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rockbox/apps/metadata.h
Dominik Riebeling 70b81e65cc Android: install codecs as native libs instead of extracting them (FS#12134). 
Codec files are loaded as dynamic libraries. Instead of extracting them from
the packaged libmisc.so and therefore having them present twice on the device
put them into the apk as native libraries. Decreases the size of the installed
Rockbox by the compressed size of the codecs. Also, the extraction on first
Rockbox startup gets notably faster since it's less data to extract.

git-svn-id: svn://svn.rockbox.org/rockbox/trunk@29940 a1c6a512-1295-4272-9138-f99709370657
2011-05-31 21:26:18 +00:00

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
*
* 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.
*
****************************************************************************/
#ifndef _METADATA_H
#define _METADATA_H
#include <stdbool.h>
#include "config.h"
#include "file.h"
/* Audio file types. */
/* NOTE: The values of the AFMT_* items are used for the %fc tag in the WPS
- so new entries MUST be added to the end to maintain compatibility.
*/
enum
{
AFMT_UNKNOWN = 0, /* Unknown file format */
/* start formats */
AFMT_MPA_L1, /* MPEG Audio layer 1 */
AFMT_MPA_L2, /* MPEG Audio layer 2 */
AFMT_MPA_L3, /* MPEG Audio layer 3 */
#if CONFIG_CODEC == SWCODEC
AFMT_AIFF, /* Audio Interchange File Format */
AFMT_PCM_WAV, /* Uncompressed PCM in a WAV file */
AFMT_OGG_VORBIS, /* Ogg Vorbis */
AFMT_FLAC, /* FLAC */
AFMT_MPC_SV7, /* Musepack SV7 */
AFMT_A52, /* A/52 (aka AC3) audio */
AFMT_WAVPACK, /* WavPack */
AFMT_MP4_ALAC, /* Apple Lossless Audio Codec */
AFMT_MP4_AAC, /* Advanced Audio Coding (AAC) in M4A container */
AFMT_SHN, /* Shorten */
AFMT_SID, /* SID File Format */
AFMT_ADX, /* ADX File Format */
AFMT_NSF, /* NESM (NES Sound Format) */
AFMT_SPEEX, /* Ogg Speex speech */
AFMT_SPC, /* SPC700 save state */
AFMT_APE, /* Monkey's Audio (APE) */
AFMT_WMA, /* WMAV1/V2 in ASF */
AFMT_WMAPRO, /* WMA Professional in ASF */
AFMT_MOD, /* Amiga MOD File Format */
AFMT_SAP, /* Atari 8Bit SAP Format */
AFMT_RM_COOK, /* Cook in RM/RA */
AFMT_RM_AAC, /* AAC in RM/RA */
AFMT_RM_AC3, /* AC3 in RM/RA */
AFMT_RM_ATRAC3, /* ATRAC3 in RM/RA */
AFMT_CMC, /* Atari 8bit cmc format */
AFMT_CM3, /* Atari 8bit cm3 format */
AFMT_CMR, /* Atari 8bit cmr format */
AFMT_CMS, /* Atari 8bit cms format */
AFMT_DMC, /* Atari 8bit dmc format */
AFMT_DLT, /* Atari 8bit dlt format */
AFMT_MPT, /* Atari 8bit mpt format */
AFMT_MPD, /* Atari 8bit mpd format */
AFMT_RMT, /* Atari 8bit rmt format */
AFMT_TMC, /* Atari 8bit tmc format */
AFMT_TM8, /* Atari 8bit tm8 format */
AFMT_TM2, /* Atari 8bit tm2 format */
AFMT_OMA_ATRAC3, /* Atrac3 in Sony OMA container */
AFMT_SMAF, /* SMAF */
AFMT_AU, /* Sun Audio file */
AFMT_VOX, /* VOX */
AFMT_WAVE64, /* Wave64 */
AFMT_TTA, /* True Audio */
AFMT_WMAVOICE, /* WMA Voice in ASF */
AFMT_MPC_SV8, /* Musepack SV8 */
AFMT_MP4_AAC_HE, /* Advanced Audio Coding (AAC-HE) in M4A container */
#endif
/* add new formats at any index above this line to have a sensible order -
specified array index inits are used */
/* format arrays defined in id3.c */
AFMT_NUM_CODECS,
#if CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING)
/* masks to decompose parts */
CODEC_AFMT_MASK = 0x0fff,
CODEC_TYPE_MASK = 0x7000,
/* switch for specifying codec type when requesting a filename */
CODEC_TYPE_DECODER = (0 << 12), /* default */
CODEC_TYPE_ENCODER = (1 << 12),
#endif /* CONFIG_CODEC == SWCODEC && defined(HAVE_RECORDING) */
};
#if CONFIG_CODEC == SWCODEC
#if (CONFIG_PLATFORM & PLATFORM_ANDROID)
#define CODEC_EXTENSION "so"
#define CODEC_PREFIX "lib"
#else
#define CODEC_EXTENSION "codec"
#define CODEC_PREFIX ""
#endif
#ifdef HAVE_RECORDING
enum rec_format_indexes
{
__REC_FORMAT_START_INDEX = -1,
/* start formats */
REC_FORMAT_PCM_WAV,
REC_FORMAT_AIFF,
REC_FORMAT_WAVPACK,
REC_FORMAT_MPA_L3,
/* add new formats at any index above this line to have a sensible order -
specified array index inits are used
REC_FORMAT_CFG_NUM_BITS should allocate enough bits to hold the range
REC_FORMAT_CFG_VALUE_LIST should be in same order as indexes
*/
REC_NUM_FORMATS,
REC_FORMAT_DEFAULT = REC_FORMAT_PCM_WAV,
REC_FORMAT_CFG_NUM_BITS = 2
};
#define REC_FORMAT_CFG_VAL_LIST "wave,aiff,wvpk,mpa3"
/* get REC_FORMAT_* corresponding AFMT_* */
extern const int rec_format_afmt[REC_NUM_FORMATS];
/* get AFMT_* corresponding REC_FORMAT_* */
extern const int afmt_rec_format[AFMT_NUM_CODECS];
#define AFMT_ENTRY(label, root_fname, enc_root_fname, func, ext_list) \
{ label, root_fname, enc_root_fname, func, ext_list }
#else /* !HAVE_RECORDING */
#define AFMT_ENTRY(label, root_fname, enc_root_fname, func, ext_list) \
{ label, root_fname, func, ext_list }
#endif /* HAVE_RECORDING */
#else /* !SWCODEC */
#define AFMT_ENTRY(label, root_fname, enc_root_fname, func, ext_list) \
{ label, func, ext_list }
#endif /* CONFIG_CODEC == SWCODEC */
/** Database of audio formats **/
/* record describing the audio format */
struct mp3entry;
struct afmt_entry
{
const char *label; /* format label */
#if CONFIG_CODEC == SWCODEC
const char *codec_root_fn; /* root codec filename (sans _enc and .codec) */
#ifdef HAVE_RECORDING
const char *codec_enc_root_fn; /* filename of encoder codec */
#endif
#endif
bool (*parse_func)(int fd, struct mp3entry *id3); /* return true on success */
const char *ext_list; /* NULL terminated extension
list for type with the first as
the default for recording */
};
/* database of labels and codecs. add formats per above enum */
extern const struct afmt_entry audio_formats[AFMT_NUM_CODECS];
#if MEMORYSIZE > 2
#define ID3V2_BUF_SIZE 900
#define ID3V2_MAX_ITEM_SIZE 240
#else
#define ID3V2_BUF_SIZE 300
#define ID3V2_MAX_ITEM_SIZE 90
#endif
enum {
ID3_VER_1_0 = 1,
ID3_VER_1_1,
ID3_VER_2_2,
ID3_VER_2_3,
ID3_VER_2_4
};
#ifdef HAVE_ALBUMART
enum mp3_aa_type {
AA_TYPE_UNSYNC = -1,
AA_TYPE_UNKNOWN,
AA_TYPE_BMP,
AA_TYPE_PNG,
AA_TYPE_JPG,
};
struct mp3_albumart {
enum mp3_aa_type type;
int size;
off_t pos;
};
#endif
struct mp3entry {
char path[MAX_PATH];
char* title;
char* artist;
char* album;
char* genre_string;
char* disc_string;
char* track_string;
char* year_string;
char* composer;
char* comment;
char* albumartist;
char* grouping;
int discnum;
int tracknum;
int layer;
int year;
unsigned char id3version;
unsigned int codectype;
unsigned int bitrate;
unsigned long frequency;
unsigned long id3v2len;
unsigned long id3v1len;
unsigned long first_frame_offset; /* Byte offset to first real MP3 frame.
Used for skipping leading garbage to
avoid gaps between tracks. */
unsigned long filesize; /* without headers; in bytes */
unsigned long length; /* song length in ms */
unsigned long elapsed; /* ms played */
int lead_trim; /* Number of samples to skip at the beginning */
int tail_trim; /* Number of samples to remove from the end */
/* Added for Vorbis, used by mp4 parser as well. */
unsigned long samples; /* number of samples in track */
/* MP3 stream specific info */
unsigned long frame_count; /* number of frames in the file (if VBR) */
/* Used for A52/AC3 */
unsigned long bytesperframe; /* number of bytes per frame (if CBR) */
/* Xing VBR fields */
bool vbr;
bool has_toc; /* True if there is a VBR header in the file */
unsigned char toc[100]; /* table of contents */
/* Added for ATRAC3 */
unsigned int channels; /* Number of channels in the stream */
unsigned int extradata_size; /* Size (in bytes) of the codec's extradata from the container */
/* Added for AAC HE SBR */
bool needs_upsampling_correction; /* flag used by aac codec */
/* these following two fields are used for local buffering */
char id3v2buf[ID3V2_BUF_SIZE];
char id3v1buf[4][92];
/* resume related */
unsigned long offset; /* bytes played */
int index; /* playlist index */
#ifdef HAVE_TAGCACHE
unsigned char autoresumable; /* caches result of autoresumable() */
/* runtime database fields */
long tagcache_idx; /* 0=invalid, otherwise idx+1 */
int rating;
int score;
long playcount;
long lastplayed;
long playtime;
#endif
/* replaygain support */
#if CONFIG_CODEC == SWCODEC
long track_level; /* holds the level in dB * (1<<FP_BITS) */
long album_level;
long track_gain; /* s19.12 signed fixed point. 0 for no gain. */
long album_gain;
long track_peak; /* s19.12 signed fixed point. 0 for no peak. */
long album_peak;
#endif
#ifdef HAVE_ALBUMART
bool embed_albumart;
struct mp3_albumart albumart;
#endif
/* Cuesheet support */
struct cuesheet *cuesheet;
/* Musicbrainz Track ID */
char* mb_track_id;
};
unsigned int probe_file_format(const char *filename);
bool get_metadata(struct mp3entry* id3, int fd, const char* trackname);
bool mp3info(struct mp3entry *entry, const char *filename);
void adjust_mp3entry(struct mp3entry *entry, void *dest, const void *orig);
void copy_mp3entry(struct mp3entry *dest, const struct mp3entry *orig);
void wipe_mp3entry(struct mp3entry *id3);
#if CONFIG_CODEC == SWCODEC
void fill_metadata_from_path(struct mp3entry *id3, const char *trackname);
int get_audio_base_codec_type(int type);
void strip_tags(int handle_id);
enum data_type get_audio_base_data_type(int afmt);
bool format_buffers_with_offset(int afmt);
#endif
#ifdef HAVE_TAGCACHE
bool autoresumable(struct mp3entry *id3);
#endif
#endif
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