rockbox/apps/metadata.h
Alexander Levin 63c4ef9f57 Rename 'mp3entry.embed_albumart' to 'mp3entry.has_embedded_albumart' (FS#12470). No functional changes.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@31405 a1c6a512-1295-4272-9138-f99709370657
2011-12-22 18:48:43 +00:00

353 lines
11 KiB
C

/***************************************************************************
* __________ __ ___.
* 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 */
AFMT_AY, /* AY (ZX Spectrum, Amstrad CPC Sound Format) */
AFMT_GBS, /* GBS (Game Boy Sound Format) */
AFMT_HES, /* HES (Hudson Entertainment System Sound Format) */
AFMT_SGC, /* SGC (Sega Master System, Game Gear, Coleco Vision Sound Format) */
AFMT_VGM, /* VGM (Video Game Music Format) */
AFMT_KSS, /* KSS (MSX computer KSS Music File) */
#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_* */
/* unused: 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
enum character_encoding {
CHAR_ENC_ISO_8859_1 = 1,
CHAR_ENC_UTF_8,
CHAR_ENC_UTF_16_LE,
CHAR_ENC_UTF_16_BE,
};
/* cache embedded cuesheet details */
struct embedded_cuesheet {
bool present;
int size;
off_t pos;
enum character_encoding encoding;
};
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 has_embedded_albumart;
struct mp3_albumart albumart;
#endif
/* Cuesheet support */
struct embedded_cuesheet embed_cuesheet;
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