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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
#define CODEC_EXTENSION "codec"

#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 120
#else
#define ID3V2_BUF_SIZE 300
#define ID3V2_MAX_ITEM_SIZE 60
#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 version;
    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 vbr_header_pos;
    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 */
    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_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);

#if CONFIG_CODEC == SWCODEC
void strip_tags(int handle_id);
#endif

#ifdef HAVE_TAGCACHE
bool autoresumable(struct mp3entry *id3);
#endif

#endif