->pos))
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{
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bytesdone = newpos->pos;
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}
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ci->seek_complete();
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}
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aubuf = (uint8_t *)ci->request_buffer(&n, format.chunksize);
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if (n == 0)
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break; /* End of stream */
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if (bytesdone + n > format.numbytes) {
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n = format.numbytes - bytesdone;
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endofstream = 1;
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}
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status = codec->decode(aubuf, n, samples, &bufcount);
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if (status == CODEC_ERROR)
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{
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DEBUGF("codec error\n");
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goto done;
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}
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ci->pcmbuf_insert(samples, NULL, bufcount);
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ci->advance_buffer(n);
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bytesdone += n;
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decodedsamples += bufcount;
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if (bytesdone >= format.numbytes)
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endofstream = 1;
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ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
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}
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status = CODEC_OK;
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done:
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if (ci->request_next_track())
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goto next_track;
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exit:
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return status;
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}
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@ -91,6 +91,7 @@ $(CODECDIR)/atrac3_oma.codec : $(CODECDIR)/libatrac.a
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$(CODECDIR)/aiff.codec : $(CODECDIR)/libpcm.a
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$(CODECDIR)/wav.codec : $(CODECDIR)/libpcm.a
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$(CODECDIR)/smaf.codec : $(CODECDIR)/libpcm.a
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$(CODECDIR)/au.codec : $(CODECDIR)/libpcm.a
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$(CODECS): $(CODECLIB) # this must be last in codec dependency list
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@ -102,6 +102,8 @@ static const struct filetype inbuilt_filetypes[] = {
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{ "aa3", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
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{ "at3", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
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{ "mmf", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
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{ "au", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
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{ "snd", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA },
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#endif
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{ "m3u", FILE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST },
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{ "m3u8",FILE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST },
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@ -168,6 +168,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] =
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/* SMAF (Synthetic music Mobile Application Format) */
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[AFMT_SMAF] =
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AFMT_ENTRY("SMAF", "smaf", NULL, "mmf\0" ),
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/* Sun Audio file */
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[AFMT_AU] =
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AFMT_ENTRY("AU", "au", NULL, "au\0snd\0" ),
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#endif
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};
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@ -458,6 +461,14 @@ bool get_metadata(struct mp3entry* id3, int fd, const char* trackname)
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return false;
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}
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break;
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case AFMT_AU:
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if (!get_au_metadata(fd, id3))
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{
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DEBUGF("get_au_metadata error\n");
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return false;
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}
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break;
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#endif /* CONFIG_CODEC == SWCODEC */
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@ -79,6 +79,7 @@ enum
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AFMT_TM2, /* Atari 8bit tm2 format */
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AFMT_OMA_ATRAC3, /* Atrac3 in Sony OMA container */
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AFMT_SMAF, /* SMAF */
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AFMT_AU, /* Sun Audio file */
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#endif
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/* add new formats at any index above this line to have a sensible order -
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122
apps/metadata/au.c
Normal file
122
apps/metadata/au.c
Normal file
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@ -0,0 +1,122 @@
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/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
|
||||
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
|
||||
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
|
||||
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
|
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* \/ \/ \/ \/ \/
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* $Id$
|
||||
*
|
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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
|
||||
* modify it under the terms of the GNU General Public License
|
||||
* 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 int 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)
|
||||
{
|
||||
/* 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;
|
||||
}
|
||||
|
|
@ -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);
|
||||
|
|
|
|||
Loading…
Reference in a new issue