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Redesigned the audio encoding/decoding classes to get rid of libsndfile

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This commit is contained in:
Laurent Gomila 2014-07-20 17:48:40 +02:00 • committed by Mario Liebisch
parent 34692d5a39
commit f0608eaed8
68 changed files with 4523 additions and 2241 deletions
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666
extlibs/headers/libsndfile/android/sndfile.h vendored
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/*
** Copyright (C) 1999-2011Erik de Castro Lopo <erikd@mega-nerd.com>
**
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU Lesser General Public License as published by
** the Free Software Foundation; either version 2.1 of the License, or
** (at your option) any later version.
**
** This program is distributed in the hope that it will be useful,
** but WITHOUT ANY WARRANTY; without even the implied warranty of
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
** GNU Lesser General Public License for more details.
**
** You should have received a copy of the GNU Lesser General Public License
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
/*
** sndfile.h -- system-wide definitions
**
** API documentation is in the doc/ directory of the source code tarball
** and at http://www.mega-nerd.com/libsndfile/api.html.
*/
#ifndef SNDFILE_H
#define SNDFILE_H
/* This is the version 1.0.X header file. */
#define SNDFILE_1
#include <stdio.h>
#include <sys/types.h>
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
/* The following file types can be read and written.
** A file type would consist of a major type (ie SF_FORMAT_WAV) bitwise
** ORed with a minor type (ie SF_FORMAT_PCM). SF_FORMAT_TYPEMASK and
** SF_FORMAT_SUBMASK can be used to separate the major and minor file
** types.
*/
enum
{ /* Major formats. */
SF_FORMAT_WAV = 0x010000, /* Microsoft WAV format (little endian default). */
SF_FORMAT_AIFF = 0x020000, /* Apple/SGI AIFF format (big endian). */
SF_FORMAT_AU = 0x030000, /* Sun/NeXT AU format (big endian). */
SF_FORMAT_RAW = 0x040000, /* RAW PCM data. */
SF_FORMAT_PAF = 0x050000, /* Ensoniq PARIS file format. */
SF_FORMAT_SVX = 0x060000, /* Amiga IFF / SVX8 / SV16 format. */
SF_FORMAT_NIST = 0x070000, /* Sphere NIST format. */
SF_FORMAT_VOC = 0x080000, /* VOC files. */
SF_FORMAT_IRCAM = 0x0A0000, /* Berkeley/IRCAM/CARL */
SF_FORMAT_W64 = 0x0B0000, /* Sonic Foundry's 64 bit RIFF/WAV */
SF_FORMAT_MAT4 = 0x0C0000, /* Matlab (tm) V4.2 / GNU Octave 2.0 */
SF_FORMAT_MAT5 = 0x0D0000, /* Matlab (tm) V5.0 / GNU Octave 2.1 */
SF_FORMAT_PVF = 0x0E0000, /* Portable Voice Format */
SF_FORMAT_XI = 0x0F0000, /* Fasttracker 2 Extended Instrument */
SF_FORMAT_HTK = 0x100000, /* HMM Tool Kit format */
SF_FORMAT_SDS = 0x110000, /* Midi Sample Dump Standard */
SF_FORMAT_AVR = 0x120000, /* Audio Visual Research */
SF_FORMAT_WAVEX = 0x130000, /* MS WAVE with WAVEFORMATEX */
SF_FORMAT_SD2 = 0x160000, /* Sound Designer 2 */
SF_FORMAT_FLAC = 0x170000, /* FLAC lossless file format */
SF_FORMAT_CAF = 0x180000, /* Core Audio File format */
SF_FORMAT_WVE = 0x190000, /* Psion WVE format */
SF_FORMAT_OGG = 0x200000, /* Xiph OGG container */
SF_FORMAT_MPC2K = 0x210000, /* Akai MPC 2000 sampler */
SF_FORMAT_RF64 = 0x220000, /* RF64 WAV file */
/* Subtypes from here on. */
SF_FORMAT_PCM_S8 = 0x0001, /* Signed 8 bit data */
SF_FORMAT_PCM_16 = 0x0002, /* Signed 16 bit data */
SF_FORMAT_PCM_24 = 0x0003, /* Signed 24 bit data */
SF_FORMAT_PCM_32 = 0x0004, /* Signed 32 bit data */
SF_FORMAT_PCM_U8 = 0x0005, /* Unsigned 8 bit data (WAV and RAW only) */
SF_FORMAT_FLOAT = 0x0006, /* 32 bit float data */
SF_FORMAT_DOUBLE = 0x0007, /* 64 bit float data */
SF_FORMAT_ULAW = 0x0010, /* U-Law encoded. */
SF_FORMAT_ALAW = 0x0011, /* A-Law encoded. */
SF_FORMAT_IMA_ADPCM = 0x0012, /* IMA ADPCM. */
SF_FORMAT_MS_ADPCM = 0x0013, /* Microsoft ADPCM. */
SF_FORMAT_GSM610 = 0x0020, /* GSM 6.10 encoding. */
SF_FORMAT_VOX_ADPCM = 0x0021, /* OKI / Dialogix ADPCM */
SF_FORMAT_G721_32 = 0x0030, /* 32kbs G721 ADPCM encoding. */
SF_FORMAT_G723_24 = 0x0031, /* 24kbs G723 ADPCM encoding. */
SF_FORMAT_G723_40 = 0x0032, /* 40kbs G723 ADPCM encoding. */
SF_FORMAT_DWVW_12 = 0x0040, /* 12 bit Delta Width Variable Word encoding. */
SF_FORMAT_DWVW_16 = 0x0041, /* 16 bit Delta Width Variable Word encoding. */
SF_FORMAT_DWVW_24 = 0x0042, /* 24 bit Delta Width Variable Word encoding. */
SF_FORMAT_DWVW_N = 0x0043, /* N bit Delta Width Variable Word encoding. */
SF_FORMAT_DPCM_8 = 0x0050, /* 8 bit differential PCM (XI only) */
SF_FORMAT_DPCM_16 = 0x0051, /* 16 bit differential PCM (XI only) */
SF_FORMAT_VORBIS = 0x0060, /* Xiph Vorbis encoding. */
/* Endian-ness options. */
SF_ENDIAN_FILE = 0x00000000, /* Default file endian-ness. */
SF_ENDIAN_LITTLE = 0x10000000, /* Force little endian-ness. */
SF_ENDIAN_BIG = 0x20000000, /* Force big endian-ness. */
SF_ENDIAN_CPU = 0x30000000, /* Force CPU endian-ness. */
SF_FORMAT_SUBMASK = 0x0000FFFF,
SF_FORMAT_TYPEMASK = 0x0FFF0000,
SF_FORMAT_ENDMASK = 0x30000000
} ;
/*
** The following are the valid command numbers for the sf_command()
** interface. The use of these commands is documented in the file
** command.html in the doc directory of the source code distribution.
*/
enum
{ SFC_GET_LIB_VERSION = 0x1000,
SFC_GET_LOG_INFO = 0x1001,
SFC_GET_CURRENT_SF_INFO = 0x1002,
SFC_GET_NORM_DOUBLE = 0x1010,
SFC_GET_NORM_FLOAT = 0x1011,
SFC_SET_NORM_DOUBLE = 0x1012,
SFC_SET_NORM_FLOAT = 0x1013,
SFC_SET_SCALE_FLOAT_INT_READ = 0x1014,
SFC_SET_SCALE_INT_FLOAT_WRITE = 0x1015,
SFC_GET_SIMPLE_FORMAT_COUNT = 0x1020,
SFC_GET_SIMPLE_FORMAT = 0x1021,
SFC_GET_FORMAT_INFO = 0x1028,
SFC_GET_FORMAT_MAJOR_COUNT = 0x1030,
SFC_GET_FORMAT_MAJOR = 0x1031,
SFC_GET_FORMAT_SUBTYPE_COUNT = 0x1032,
SFC_GET_FORMAT_SUBTYPE = 0x1033,
SFC_CALC_SIGNAL_MAX = 0x1040,
SFC_CALC_NORM_SIGNAL_MAX = 0x1041,
SFC_CALC_MAX_ALL_CHANNELS = 0x1042,
SFC_CALC_NORM_MAX_ALL_CHANNELS = 0x1043,
SFC_GET_SIGNAL_MAX = 0x1044,
SFC_GET_MAX_ALL_CHANNELS = 0x1045,
SFC_SET_ADD_PEAK_CHUNK = 0x1050,
SFC_SET_ADD_HEADER_PAD_CHUNK = 0x1051,
SFC_UPDATE_HEADER_NOW = 0x1060,
SFC_SET_UPDATE_HEADER_AUTO = 0x1061,
SFC_FILE_TRUNCATE = 0x1080,
SFC_SET_RAW_START_OFFSET = 0x1090,
SFC_SET_DITHER_ON_WRITE = 0x10A0,
SFC_SET_DITHER_ON_READ = 0x10A1,
SFC_GET_DITHER_INFO_COUNT = 0x10A2,
SFC_GET_DITHER_INFO = 0x10A3,
SFC_GET_EMBED_FILE_INFO = 0x10B0,
SFC_SET_CLIPPING = 0x10C0,
SFC_GET_CLIPPING = 0x10C1,
SFC_GET_INSTRUMENT = 0x10D0,
SFC_SET_INSTRUMENT = 0x10D1,
SFC_GET_LOOP_INFO = 0x10E0,
SFC_GET_BROADCAST_INFO = 0x10F0,
SFC_SET_BROADCAST_INFO = 0x10F1,
SFC_GET_CHANNEL_MAP_INFO = 0x1100,
SFC_SET_CHANNEL_MAP_INFO = 0x1101,
SFC_RAW_DATA_NEEDS_ENDSWAP = 0x1110,
/* Support for Wavex Ambisonics Format */
SFC_WAVEX_SET_AMBISONIC = 0x1200,
SFC_WAVEX_GET_AMBISONIC = 0x1201,
SFC_SET_VBR_ENCODING_QUALITY = 0x1300,
/* Following commands for testing only. */
SFC_TEST_IEEE_FLOAT_REPLACE = 0x6001,
/*
** SFC_SET_ADD_* values are deprecated and will disappear at some
** time in the future. They are guaranteed to be here up to and
** including version 1.0.8 to avoid breakage of existng software.
** They currently do nothing and will continue to do nothing.
*/
SFC_SET_ADD_DITHER_ON_WRITE = 0x1070,
SFC_SET_ADD_DITHER_ON_READ = 0x1071
} ;
/*
** String types that can be set and read from files. Not all file types
** support this and even the file types which support one, may not support
** all string types.
*/
enum
{ SF_STR_TITLE = 0x01,
SF_STR_COPYRIGHT = 0x02,
SF_STR_SOFTWARE = 0x03,
SF_STR_ARTIST = 0x04,
SF_STR_COMMENT = 0x05,
SF_STR_DATE = 0x06,
SF_STR_ALBUM = 0x07,
SF_STR_LICENSE = 0x08,
SF_STR_TRACKNUMBER = 0x09,
SF_STR_GENRE = 0x10
} ;
/*
** Use the following as the start and end index when doing metadata
** transcoding.
*/
#define SF_STR_FIRST SF_STR_TITLE
#define SF_STR_LAST SF_STR_GENRE
enum
{ /* True and false */
SF_FALSE = 0,
SF_TRUE = 1,
/* Modes for opening files. */
SFM_READ = 0x10,
SFM_WRITE = 0x20,
SFM_RDWR = 0x30,
SF_AMBISONIC_NONE = 0x40,
SF_AMBISONIC_B_FORMAT = 0x41
} ;
/* Public error values. These are guaranteed to remain unchanged for the duration
** of the library major version number.
** There are also a large number of private error numbers which are internal to
** the library which can change at any time.
*/
enum
{ SF_ERR_NO_ERROR = 0,
SF_ERR_UNRECOGNISED_FORMAT = 1,
SF_ERR_SYSTEM = 2,
SF_ERR_MALFORMED_FILE = 3,
SF_ERR_UNSUPPORTED_ENCODING = 4
} ;
/* Channel map values (used with SFC_SET/GET_CHANNEL_MAP).
*/
enum
{ SF_CHANNEL_MAP_INVALID = 0,
SF_CHANNEL_MAP_MONO = 1,
SF_CHANNEL_MAP_LEFT, /* Apple calls this 'Left' */
SF_CHANNEL_MAP_RIGHT, /* Apple calls this 'Right' */
SF_CHANNEL_MAP_CENTER, /* Apple calls this 'Center' */
SF_CHANNEL_MAP_FRONT_LEFT,
SF_CHANNEL_MAP_FRONT_RIGHT,
SF_CHANNEL_MAP_FRONT_CENTER,
SF_CHANNEL_MAP_REAR_CENTER, /* Apple calls this 'Center Surround', Msft calls this 'Back Center' */
SF_CHANNEL_MAP_REAR_LEFT, /* Apple calls this 'Left Surround', Msft calls this 'Back Left' */
SF_CHANNEL_MAP_REAR_RIGHT, /* Apple calls this 'Right Surround', Msft calls this 'Back Right' */
SF_CHANNEL_MAP_LFE, /* Apple calls this 'LFEScreen', Msft calls this 'Low Frequency' */
SF_CHANNEL_MAP_FRONT_LEFT_OF_CENTER, /* Apple calls this 'Left Center' */
SF_CHANNEL_MAP_FRONT_RIGHT_OF_CENTER, /* Apple calls this 'Right Center */
SF_CHANNEL_MAP_SIDE_LEFT, /* Apple calls this 'Left Surround Direct' */
SF_CHANNEL_MAP_SIDE_RIGHT, /* Apple calls this 'Right Surround Direct' */
SF_CHANNEL_MAP_TOP_CENTER, /* Apple calls this 'Top Center Surround' */
SF_CHANNEL_MAP_TOP_FRONT_LEFT, /* Apple calls this 'Vertical Height Left' */
SF_CHANNEL_MAP_TOP_FRONT_RIGHT, /* Apple calls this 'Vertical Height Right' */
SF_CHANNEL_MAP_TOP_FRONT_CENTER, /* Apple calls this 'Vertical Height Center' */
SF_CHANNEL_MAP_TOP_REAR_LEFT, /* Apple and MS call this 'Top Back Left' */
SF_CHANNEL_MAP_TOP_REAR_RIGHT, /* Apple and MS call this 'Top Back Right' */
SF_CHANNEL_MAP_TOP_REAR_CENTER, /* Apple and MS call this 'Top Back Center' */
SF_CHANNEL_MAP_AMBISONIC_B_W,
SF_CHANNEL_MAP_AMBISONIC_B_X,
SF_CHANNEL_MAP_AMBISONIC_B_Y,
SF_CHANNEL_MAP_AMBISONIC_B_Z,
SF_CHANNEL_MAP_MAX
} ;
/* A SNDFILE* pointer can be passed around much like stdio.h's FILE* pointer. */
typedef struct SNDFILE_tag SNDFILE ;
/* The following typedef is system specific and is defined when libsndfile is
** compiled. sf_count_t will be a 64 bit value when the underlying OS allows
** 64 bit file offsets.
** On windows, we need to allow the same header file to be compiler by both GCC
** and the Microsoft compiler.
*/
#if (defined (_MSCVER) || defined (_MSC_VER))
typedef __int64 sf_count_t ;
#define SF_COUNT_MAX 0x7fffffffffffffffi64
#else
typedef off_t sf_count_t ;
#define SF_COUNT_MAX 0x7FFFFFFF
#endif
/* A pointer to a SF_INFO structure is passed to sf_open () and filled in.
** On write, the SF_INFO structure is filled in by the user and passed into
** sf_open ().
*/
struct SF_INFO
{ sf_count_t frames ; /* Used to be called samples. Changed to avoid confusion. */
int samplerate ;
int channels ;
int format ;
int sections ;
int seekable ;
} ;
typedef struct SF_INFO SF_INFO ;
/* The SF_FORMAT_INFO struct is used to retrieve information about the sound
** file formats libsndfile supports using the sf_command () interface.
**
** Using this interface will allow applications to support new file formats
** and encoding types when libsndfile is upgraded, without requiring
** re-compilation of the application.
**
** Please consult the libsndfile documentation (particularly the information
** on the sf_command () interface) for examples of its use.
*/
typedef struct
{ int format ;
const char *name ;
const char *extension ;
} SF_FORMAT_INFO ;
/*
** Enums and typedefs for adding dither on read and write.
** See the html documentation for sf_command(), SFC_SET_DITHER_ON_WRITE
** and SFC_SET_DITHER_ON_READ.
*/
enum
{ SFD_DEFAULT_LEVEL = 0,
SFD_CUSTOM_LEVEL = 0x40000000,
SFD_NO_DITHER = 500,
SFD_WHITE = 501,
SFD_TRIANGULAR_PDF = 502
} ;
typedef struct
{ int type ;
double level ;
const char *name ;
} SF_DITHER_INFO ;
/* Struct used to retrieve information about a file embedded within a
** larger file. See SFC_GET_EMBED_FILE_INFO.
*/
typedef struct
{ sf_count_t offset ;
sf_count_t length ;
} SF_EMBED_FILE_INFO ;
/*
** Structs used to retrieve music sample information from a file.
*/
enum
{ /*
** The loop mode field in SF_INSTRUMENT will be one of the following.
*/
SF_LOOP_NONE = 800,
SF_LOOP_FORWARD,
SF_LOOP_BACKWARD,
SF_LOOP_ALTERNATING
} ;
typedef struct
{ int gain ;
char basenote, detune ;
char velocity_lo, velocity_hi ;
char key_lo, key_hi ;
int loop_count ;
struct
{ int mode ;
unsigned int start ;
unsigned int end ;
unsigned int count ;
} loops [16] ; /* make variable in a sensible way */
} SF_INSTRUMENT ;
/* Struct used to retrieve loop information from a file.*/
typedef struct
{
short time_sig_num ; /* any positive integer > 0 */
short time_sig_den ; /* any positive power of 2 > 0 */
int loop_mode ; /* see SF_LOOP enum */
int num_beats ; /* this is NOT the amount of quarter notes !!!*/
/* a full bar of 4/4 is 4 beats */
/* a full bar of 7/8 is 7 beats */
float bpm ; /* suggestion, as it can be calculated using other fields:*/
/* file's lenght, file's sampleRate and our time_sig_den*/
/* -> bpms are always the amount of _quarter notes_ per minute */
int root_key ; /* MIDI note, or -1 for None */
int future [6] ;
} SF_LOOP_INFO ;
/* Struct used to retrieve broadcast (EBU) information from a file.
** Strongly (!) based on EBU "bext" chunk format used in Broadcast WAVE.
*/
#define SF_BROADCAST_INFO_VAR(coding_hist_size) \
struct \
{ char description [256] ; \
char originator [32] ; \
char originator_reference [32] ; \
char origination_date [10] ; \
char origination_time [8] ; \
unsigned int time_reference_low ; \
unsigned int time_reference_high ; \
short version ; \
char umid [64] ; \
char reserved [190] ; \
unsigned int coding_history_size ; \
char coding_history [coding_hist_size] ; \
}
/* SF_BROADCAST_INFO is the above struct with coding_history field of 256 bytes. */
typedef SF_BROADCAST_INFO_VAR (256) SF_BROADCAST_INFO ;
/* Virtual I/O functionality. */
typedef sf_count_t (*sf_vio_get_filelen) (void *user_data) ;
typedef sf_count_t (*sf_vio_seek) (sf_count_t offset, int whence, void *user_data) ;
typedef sf_count_t (*sf_vio_read) (void *ptr, sf_count_t count, void *user_data) ;
typedef sf_count_t (*sf_vio_write) (const void *ptr, sf_count_t count, void *user_data) ;
typedef sf_count_t (*sf_vio_tell) (void *user_data) ;
struct SF_VIRTUAL_IO
{ sf_vio_get_filelen get_filelen ;
sf_vio_seek seek ;
sf_vio_read read ;
sf_vio_write write ;
sf_vio_tell tell ;
} ;
typedef struct SF_VIRTUAL_IO SF_VIRTUAL_IO ;
/* Open the specified file for read, write or both. On error, this will
** return a NULL pointer. To find the error number, pass a NULL SNDFILE
** to sf_strerror ().
** All calls to sf_open() should be matched with a call to sf_close().
*/
SNDFILE* sf_open (const char *path, int mode, SF_INFO *sfinfo) ;
/* Use the existing file descriptor to create a SNDFILE object. If close_desc
** is TRUE, the file descriptor will be closed when sf_close() is called. If
** it is FALSE, the descritor will not be closed.
** When passed a descriptor like this, the library will assume that the start
** of file header is at the current file offset. This allows sound files within
** larger container files to be read and/or written.
** On error, this will return a NULL pointer. To find the error number, pass a
** NULL SNDFILE to sf_strerror ().
** All calls to sf_open_fd() should be matched with a call to sf_close().
*/
SNDFILE* sf_open_fd (int fd, int mode, SF_INFO *sfinfo, int close_desc) ;
SNDFILE* sf_open_virtual (SF_VIRTUAL_IO *sfvirtual, int mode, SF_INFO *sfinfo, void *user_data) ;
/* sf_error () returns a error number which can be translated to a text
** string using sf_error_number().
*/
int sf_error (SNDFILE *sndfile) ;
/* sf_strerror () returns to the caller a pointer to the current error message for
** the given SNDFILE.
*/
const char* sf_strerror (SNDFILE *sndfile) ;
/* sf_error_number () allows the retrieval of the error string for each internal
** error number.
**
*/
const char* sf_error_number (int errnum) ;
/* The following two error functions are deprecated but they will remain in the
** library for the forseeable future. The function sf_strerror() should be used
** in their place.
*/
int sf_perror (SNDFILE *sndfile) ;
int sf_error_str (SNDFILE *sndfile, char* str, size_t len) ;
/* Return TRUE if fields of the SF_INFO struct are a valid combination of values. */
int sf_command (SNDFILE *sndfile, int command, void *data, int datasize) ;
/* Return TRUE if fields of the SF_INFO struct are a valid combination of values. */
int sf_format_check (const SF_INFO *info) ;
/* Seek within the waveform data chunk of the SNDFILE. sf_seek () uses
** the same values for whence (SEEK_SET, SEEK_CUR and SEEK_END) as
** stdio.h function fseek ().
** An offset of zero with whence set to SEEK_SET will position the
** read / write pointer to the first data sample.
** On success sf_seek returns the current position in (multi-channel)
** samples from the start of the file.
** Please see the libsndfile documentation for moving the read pointer
** separately from the write pointer on files open in mode SFM_RDWR.
** On error all of these functions return -1.
*/
sf_count_t sf_seek (SNDFILE *sndfile, sf_count_t frames, int whence) ;
/* Functions for retrieving and setting string data within sound files.
** Not all file types support this features; AIFF and WAV do. For both
** functions, the str_type parameter must be one of the SF_STR_* values
** defined above.
** On error, sf_set_string() returns non-zero while sf_get_string()
** returns NULL.
*/
int sf_set_string (SNDFILE *sndfile, int str_type, const char* str) ;
const char* sf_get_string (SNDFILE *sndfile, int str_type) ;
/* Return the library version string. */
const char * sf_version_string (void) ;
/* Functions for reading/writing the waveform data of a sound file.
*/
sf_count_t sf_read_raw (SNDFILE *sndfile, void *ptr, sf_count_t bytes) ;
sf_count_t sf_write_raw (SNDFILE *sndfile, const void *ptr, sf_count_t bytes) ;
/* Functions for reading and writing the data chunk in terms of frames.
** The number of items actually read/written = frames * number of channels.
** sf_xxxx_raw read/writes the raw data bytes from/to the file
** sf_xxxx_short passes data in the native short format
** sf_xxxx_int passes data in the native int format
** sf_xxxx_float passes data in the native float format
** sf_xxxx_double passes data in the native double format
** All of these read/write function return number of frames read/written.
*/
sf_count_t sf_readf_short (SNDFILE *sndfile, short *ptr, sf_count_t frames) ;
sf_count_t sf_writef_short (SNDFILE *sndfile, const short *ptr, sf_count_t frames) ;
sf_count_t sf_readf_int (SNDFILE *sndfile, int *ptr, sf_count_t frames) ;
sf_count_t sf_writef_int (SNDFILE *sndfile, const int *ptr, sf_count_t frames) ;
sf_count_t sf_readf_float (SNDFILE *sndfile, float *ptr, sf_count_t frames) ;
sf_count_t sf_writef_float (SNDFILE *sndfile, const float *ptr, sf_count_t frames) ;
sf_count_t sf_readf_double (SNDFILE *sndfile, double *ptr, sf_count_t frames) ;
sf_count_t sf_writef_double (SNDFILE *sndfile, const double *ptr, sf_count_t frames) ;
/* Functions for reading and writing the data chunk in terms of items.
** Otherwise similar to above.
** All of these read/write function return number of items read/written.
*/
sf_count_t sf_read_short (SNDFILE *sndfile, short *ptr, sf_count_t items) ;
sf_count_t sf_write_short (SNDFILE *sndfile, const short *ptr, sf_count_t items) ;
sf_count_t sf_read_int (SNDFILE *sndfile, int *ptr, sf_count_t items) ;
sf_count_t sf_write_int (SNDFILE *sndfile, const int *ptr, sf_count_t items) ;
sf_count_t sf_read_float (SNDFILE *sndfile, float *ptr, sf_count_t items) ;
sf_count_t sf_write_float (SNDFILE *sndfile, const float *ptr, sf_count_t items) ;
sf_count_t sf_read_double (SNDFILE *sndfile, double *ptr, sf_count_t items) ;
sf_count_t sf_write_double (SNDFILE *sndfile, const double *ptr, sf_count_t items) ;
/* Close the SNDFILE and clean up all memory allocations associated with this
** file.
** Returns 0 on success, or an error number.
*/
int sf_close (SNDFILE *sndfile) ;
/* If the file is opened SFM_WRITE or SFM_RDWR, call fsync() on the file
** to force the writing of data to disk. If the file is opened SFM_READ
** no action is taken.
*/
void sf_write_sync (SNDFILE *sndfile) ;
/* The function sf_wchar_open() is Windows Only!
** Open a file passing in a Windows Unicode filename. Otherwise, this is
** the same as sf_open().
**
** In order for this to work, you need to do the following:
**
** #include <windows.h>
** #define ENABLE_SNDFILE_WINDOWS_PROTOTYPES 1
** #including <sndfile.h>
*/
#if (defined (ENABLE_SNDFILE_WINDOWS_PROTOTYPES) && ENABLE_SNDFILE_WINDOWS_PROTOTYPES)
SNDFILE* sf_wchar_open (LPCWSTR wpath, int mode, SF_INFO *sfinfo) ;
#endif
#ifdef __cplusplus
} /* extern "C" */
#endif /* __cplusplus */
#endif /* SNDFILE_H */

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/*
** Copyright (C) 1999-2011Erik de Castro Lopo <erikd@mega-nerd.com>
**
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU Lesser General Public License as published by
** the Free Software Foundation; either version 2.1 of the License, or
** (at your option) any later version.
**
** This program is distributed in the hope that it will be useful,
** but WITHOUT ANY WARRANTY; without even the implied warranty of
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
** GNU Lesser General Public License for more details.
**
** You should have received a copy of the GNU Lesser General Public License
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
/*
** sndfile.h -- system-wide definitions
**
** API documentation is in the doc/ directory of the source code tarball
** and at http://www.mega-nerd.com/libsndfile/api.html.
*/
#ifndef SNDFILE_H
#define SNDFILE_H
/* This is the version 1.0.X header file. */
#define SNDFILE_1
#include <stdio.h>
#include <sys/types.h>
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
/* The following file types can be read and written.
** A file type would consist of a major type (ie SF_FORMAT_WAV) bitwise
** ORed with a minor type (ie SF_FORMAT_PCM). SF_FORMAT_TYPEMASK and
** SF_FORMAT_SUBMASK can be used to separate the major and minor file
** types.
*/
enum
{ /* Major formats. */
SF_FORMAT_WAV = 0x010000, /* Microsoft WAV format (little endian default). */
SF_FORMAT_AIFF = 0x020000, /* Apple/SGI AIFF format (big endian). */
SF_FORMAT_AU = 0x030000, /* Sun/NeXT AU format (big endian). */
SF_FORMAT_RAW = 0x040000, /* RAW PCM data. */
SF_FORMAT_PAF = 0x050000, /* Ensoniq PARIS file format. */
SF_FORMAT_SVX = 0x060000, /* Amiga IFF / SVX8 / SV16 format. */
SF_FORMAT_NIST = 0x070000, /* Sphere NIST format. */
SF_FORMAT_VOC = 0x080000, /* VOC files. */
SF_FORMAT_IRCAM = 0x0A0000, /* Berkeley/IRCAM/CARL */
SF_FORMAT_W64 = 0x0B0000, /* Sonic Foundry's 64 bit RIFF/WAV */
SF_FORMAT_MAT4 = 0x0C0000, /* Matlab (tm) V4.2 / GNU Octave 2.0 */
SF_FORMAT_MAT5 = 0x0D0000, /* Matlab (tm) V5.0 / GNU Octave 2.1 */
SF_FORMAT_PVF = 0x0E0000, /* Portable Voice Format */
SF_FORMAT_XI = 0x0F0000, /* Fasttracker 2 Extended Instrument */
SF_FORMAT_HTK = 0x100000, /* HMM Tool Kit format */
SF_FORMAT_SDS = 0x110000, /* Midi Sample Dump Standard */
SF_FORMAT_AVR = 0x120000, /* Audio Visual Research */
SF_FORMAT_WAVEX = 0x130000, /* MS WAVE with WAVEFORMATEX */
SF_FORMAT_SD2 = 0x160000, /* Sound Designer 2 */
SF_FORMAT_FLAC = 0x170000, /* FLAC lossless file format */
SF_FORMAT_CAF = 0x180000, /* Core Audio File format */
SF_FORMAT_WVE = 0x190000, /* Psion WVE format */
SF_FORMAT_OGG = 0x200000, /* Xiph OGG container */
SF_FORMAT_MPC2K = 0x210000, /* Akai MPC 2000 sampler */
SF_FORMAT_RF64 = 0x220000, /* RF64 WAV file */
/* Subtypes from here on. */
SF_FORMAT_PCM_S8 = 0x0001, /* Signed 8 bit data */
SF_FORMAT_PCM_16 = 0x0002, /* Signed 16 bit data */
SF_FORMAT_PCM_24 = 0x0003, /* Signed 24 bit data */
SF_FORMAT_PCM_32 = 0x0004, /* Signed 32 bit data */
SF_FORMAT_PCM_U8 = 0x0005, /* Unsigned 8 bit data (WAV and RAW only) */
SF_FORMAT_FLOAT = 0x0006, /* 32 bit float data */
SF_FORMAT_DOUBLE = 0x0007, /* 64 bit float data */
SF_FORMAT_ULAW = 0x0010, /* U-Law encoded. */
SF_FORMAT_ALAW = 0x0011, /* A-Law encoded. */
SF_FORMAT_IMA_ADPCM = 0x0012, /* IMA ADPCM. */
SF_FORMAT_MS_ADPCM = 0x0013, /* Microsoft ADPCM. */
SF_FORMAT_GSM610 = 0x0020, /* GSM 6.10 encoding. */
SF_FORMAT_VOX_ADPCM = 0x0021, /* OKI / Dialogix ADPCM */
SF_FORMAT_G721_32 = 0x0030, /* 32kbs G721 ADPCM encoding. */
SF_FORMAT_G723_24 = 0x0031, /* 24kbs G723 ADPCM encoding. */
SF_FORMAT_G723_40 = 0x0032, /* 40kbs G723 ADPCM encoding. */
SF_FORMAT_DWVW_12 = 0x0040, /* 12 bit Delta Width Variable Word encoding. */
SF_FORMAT_DWVW_16 = 0x0041, /* 16 bit Delta Width Variable Word encoding. */
SF_FORMAT_DWVW_24 = 0x0042, /* 24 bit Delta Width Variable Word encoding. */
SF_FORMAT_DWVW_N = 0x0043, /* N bit Delta Width Variable Word encoding. */
SF_FORMAT_DPCM_8 = 0x0050, /* 8 bit differential PCM (XI only) */
SF_FORMAT_DPCM_16 = 0x0051, /* 16 bit differential PCM (XI only) */
SF_FORMAT_VORBIS = 0x0060, /* Xiph Vorbis encoding. */
/* Endian-ness options. */
SF_ENDIAN_FILE = 0x00000000, /* Default file endian-ness. */
SF_ENDIAN_LITTLE = 0x10000000, /* Force little endian-ness. */
SF_ENDIAN_BIG = 0x20000000, /* Force big endian-ness. */
SF_ENDIAN_CPU = 0x30000000, /* Force CPU endian-ness. */
SF_FORMAT_SUBMASK = 0x0000FFFF,
SF_FORMAT_TYPEMASK = 0x0FFF0000,
SF_FORMAT_ENDMASK = 0x30000000
} ;
/*
** The following are the valid command numbers for the sf_command()
** interface. The use of these commands is documented in the file
** command.html in the doc directory of the source code distribution.
*/
enum
{ SFC_GET_LIB_VERSION = 0x1000,
SFC_GET_LOG_INFO = 0x1001,
SFC_GET_CURRENT_SF_INFO = 0x1002,
SFC_GET_NORM_DOUBLE = 0x1010,
SFC_GET_NORM_FLOAT = 0x1011,
SFC_SET_NORM_DOUBLE = 0x1012,
SFC_SET_NORM_FLOAT = 0x1013,
SFC_SET_SCALE_FLOAT_INT_READ = 0x1014,
SFC_SET_SCALE_INT_FLOAT_WRITE = 0x1015,
SFC_GET_SIMPLE_FORMAT_COUNT = 0x1020,
SFC_GET_SIMPLE_FORMAT = 0x1021,
SFC_GET_FORMAT_INFO = 0x1028,
SFC_GET_FORMAT_MAJOR_COUNT = 0x1030,
SFC_GET_FORMAT_MAJOR = 0x1031,
SFC_GET_FORMAT_SUBTYPE_COUNT = 0x1032,
SFC_GET_FORMAT_SUBTYPE = 0x1033,
SFC_CALC_SIGNAL_MAX = 0x1040,
SFC_CALC_NORM_SIGNAL_MAX = 0x1041,
SFC_CALC_MAX_ALL_CHANNELS = 0x1042,
SFC_CALC_NORM_MAX_ALL_CHANNELS = 0x1043,
SFC_GET_SIGNAL_MAX = 0x1044,
SFC_GET_MAX_ALL_CHANNELS = 0x1045,
SFC_SET_ADD_PEAK_CHUNK = 0x1050,
SFC_SET_ADD_HEADER_PAD_CHUNK = 0x1051,
SFC_UPDATE_HEADER_NOW = 0x1060,
SFC_SET_UPDATE_HEADER_AUTO = 0x1061,
SFC_FILE_TRUNCATE = 0x1080,
SFC_SET_RAW_START_OFFSET = 0x1090,
SFC_SET_DITHER_ON_WRITE = 0x10A0,
SFC_SET_DITHER_ON_READ = 0x10A1,
SFC_GET_DITHER_INFO_COUNT = 0x10A2,
SFC_GET_DITHER_INFO = 0x10A3,
SFC_GET_EMBED_FILE_INFO = 0x10B0,
SFC_SET_CLIPPING = 0x10C0,
SFC_GET_CLIPPING = 0x10C1,
SFC_GET_INSTRUMENT = 0x10D0,
SFC_SET_INSTRUMENT = 0x10D1,
SFC_GET_LOOP_INFO = 0x10E0,
SFC_GET_BROADCAST_INFO = 0x10F0,
SFC_SET_BROADCAST_INFO = 0x10F1,
SFC_GET_CHANNEL_MAP_INFO = 0x1100,
SFC_SET_CHANNEL_MAP_INFO = 0x1101,
SFC_RAW_DATA_NEEDS_ENDSWAP = 0x1110,
/* Support for Wavex Ambisonics Format */
SFC_WAVEX_SET_AMBISONIC = 0x1200,
SFC_WAVEX_GET_AMBISONIC = 0x1201,
SFC_SET_VBR_ENCODING_QUALITY = 0x1300,
/* Following commands for testing only. */
SFC_TEST_IEEE_FLOAT_REPLACE = 0x6001,
/*
** SFC_SET_ADD_* values are deprecated and will disappear at some
** time in the future. They are guaranteed to be here up to and
** including version 1.0.8 to avoid breakage of existng software.
** They currently do nothing and will continue to do nothing.
*/
SFC_SET_ADD_DITHER_ON_WRITE = 0x1070,
SFC_SET_ADD_DITHER_ON_READ = 0x1071
} ;
/*
** String types that can be set and read from files. Not all file types
** support this and even the file types which support one, may not support
** all string types.
*/
enum
{ SF_STR_TITLE = 0x01,
SF_STR_COPYRIGHT = 0x02,
SF_STR_SOFTWARE = 0x03,
SF_STR_ARTIST = 0x04,
SF_STR_COMMENT = 0x05,
SF_STR_DATE = 0x06,
SF_STR_ALBUM = 0x07,
SF_STR_LICENSE = 0x08,
SF_STR_TRACKNUMBER = 0x09,
SF_STR_GENRE = 0x10
} ;
/*
** Use the following as the start and end index when doing metadata
** transcoding.
*/
#define SF_STR_FIRST SF_STR_TITLE
#define SF_STR_LAST SF_STR_LICENSE
enum
{ /* True and false */
SF_FALSE = 0,
SF_TRUE = 1,
/* Modes for opening files. */
SFM_READ = 0x10,
SFM_WRITE = 0x20,
SFM_RDWR = 0x30,
SF_AMBISONIC_NONE = 0x40,
SF_AMBISONIC_B_FORMAT = 0x41
} ;
/* Public error values. These are guaranteed to remain unchanged for the duration
** of the library major version number.
** There are also a large number of private error numbers which are internal to
** the library which can change at any time.
*/
enum
{ SF_ERR_NO_ERROR = 0,
SF_ERR_UNRECOGNISED_FORMAT = 1,
SF_ERR_SYSTEM = 2,
SF_ERR_MALFORMED_FILE = 3,
SF_ERR_UNSUPPORTED_ENCODING = 4
} ;
/* Channel map values (used with SFC_SET/GET_CHANNEL_MAP).
*/
enum
{ SF_CHANNEL_MAP_INVALID = 0,
SF_CHANNEL_MAP_MONO = 1,
SF_CHANNEL_MAP_LEFT, /* Apple calls this 'Left' */
SF_CHANNEL_MAP_RIGHT, /* Apple calls this 'Right' */
SF_CHANNEL_MAP_CENTER, /* Apple calls this 'Center' */
SF_CHANNEL_MAP_FRONT_LEFT,
SF_CHANNEL_MAP_FRONT_RIGHT,
SF_CHANNEL_MAP_FRONT_CENTER,
SF_CHANNEL_MAP_REAR_CENTER, /* Apple calls this 'Center Surround', Msft calls this 'Back Center' */
SF_CHANNEL_MAP_REAR_LEFT, /* Apple calls this 'Left Surround', Msft calls this 'Back Left' */
SF_CHANNEL_MAP_REAR_RIGHT, /* Apple calls this 'Right Surround', Msft calls this 'Back Right' */
SF_CHANNEL_MAP_LFE, /* Apple calls this 'LFEScreen', Msft calls this 'Low Frequency' */
SF_CHANNEL_MAP_FRONT_LEFT_OF_CENTER, /* Apple calls this 'Left Center' */
SF_CHANNEL_MAP_FRONT_RIGHT_OF_CENTER, /* Apple calls this 'Right Center */
SF_CHANNEL_MAP_SIDE_LEFT, /* Apple calls this 'Left Surround Direct' */
SF_CHANNEL_MAP_SIDE_RIGHT, /* Apple calls this 'Right Surround Direct' */
SF_CHANNEL_MAP_TOP_CENTER, /* Apple calls this 'Top Center Surround' */
SF_CHANNEL_MAP_TOP_FRONT_LEFT, /* Apple calls this 'Vertical Height Left' */
SF_CHANNEL_MAP_TOP_FRONT_RIGHT, /* Apple calls this 'Vertical Height Right' */
SF_CHANNEL_MAP_TOP_FRONT_CENTER, /* Apple calls this 'Vertical Height Center' */
SF_CHANNEL_MAP_TOP_REAR_LEFT, /* Apple and MS call this 'Top Back Left' */
SF_CHANNEL_MAP_TOP_REAR_RIGHT, /* Apple and MS call this 'Top Back Right' */
SF_CHANNEL_MAP_TOP_REAR_CENTER, /* Apple and MS call this 'Top Back Center' */
SF_CHANNEL_MAP_AMBISONIC_B_W,
SF_CHANNEL_MAP_AMBISONIC_B_X,
SF_CHANNEL_MAP_AMBISONIC_B_Y,
SF_CHANNEL_MAP_AMBISONIC_B_Z,
SF_CHANNEL_MAP_MAX
} ;
/* A SNDFILE* pointer can be passed around much like stdio.h's FILE* pointer. */
typedef struct SNDFILE_tag SNDFILE ;
/* The following typedef is system specific and is defined when libsndfile is
** compiled. sf_count_t will be a 64 bit value when the underlying OS allows
** 64 bit file offsets.
** On windows, we need to allow the same header file to be compiler by both GCC
** and the Microsoft compiler.
*/
#if (defined (_MSCVER) || defined (_MSC_VER))
typedef __int64 sf_count_t ;
#define SF_COUNT_MAX 0x7fffffffffffffffi64
#else
typedef int64_t sf_count_t ;
#define SF_COUNT_MAX 0x7FFFFFFFFFFFFFFFLL
#endif
/* A pointer to a SF_INFO structure is passed to sf_open () and filled in.
** On write, the SF_INFO structure is filled in by the user and passed into
** sf_open ().
*/
struct SF_INFO
{ sf_count_t frames ; /* Used to be called samples. Changed to avoid confusion. */
int samplerate ;
int channels ;
int format ;
int sections ;
int seekable ;
} ;
typedef struct SF_INFO SF_INFO ;
/* The SF_FORMAT_INFO struct is used to retrieve information about the sound
** file formats libsndfile supports using the sf_command () interface.
**
** Using this interface will allow applications to support new file formats
** and encoding types when libsndfile is upgraded, without requiring
** re-compilation of the application.
**
** Please consult the libsndfile documentation (particularly the information
** on the sf_command () interface) for examples of its use.
*/
typedef struct
{ int format ;
const char *name ;
const char *extension ;
} SF_FORMAT_INFO ;
/*
** Enums and typedefs for adding dither on read and write.
** See the html documentation for sf_command(), SFC_SET_DITHER_ON_WRITE
** and SFC_SET_DITHER_ON_READ.
*/
enum
{ SFD_DEFAULT_LEVEL = 0,
SFD_CUSTOM_LEVEL = 0x40000000,
SFD_NO_DITHER = 500,
SFD_WHITE = 501,
SFD_TRIANGULAR_PDF = 502
} ;
typedef struct
{ int type ;
double level ;
const char *name ;
} SF_DITHER_INFO ;
/* Struct used to retrieve information about a file embedded within a
** larger file. See SFC_GET_EMBED_FILE_INFO.
*/
typedef struct
{ sf_count_t offset ;
sf_count_t length ;
} SF_EMBED_FILE_INFO ;
/*
** Structs used to retrieve music sample information from a file.
*/
enum
{ /*
** The loop mode field in SF_INSTRUMENT will be one of the following.
*/
SF_LOOP_NONE = 800,
SF_LOOP_FORWARD,
SF_LOOP_BACKWARD,
SF_LOOP_ALTERNATING
} ;
typedef struct
{ int gain ;
char basenote, detune ;
char velocity_lo, velocity_hi ;
char key_lo, key_hi ;
int loop_count ;
struct
{ int mode ;
unsigned int start ;
unsigned int end ;
unsigned int count ;
} loops [16] ; /* make variable in a sensible way */
} SF_INSTRUMENT ;
/* Struct used to retrieve loop information from a file.*/
typedef struct
{
short time_sig_num ; /* any positive integer > 0 */
short time_sig_den ; /* any positive power of 2 > 0 */
int loop_mode ; /* see SF_LOOP enum */
int num_beats ; /* this is NOT the amount of quarter notes !!!*/
/* a full bar of 4/4 is 4 beats */
/* a full bar of 7/8 is 7 beats */
float bpm ; /* suggestion, as it can be calculated using other fields:*/
/* file's lenght, file's sampleRate and our time_sig_den*/
/* -> bpms are always the amount of _quarter notes_ per minute */
int root_key ; /* MIDI note, or -1 for None */
int future [6] ;
} SF_LOOP_INFO ;
/* Struct used to retrieve broadcast (EBU) information from a file.
** Strongly (!) based on EBU "bext" chunk format used in Broadcast WAVE.
*/
#define SF_BROADCAST_INFO_VAR(coding_hist_size) \
struct \
{ char description [256] ; \
char originator [32] ; \
char originator_reference [32] ; \
char origination_date [10] ; \
char origination_time [8] ; \
unsigned int time_reference_low ; \
unsigned int time_reference_high ; \
short version ; \
char umid [64] ; \
char reserved [190] ; \
unsigned int coding_history_size ; \
char coding_history [coding_hist_size] ; \
}
/* SF_BROADCAST_INFO is the above struct with coding_history field of 256 bytes. */
typedef SF_BROADCAST_INFO_VAR (256) SF_BROADCAST_INFO ;
/* Virtual I/O functionality. */
typedef sf_count_t (*sf_vio_get_filelen) (void *user_data) ;
typedef sf_count_t (*sf_vio_seek) (sf_count_t offset, int whence, void *user_data) ;
typedef sf_count_t (*sf_vio_read) (void *ptr, sf_count_t count, void *user_data) ;
typedef sf_count_t (*sf_vio_write) (const void *ptr, sf_count_t count, void *user_data) ;
typedef sf_count_t (*sf_vio_tell) (void *user_data) ;
struct SF_VIRTUAL_IO
{ sf_vio_get_filelen get_filelen ;
sf_vio_seek seek ;
sf_vio_read read ;
sf_vio_write write ;
sf_vio_tell tell ;
} ;
typedef struct SF_VIRTUAL_IO SF_VIRTUAL_IO ;
/* Open the specified file for read, write or both. On error, this will
** return a NULL pointer. To find the error number, pass a NULL SNDFILE
** to sf_strerror ().
** All calls to sf_open() should be matched with a call to sf_close().
*/
SNDFILE* sf_open (const char *path, int mode, SF_INFO *sfinfo) ;
/* Use the existing file descriptor to create a SNDFILE object. If close_desc
** is TRUE, the file descriptor will be closed when sf_close() is called. If
** it is FALSE, the descritor will not be closed.
** When passed a descriptor like this, the library will assume that the start
** of file header is at the current file offset. This allows sound files within
** larger container files to be read and/or written.
** On error, this will return a NULL pointer. To find the error number, pass a
** NULL SNDFILE to sf_strerror ().
** All calls to sf_open_fd() should be matched with a call to sf_close().
*/
SNDFILE* sf_open_fd (int fd, int mode, SF_INFO *sfinfo, int close_desc) ;
SNDFILE* sf_open_virtual (SF_VIRTUAL_IO *sfvirtual, int mode, SF_INFO *sfinfo, void *user_data) ;
/* sf_error () returns a error number which can be translated to a text
** string using sf_error_number().
*/
int sf_error (SNDFILE *sndfile) ;
/* sf_strerror () returns to the caller a pointer to the current error message for
** the given SNDFILE.
*/
const char* sf_strerror (SNDFILE *sndfile) ;
/* sf_error_number () allows the retrieval of the error string for each internal
** error number.
**
*/
const char* sf_error_number (int errnum) ;
/* The following two error functions are deprecated but they will remain in the
** library for the forseeable future. The function sf_strerror() should be used
** in their place.
*/
int sf_perror (SNDFILE *sndfile) ;
int sf_error_str (SNDFILE *sndfile, char* str, size_t len) ;
/* Return TRUE if fields of the SF_INFO struct are a valid combination of values. */
int sf_command (SNDFILE *sndfile, int command, void *data, int datasize) ;
/* Return TRUE if fields of the SF_INFO struct are a valid combination of values. */
int sf_format_check (const SF_INFO *info) ;
/* Seek within the waveform data chunk of the SNDFILE. sf_seek () uses
** the same values for whence (SEEK_SET, SEEK_CUR and SEEK_END) as
** stdio.h function fseek ().
** An offset of zero with whence set to SEEK_SET will position the
** read / write pointer to the first data sample.
** On success sf_seek returns the current position in (multi-channel)
** samples from the start of the file.
** Please see the libsndfile documentation for moving the read pointer
** separately from the write pointer on files open in mode SFM_RDWR.
** On error all of these functions return -1.
*/
sf_count_t sf_seek (SNDFILE *sndfile, sf_count_t frames, int whence) ;
/* Functions for retrieving and setting string data within sound files.
** Not all file types support this features; AIFF and WAV do. For both
** functions, the str_type parameter must be one of the SF_STR_* values
** defined above.
** On error, sf_set_string() returns non-zero while sf_get_string()
** returns NULL.
*/
int sf_set_string (SNDFILE *sndfile, int str_type, const char* str) ;
const char* sf_get_string (SNDFILE *sndfile, int str_type) ;
/* Return the library version string. */
const char * sf_version_string (void) ;
/* Functions for reading/writing the waveform data of a sound file.
*/
sf_count_t sf_read_raw (SNDFILE *sndfile, void *ptr, sf_count_t bytes) ;
sf_count_t sf_write_raw (SNDFILE *sndfile, const void *ptr, sf_count_t bytes) ;
/* Functions for reading and writing the data chunk in terms of frames.
** The number of items actually read/written = frames * number of channels.
** sf_xxxx_raw read/writes the raw data bytes from/to the file
** sf_xxxx_short passes data in the native short format
** sf_xxxx_int passes data in the native int format
** sf_xxxx_float passes data in the native float format
** sf_xxxx_double passes data in the native double format
** All of these read/write function return number of frames read/written.
*/
sf_count_t sf_readf_short (SNDFILE *sndfile, short *ptr, sf_count_t frames) ;
sf_count_t sf_writef_short (SNDFILE *sndfile, const short *ptr, sf_count_t frames) ;
sf_count_t sf_readf_int (SNDFILE *sndfile, int *ptr, sf_count_t frames) ;
sf_count_t sf_writef_int (SNDFILE *sndfile, const int *ptr, sf_count_t frames) ;
sf_count_t sf_readf_float (SNDFILE *sndfile, float *ptr, sf_count_t frames) ;
sf_count_t sf_writef_float (SNDFILE *sndfile, const float *ptr, sf_count_t frames) ;
sf_count_t sf_readf_double (SNDFILE *sndfile, double *ptr, sf_count_t frames) ;
sf_count_t sf_writef_double (SNDFILE *sndfile, const double *ptr, sf_count_t frames) ;
/* Functions for reading and writing the data chunk in terms of items.
** Otherwise similar to above.
** All of these read/write function return number of items read/written.
*/
sf_count_t sf_read_short (SNDFILE *sndfile, short *ptr, sf_count_t items) ;
sf_count_t sf_write_short (SNDFILE *sndfile, const short *ptr, sf_count_t items) ;
sf_count_t sf_read_int (SNDFILE *sndfile, int *ptr, sf_count_t items) ;
sf_count_t sf_write_int (SNDFILE *sndfile, const int *ptr, sf_count_t items) ;
sf_count_t sf_read_float (SNDFILE *sndfile, float *ptr, sf_count_t items) ;
sf_count_t sf_write_float (SNDFILE *sndfile, const float *ptr, sf_count_t items) ;
sf_count_t sf_read_double (SNDFILE *sndfile, double *ptr, sf_count_t items) ;
sf_count_t sf_write_double (SNDFILE *sndfile, const double *ptr, sf_count_t items) ;
/* Close the SNDFILE and clean up all memory allocations associated with this
** file.
** Returns 0 on success, or an error number.
*/
int sf_close (SNDFILE *sndfile) ;
/* If the file is opened SFM_WRITE or SFM_RDWR, call fsync() on the file
** to force the writing of data to disk. If the file is opened SFM_READ
** no action is taken.
*/
void sf_write_sync (SNDFILE *sndfile) ;
/* The function sf_wchar_open() is Windows Only!
** Open a file passing in a Windows Unicode filename. Otherwise, this is
** the same as sf_open().
**
** In order for this to work, you need to do the following:
**
** #include <windows.h>
** #define ENABLE_SNDFILE_WINDOWS_PROTOTYPES 1
** #including <sndfile.h>
*/
#if ENABLE_SNDFILE_WINDOWS_PROTOTYPES
SNDFILE* sf_wchar_open (LPCWSTR wpath, int mode, SF_INFO *sfinfo) ;
#endif
#ifdef __cplusplus
} /* extern "C" */
#endif /* __cplusplus */
#endif /* SNDFILE_H */

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@ -1,666 +0,0 @@
/*
** Copyright (C) 1999-2011Erik de Castro Lopo <erikd@mega-nerd.com>
**
** This program is free software; you can redistribute it and/or modify
** it under the terms of the GNU Lesser General Public License as published by
** the Free Software Foundation; either version 2.1 of the License, or
** (at your option) any later version.
**
** This program is distributed in the hope that it will be useful,
** but WITHOUT ANY WARRANTY; without even the implied warranty of
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
** GNU Lesser General Public License for more details.
**
** You should have received a copy of the GNU Lesser General Public License
** along with this program; if not, write to the Free Software
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
*/
/*
** sndfile.h -- system-wide definitions
**
** API documentation is in the doc/ directory of the source code tarball
** and at http://www.mega-nerd.com/libsndfile/api.html.
*/
#ifndef SNDFILE_H
#define SNDFILE_H
/* This is the version 1.0.X header file. */
#define SNDFILE_1
#include <stdio.h>
#include <sys/types.h>
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
/* The following file types can be read and written.
** A file type would consist of a major type (ie SF_FORMAT_WAV) bitwise
** ORed with a minor type (ie SF_FORMAT_PCM). SF_FORMAT_TYPEMASK and
** SF_FORMAT_SUBMASK can be used to separate the major and minor file
** types.
*/
enum
{ /* Major formats. */
SF_FORMAT_WAV = 0x010000, /* Microsoft WAV format (little endian default). */
SF_FORMAT_AIFF = 0x020000, /* Apple/SGI AIFF format (big endian). */
SF_FORMAT_AU = 0x030000, /* Sun/NeXT AU format (big endian). */
SF_FORMAT_RAW = 0x040000, /* RAW PCM data. */
SF_FORMAT_PAF = 0x050000, /* Ensoniq PARIS file format. */
SF_FORMAT_SVX = 0x060000, /* Amiga IFF / SVX8 / SV16 format. */
SF_FORMAT_NIST = 0x070000, /* Sphere NIST format. */
SF_FORMAT_VOC = 0x080000, /* VOC files. */
SF_FORMAT_IRCAM = 0x0A0000, /* Berkeley/IRCAM/CARL */
SF_FORMAT_W64 = 0x0B0000, /* Sonic Foundry's 64 bit RIFF/WAV */
SF_FORMAT_MAT4 = 0x0C0000, /* Matlab (tm) V4.2 / GNU Octave 2.0 */
SF_FORMAT_MAT5 = 0x0D0000, /* Matlab (tm) V5.0 / GNU Octave 2.1 */
SF_FORMAT_PVF = 0x0E0000, /* Portable Voice Format */
SF_FORMAT_XI = 0x0F0000, /* Fasttracker 2 Extended Instrument */
SF_FORMAT_HTK = 0x100000, /* HMM Tool Kit format */
SF_FORMAT_SDS = 0x110000, /* Midi Sample Dump Standard */
SF_FORMAT_AVR = 0x120000, /* Audio Visual Research */
SF_FORMAT_WAVEX = 0x130000, /* MS WAVE with WAVEFORMATEX */
SF_FORMAT_SD2 = 0x160000, /* Sound Designer 2 */
SF_FORMAT_FLAC = 0x170000, /* FLAC lossless file format */
SF_FORMAT_CAF = 0x180000, /* Core Audio File format */
SF_FORMAT_WVE = 0x190000, /* Psion WVE format */
SF_FORMAT_OGG = 0x200000, /* Xiph OGG container */
SF_FORMAT_MPC2K = 0x210000, /* Akai MPC 2000 sampler */
SF_FORMAT_RF64 = 0x220000, /* RF64 WAV file */
/* Subtypes from here on. */
SF_FORMAT_PCM_S8 = 0x0001, /* Signed 8 bit data */
SF_FORMAT_PCM_16 = 0x0002, /* Signed 16 bit data */
SF_FORMAT_PCM_24 = 0x0003, /* Signed 24 bit data */
SF_FORMAT_PCM_32 = 0x0004, /* Signed 32 bit data */
SF_FORMAT_PCM_U8 = 0x0005, /* Unsigned 8 bit data (WAV and RAW only) */
SF_FORMAT_FLOAT = 0x0006, /* 32 bit float data */
SF_FORMAT_DOUBLE = 0x0007, /* 64 bit float data */
SF_FORMAT_ULAW = 0x0010, /* U-Law encoded. */
SF_FORMAT_ALAW = 0x0011, /* A-Law encoded. */
SF_FORMAT_IMA_ADPCM = 0x0012, /* IMA ADPCM. */
SF_FORMAT_MS_ADPCM = 0x0013, /* Microsoft ADPCM. */
SF_FORMAT_GSM610 = 0x0020, /* GSM 6.10 encoding. */
SF_FORMAT_VOX_ADPCM = 0x0021, /* OKI / Dialogix ADPCM */
SF_FORMAT_G721_32 = 0x0030, /* 32kbs G721 ADPCM encoding. */
SF_FORMAT_G723_24 = 0x0031, /* 24kbs G723 ADPCM encoding. */
SF_FORMAT_G723_40 = 0x0032, /* 40kbs G723 ADPCM encoding. */
SF_FORMAT_DWVW_12 = 0x0040, /* 12 bit Delta Width Variable Word encoding. */
SF_FORMAT_DWVW_16 = 0x0041, /* 16 bit Delta Width Variable Word encoding. */
SF_FORMAT_DWVW_24 = 0x0042, /* 24 bit Delta Width Variable Word encoding. */
SF_FORMAT_DWVW_N = 0x0043, /* N bit Delta Width Variable Word encoding. */
SF_FORMAT_DPCM_8 = 0x0050, /* 8 bit differential PCM (XI only) */
SF_FORMAT_DPCM_16 = 0x0051, /* 16 bit differential PCM (XI only) */
SF_FORMAT_VORBIS = 0x0060, /* Xiph Vorbis encoding. */
/* Endian-ness options. */
SF_ENDIAN_FILE = 0x00000000, /* Default file endian-ness. */
SF_ENDIAN_LITTLE = 0x10000000, /* Force little endian-ness. */
SF_ENDIAN_BIG = 0x20000000, /* Force big endian-ness. */
SF_ENDIAN_CPU = 0x30000000, /* Force CPU endian-ness. */
SF_FORMAT_SUBMASK = 0x0000FFFF,
SF_FORMAT_TYPEMASK = 0x0FFF0000,
SF_FORMAT_ENDMASK = 0x30000000
} ;
/*
** The following are the valid command numbers for the sf_command()
** interface. The use of these commands is documented in the file
** command.html in the doc directory of the source code distribution.
*/
enum
{ SFC_GET_LIB_VERSION = 0x1000,
SFC_GET_LOG_INFO = 0x1001,
SFC_GET_CURRENT_SF_INFO = 0x1002,
SFC_GET_NORM_DOUBLE = 0x1010,
SFC_GET_NORM_FLOAT = 0x1011,
SFC_SET_NORM_DOUBLE = 0x1012,
SFC_SET_NORM_FLOAT = 0x1013,
SFC_SET_SCALE_FLOAT_INT_READ = 0x1014,
SFC_SET_SCALE_INT_FLOAT_WRITE = 0x1015,
SFC_GET_SIMPLE_FORMAT_COUNT = 0x1020,
SFC_GET_SIMPLE_FORMAT = 0x1021,
SFC_GET_FORMAT_INFO = 0x1028,
SFC_GET_FORMAT_MAJOR_COUNT = 0x1030,
SFC_GET_FORMAT_MAJOR = 0x1031,
SFC_GET_FORMAT_SUBTYPE_COUNT = 0x1032,
SFC_GET_FORMAT_SUBTYPE = 0x1033,
SFC_CALC_SIGNAL_MAX = 0x1040,
SFC_CALC_NORM_SIGNAL_MAX = 0x1041,
SFC_CALC_MAX_ALL_CHANNELS = 0x1042,
SFC_CALC_NORM_MAX_ALL_CHANNELS = 0x1043,
SFC_GET_SIGNAL_MAX = 0x1044,
SFC_GET_MAX_ALL_CHANNELS = 0x1045,
SFC_SET_ADD_PEAK_CHUNK = 0x1050,
SFC_SET_ADD_HEADER_PAD_CHUNK = 0x1051,
SFC_UPDATE_HEADER_NOW = 0x1060,
SFC_SET_UPDATE_HEADER_AUTO = 0x1061,
SFC_FILE_TRUNCATE = 0x1080,
SFC_SET_RAW_START_OFFSET = 0x1090,
SFC_SET_DITHER_ON_WRITE = 0x10A0,
SFC_SET_DITHER_ON_READ = 0x10A1,
SFC_GET_DITHER_INFO_COUNT = 0x10A2,
SFC_GET_DITHER_INFO = 0x10A3,
SFC_GET_EMBED_FILE_INFO = 0x10B0,
SFC_SET_CLIPPING = 0x10C0,
SFC_GET_CLIPPING = 0x10C1,
SFC_GET_INSTRUMENT = 0x10D0,
SFC_SET_INSTRUMENT = 0x10D1,
SFC_GET_LOOP_INFO = 0x10E0,
SFC_GET_BROADCAST_INFO = 0x10F0,
SFC_SET_BROADCAST_INFO = 0x10F1,
SFC_GET_CHANNEL_MAP_INFO = 0x1100,
SFC_SET_CHANNEL_MAP_INFO = 0x1101,
SFC_RAW_DATA_NEEDS_ENDSWAP = 0x1110,
/* Support for Wavex Ambisonics Format */
SFC_WAVEX_SET_AMBISONIC = 0x1200,
SFC_WAVEX_GET_AMBISONIC = 0x1201,
SFC_SET_VBR_ENCODING_QUALITY = 0x1300,
/* Following commands for testing only. */
SFC_TEST_IEEE_FLOAT_REPLACE = 0x6001,
/*
** SFC_SET_ADD_* values are deprecated and will disappear at some
** time in the future. They are guaranteed to be here up to and
** including version 1.0.8 to avoid breakage of existng software.
** They currently do nothing and will continue to do nothing.
*/
SFC_SET_ADD_DITHER_ON_WRITE = 0x1070,
SFC_SET_ADD_DITHER_ON_READ = 0x1071
} ;
/*
** String types that can be set and read from files. Not all file types
** support this and even the file types which support one, may not support
** all string types.
*/
enum
{ SF_STR_TITLE = 0x01,
SF_STR_COPYRIGHT = 0x02,
SF_STR_SOFTWARE = 0x03,
SF_STR_ARTIST = 0x04,
SF_STR_COMMENT = 0x05,
SF_STR_DATE = 0x06,
SF_STR_ALBUM = 0x07,
SF_STR_LICENSE = 0x08,
SF_STR_TRACKNUMBER = 0x09,
SF_STR_GENRE = 0x10
} ;
/*
** Use the following as the start and end index when doing metadata
** transcoding.
*/
#define SF_STR_FIRST SF_STR_TITLE
#define SF_STR_LAST SF_STR_LICENSE
enum
{ /* True and false */
SF_FALSE = 0,
SF_TRUE = 1,
/* Modes for opening files. */
SFM_READ = 0x10,
SFM_WRITE = 0x20,
SFM_RDWR = 0x30,
SF_AMBISONIC_NONE = 0x40,
SF_AMBISONIC_B_FORMAT = 0x41
} ;
/* Public error values. These are guaranteed to remain unchanged for the duration
** of the library major version number.
** There are also a large number of private error numbers which are internal to
** the library which can change at any time.
*/
enum
{ SF_ERR_NO_ERROR = 0,
SF_ERR_UNRECOGNISED_FORMAT = 1,
SF_ERR_SYSTEM = 2,
SF_ERR_MALFORMED_FILE = 3,
SF_ERR_UNSUPPORTED_ENCODING = 4
} ;
/* Channel map values (used with SFC_SET/GET_CHANNEL_MAP).
*/
enum
{ SF_CHANNEL_MAP_INVALID = 0,
SF_CHANNEL_MAP_MONO = 1,
SF_CHANNEL_MAP_LEFT, /* Apple calls this 'Left' */
SF_CHANNEL_MAP_RIGHT, /* Apple calls this 'Right' */
SF_CHANNEL_MAP_CENTER, /* Apple calls this 'Center' */
SF_CHANNEL_MAP_FRONT_LEFT,
SF_CHANNEL_MAP_FRONT_RIGHT,
SF_CHANNEL_MAP_FRONT_CENTER,
SF_CHANNEL_MAP_REAR_CENTER, /* Apple calls this 'Center Surround', Msft calls this 'Back Center' */
SF_CHANNEL_MAP_REAR_LEFT, /* Apple calls this 'Left Surround', Msft calls this 'Back Left' */
SF_CHANNEL_MAP_REAR_RIGHT, /* Apple calls this 'Right Surround', Msft calls this 'Back Right' */
SF_CHANNEL_MAP_LFE, /* Apple calls this 'LFEScreen', Msft calls this 'Low Frequency' */
SF_CHANNEL_MAP_FRONT_LEFT_OF_CENTER, /* Apple calls this 'Left Center' */
SF_CHANNEL_MAP_FRONT_RIGHT_OF_CENTER, /* Apple calls this 'Right Center */
SF_CHANNEL_MAP_SIDE_LEFT, /* Apple calls this 'Left Surround Direct' */
SF_CHANNEL_MAP_SIDE_RIGHT, /* Apple calls this 'Right Surround Direct' */
SF_CHANNEL_MAP_TOP_CENTER, /* Apple calls this 'Top Center Surround' */
SF_CHANNEL_MAP_TOP_FRONT_LEFT, /* Apple calls this 'Vertical Height Left' */
SF_CHANNEL_MAP_TOP_FRONT_RIGHT, /* Apple calls this 'Vertical Height Right' */
SF_CHANNEL_MAP_TOP_FRONT_CENTER, /* Apple calls this 'Vertical Height Center' */
SF_CHANNEL_MAP_TOP_REAR_LEFT, /* Apple and MS call this 'Top Back Left' */
SF_CHANNEL_MAP_TOP_REAR_RIGHT, /* Apple and MS call this 'Top Back Right' */
SF_CHANNEL_MAP_TOP_REAR_CENTER, /* Apple and MS call this 'Top Back Center' */
SF_CHANNEL_MAP_AMBISONIC_B_W,
SF_CHANNEL_MAP_AMBISONIC_B_X,
SF_CHANNEL_MAP_AMBISONIC_B_Y,
SF_CHANNEL_MAP_AMBISONIC_B_Z,
SF_CHANNEL_MAP_MAX
} ;
/* A SNDFILE* pointer can be passed around much like stdio.h's FILE* pointer. */
typedef struct SNDFILE_tag SNDFILE ;
/* The following typedef is system specific and is defined when libsndfile is
** compiled. sf_count_t will be a 64 bit value when the underlying OS allows
** 64 bit file offsets.
** On windows, we need to allow the same header file to be compiler by both GCC
** and the Microsoft compiler.
*/
#if (defined (_MSCVER) || defined (_MSC_VER))
typedef __int64 sf_count_t ;
#define SF_COUNT_MAX 0x7fffffffffffffffi64
#else
typedef __int64 sf_count_t ;
#define SF_COUNT_MAX 0x7FFFFFFFFFFFFFFFLL
#endif
/* A pointer to a SF_INFO structure is passed to sf_open () and filled in.
** On write, the SF_INFO structure is filled in by the user and passed into
** sf_open ().
*/
struct SF_INFO
{ sf_count_t frames ; /* Used to be called samples. Changed to avoid confusion. */
int samplerate ;
int channels ;
int format ;
int sections ;
int seekable ;
} ;
typedef struct SF_INFO SF_INFO ;
/* The SF_FORMAT_INFO struct is used to retrieve information about the sound
** file formats libsndfile supports using the sf_command () interface.
**
** Using this interface will allow applications to support new file formats
** and encoding types when libsndfile is upgraded, without requiring
** re-compilation of the application.
**
** Please consult the libsndfile documentation (particularly the information
** on the sf_command () interface) for examples of its use.
*/
typedef struct
{ int format ;
const char *name ;
const char *extension ;
} SF_FORMAT_INFO ;
/*
** Enums and typedefs for adding dither on read and write.
** See the html documentation for sf_command(), SFC_SET_DITHER_ON_WRITE
** and SFC_SET_DITHER_ON_READ.
*/
enum
{ SFD_DEFAULT_LEVEL = 0,
SFD_CUSTOM_LEVEL = 0x40000000,
SFD_NO_DITHER = 500,
SFD_WHITE = 501,
SFD_TRIANGULAR_PDF = 502
} ;
typedef struct
{ int type ;
double level ;
const char *name ;
} SF_DITHER_INFO ;
/* Struct used to retrieve information about a file embedded within a
** larger file. See SFC_GET_EMBED_FILE_INFO.
*/
typedef struct
{ sf_count_t offset ;
sf_count_t length ;
} SF_EMBED_FILE_INFO ;
/*
** Structs used to retrieve music sample information from a file.
*/
enum
{ /*
** The loop mode field in SF_INSTRUMENT will be one of the following.
*/
SF_LOOP_NONE = 800,
SF_LOOP_FORWARD,
SF_LOOP_BACKWARD,
SF_LOOP_ALTERNATING
} ;
typedef struct
{ int gain ;
char basenote, detune ;
char velocity_lo, velocity_hi ;
char key_lo, key_hi ;
int loop_count ;
struct
{ int mode ;
unsigned int start ;
unsigned int end ;
unsigned int count ;
} loops [16] ; /* make variable in a sensible way */
} SF_INSTRUMENT ;
/* Struct used to retrieve loop information from a file.*/
typedef struct
{
short time_sig_num ; /* any positive integer > 0 */
short time_sig_den ; /* any positive power of 2 > 0 */
int loop_mode ; /* see SF_LOOP enum */
int num_beats ; /* this is NOT the amount of quarter notes !!!*/
/* a full bar of 4/4 is 4 beats */
/* a full bar of 7/8 is 7 beats */
float bpm ; /* suggestion, as it can be calculated using other fields:*/
/* file's lenght, file's sampleRate and our time_sig_den*/
/* -> bpms are always the amount of _quarter notes_ per minute */
int root_key ; /* MIDI note, or -1 for None */
int future [6] ;
} SF_LOOP_INFO ;
/* Struct used to retrieve broadcast (EBU) information from a file.
** Strongly (!) based on EBU "bext" chunk format used in Broadcast WAVE.
*/
#define SF_BROADCAST_INFO_VAR(coding_hist_size) \
struct \
{ char description [256] ; \
char originator [32] ; \
char originator_reference [32] ; \
char origination_date [10] ; \
char origination_time [8] ; \
unsigned int time_reference_low ; \
unsigned int time_reference_high ; \
short version ; \
char umid [64] ; \
char reserved [190] ; \
unsigned int coding_history_size ; \
char coding_history [coding_hist_size] ; \
}
/* SF_BROADCAST_INFO is the above struct with coding_history field of 256 bytes. */
typedef SF_BROADCAST_INFO_VAR (256) SF_BROADCAST_INFO ;
/* Virtual I/O functionality. */
typedef sf_count_t (*sf_vio_get_filelen) (void *user_data) ;
typedef sf_count_t (*sf_vio_seek) (sf_count_t offset, int whence, void *user_data) ;
typedef sf_count_t (*sf_vio_read) (void *ptr, sf_count_t count, void *user_data) ;
typedef sf_count_t (*sf_vio_write) (const void *ptr, sf_count_t count, void *user_data) ;
typedef sf_count_t (*sf_vio_tell) (void *user_data) ;
struct SF_VIRTUAL_IO
{ sf_vio_get_filelen get_filelen ;
sf_vio_seek seek ;
sf_vio_read read ;
sf_vio_write write ;
sf_vio_tell tell ;
} ;
typedef struct SF_VIRTUAL_IO SF_VIRTUAL_IO ;
/* Open the specified file for read, write or both. On error, this will
** return a NULL pointer. To find the error number, pass a NULL SNDFILE
** to sf_strerror ().
** All calls to sf_open() should be matched with a call to sf_close().
*/
SNDFILE* sf_open (const char *path, int mode, SF_INFO *sfinfo) ;
/* Use the existing file descriptor to create a SNDFILE object. If close_desc
** is TRUE, the file descriptor will be closed when sf_close() is called. If
** it is FALSE, the descritor will not be closed.
** When passed a descriptor like this, the library will assume that the start
** of file header is at the current file offset. This allows sound files within
** larger container files to be read and/or written.
** On error, this will return a NULL pointer. To find the error number, pass a
** NULL SNDFILE to sf_strerror ().
** All calls to sf_open_fd() should be matched with a call to sf_close().
*/
SNDFILE* sf_open_fd (int fd, int mode, SF_INFO *sfinfo, int close_desc) ;
SNDFILE* sf_open_virtual (SF_VIRTUAL_IO *sfvirtual, int mode, SF_INFO *sfinfo, void *user_data) ;
/* sf_error () returns a error number which can be translated to a text
** string using sf_error_number().
*/
int sf_error (SNDFILE *sndfile) ;
/* sf_strerror () returns to the caller a pointer to the current error message for
** the given SNDFILE.
*/
const char* sf_strerror (SNDFILE *sndfile) ;
/* sf_error_number () allows the retrieval of the error string for each internal
** error number.
**
*/
const char* sf_error_number (int errnum) ;
/* The following two error functions are deprecated but they will remain in the
** library for the forseeable future. The function sf_strerror() should be used
** in their place.
*/
int sf_perror (SNDFILE *sndfile) ;
int sf_error_str (SNDFILE *sndfile, char* str, size_t len) ;
/* Return TRUE if fields of the SF_INFO struct are a valid combination of values. */
int sf_command (SNDFILE *sndfile, int command, void *data, int datasize) ;
/* Return TRUE if fields of the SF_INFO struct are a valid combination of values. */
int sf_format_check (const SF_INFO *info) ;
/* Seek within the waveform data chunk of the SNDFILE. sf_seek () uses
** the same values for whence (SEEK_SET, SEEK_CUR and SEEK_END) as
** stdio.h function fseek ().
** An offset of zero with whence set to SEEK_SET will position the
** read / write pointer to the first data sample.
** On success sf_seek returns the current position in (multi-channel)
** samples from the start of the file.
** Please see the libsndfile documentation for moving the read pointer
** separately from the write pointer on files open in mode SFM_RDWR.
** On error all of these functions return -1.
*/
sf_count_t sf_seek (SNDFILE *sndfile, sf_count_t frames, int whence) ;
/* Functions for retrieving and setting string data within sound files.
** Not all file types support this features; AIFF and WAV do. For both
** functions, the str_type parameter must be one of the SF_STR_* values
** defined above.
** On error, sf_set_string() returns non-zero while sf_get_string()
** returns NULL.
*/
int sf_set_string (SNDFILE *sndfile, int str_type, const char* str) ;
const char* sf_get_string (SNDFILE *sndfile, int str_type) ;
/* Return the library version string. */
const char * sf_version_string (void) ;
/* Functions for reading/writing the waveform data of a sound file.
*/
sf_count_t sf_read_raw (SNDFILE *sndfile, void *ptr, sf_count_t bytes) ;
sf_count_t sf_write_raw (SNDFILE *sndfile, const void *ptr, sf_count_t bytes) ;
/* Functions for reading and writing the data chunk in terms of frames.
** The number of items actually read/written = frames * number of channels.
** sf_xxxx_raw read/writes the raw data bytes from/to the file
** sf_xxxx_short passes data in the native short format
** sf_xxxx_int passes data in the native int format
** sf_xxxx_float passes data in the native float format
** sf_xxxx_double passes data in the native double format
** All of these read/write function return number of frames read/written.
*/
sf_count_t sf_readf_short (SNDFILE *sndfile, short *ptr, sf_count_t frames) ;
sf_count_t sf_writef_short (SNDFILE *sndfile, const short *ptr, sf_count_t frames) ;
sf_count_t sf_readf_int (SNDFILE *sndfile, int *ptr, sf_count_t frames) ;
sf_count_t sf_writef_int (SNDFILE *sndfile, const int *ptr, sf_count_t frames) ;
sf_count_t sf_readf_float (SNDFILE *sndfile, float *ptr, sf_count_t frames) ;
sf_count_t sf_writef_float (SNDFILE *sndfile, const float *ptr, sf_count_t frames) ;
sf_count_t sf_readf_double (SNDFILE *sndfile, double *ptr, sf_count_t frames) ;
sf_count_t sf_writef_double (SNDFILE *sndfile, const double *ptr, sf_count_t frames) ;
/* Functions for reading and writing the data chunk in terms of items.
** Otherwise similar to above.
** All of these read/write function return number of items read/written.
*/
sf_count_t sf_read_short (SNDFILE *sndfile, short *ptr, sf_count_t items) ;
sf_count_t sf_write_short (SNDFILE *sndfile, const short *ptr, sf_count_t items) ;
sf_count_t sf_read_int (SNDFILE *sndfile, int *ptr, sf_count_t items) ;
sf_count_t sf_write_int (SNDFILE *sndfile, const int *ptr, sf_count_t items) ;
sf_count_t sf_read_float (SNDFILE *sndfile, float *ptr, sf_count_t items) ;
sf_count_t sf_write_float (SNDFILE *sndfile, const float *ptr, sf_count_t items) ;
sf_count_t sf_read_double (SNDFILE *sndfile, double *ptr, sf_count_t items) ;
sf_count_t sf_write_double (SNDFILE *sndfile, const double *ptr, sf_count_t items) ;
/* Close the SNDFILE and clean up all memory allocations associated with this
** file.
** Returns 0 on success, or an error number.
*/
int sf_close (SNDFILE *sndfile) ;
/* If the file is opened SFM_WRITE or SFM_RDWR, call fsync() on the file
** to force the writing of data to disk. If the file is opened SFM_READ
** no action is taken.
*/
void sf_write_sync (SNDFILE *sndfile) ;
/* The function sf_wchar_open() is Windows Only!
** Open a file passing in a Windows Unicode filename. Otherwise, this is
** the same as sf_open().
**
** In order for this to work, you need to do the following:
**
** #include <windows.h>
** #define ENABLE_SNDFILE_WINDOWS_PROTOTYPES 1
** #including <sndfile.h>
*/
#if ENABLE_SNDFILE_WINDOWS_PROTOTYPES
SNDFILE* sf_wchar_open (LPCWSTR wpath, int mode, SF_INFO *sfinfo) ;
#endif
#ifdef __cplusplus
} /* extern "C" */
#endif /* __cplusplus */
#endif /* SNDFILE_H */

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@ -0,0 +1,210 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: toplevel libogg include
last mod: $Id: ogg.h 18044 2011-08-01 17:55:20Z gmaxwell $
********************************************************************/
#ifndef _OGG_H
#define _OGG_H
#ifdef __cplusplus
extern "C" {
#endif
#include <stddef.h>
#include <ogg/os_types.h>
typedef struct {
void *iov_base;
size_t iov_len;
} ogg_iovec_t;
typedef struct {
long endbyte;
int endbit;
unsigned char *buffer;
unsigned char *ptr;
long storage;
} oggpack_buffer;
/* ogg_page is used to encapsulate the data in one Ogg bitstream page *****/
typedef struct {
unsigned char *header;
long header_len;
unsigned char *body;
long body_len;
} ogg_page;
/* ogg_stream_state contains the current encode/decode state of a logical
Ogg bitstream **********************************************************/
typedef struct {
unsigned char *body_data; /* bytes from packet bodies */
long body_storage; /* storage elements allocated */
long body_fill; /* elements stored; fill mark */
long body_returned; /* elements of fill returned */
int *lacing_vals; /* The values that will go to the segment table */
ogg_int64_t *granule_vals; /* granulepos values for headers. Not compact
this way, but it is simple coupled to the
lacing fifo */
long lacing_storage;
long lacing_fill;
long lacing_packet;
long lacing_returned;
unsigned char header[282]; /* working space for header encode */
int header_fill;
int e_o_s; /* set when we have buffered the last packet in the
logical bitstream */
int b_o_s; /* set after we've written the initial page
of a logical bitstream */
long serialno;
long pageno;
ogg_int64_t packetno; /* sequence number for decode; the framing
knows where there's a hole in the data,
but we need coupling so that the codec
(which is in a separate abstraction
layer) also knows about the gap */
ogg_int64_t granulepos;
} ogg_stream_state;
/* ogg_packet is used to encapsulate the data and metadata belonging
to a single raw Ogg/Vorbis packet *************************************/
typedef struct {
unsigned char *packet;
long bytes;
long b_o_s;
long e_o_s;
ogg_int64_t granulepos;
ogg_int64_t packetno; /* sequence number for decode; the framing
knows where there's a hole in the data,
but we need coupling so that the codec
(which is in a separate abstraction
layer) also knows about the gap */
} ogg_packet;
typedef struct {
unsigned char *data;
int storage;
int fill;
int returned;
int unsynced;
int headerbytes;
int bodybytes;
} ogg_sync_state;
/* Ogg BITSTREAM PRIMITIVES: bitstream ************************/
extern void oggpack_writeinit(oggpack_buffer *b);
extern int oggpack_writecheck(oggpack_buffer *b);
extern void oggpack_writetrunc(oggpack_buffer *b,long bits);
extern void oggpack_writealign(oggpack_buffer *b);
extern void oggpack_writecopy(oggpack_buffer *b,void *source,long bits);
extern void oggpack_reset(oggpack_buffer *b);
extern void oggpack_writeclear(oggpack_buffer *b);
extern void oggpack_readinit(oggpack_buffer *b,unsigned char *buf,int bytes);
extern void oggpack_write(oggpack_buffer *b,unsigned long value,int bits);
extern long oggpack_look(oggpack_buffer *b,int bits);
extern long oggpack_look1(oggpack_buffer *b);
extern void oggpack_adv(oggpack_buffer *b,int bits);
extern void oggpack_adv1(oggpack_buffer *b);
extern long oggpack_read(oggpack_buffer *b,int bits);
extern long oggpack_read1(oggpack_buffer *b);
extern long oggpack_bytes(oggpack_buffer *b);
extern long oggpack_bits(oggpack_buffer *b);
extern unsigned char *oggpack_get_buffer(oggpack_buffer *b);
extern void oggpackB_writeinit(oggpack_buffer *b);
extern int oggpackB_writecheck(oggpack_buffer *b);
extern void oggpackB_writetrunc(oggpack_buffer *b,long bits);
extern void oggpackB_writealign(oggpack_buffer *b);
extern void oggpackB_writecopy(oggpack_buffer *b,void *source,long bits);
extern void oggpackB_reset(oggpack_buffer *b);
extern void oggpackB_writeclear(oggpack_buffer *b);
extern void oggpackB_readinit(oggpack_buffer *b,unsigned char *buf,int bytes);
extern void oggpackB_write(oggpack_buffer *b,unsigned long value,int bits);
extern long oggpackB_look(oggpack_buffer *b,int bits);
extern long oggpackB_look1(oggpack_buffer *b);
extern void oggpackB_adv(oggpack_buffer *b,int bits);
extern void oggpackB_adv1(oggpack_buffer *b);
extern long oggpackB_read(oggpack_buffer *b,int bits);
extern long oggpackB_read1(oggpack_buffer *b);
extern long oggpackB_bytes(oggpack_buffer *b);
extern long oggpackB_bits(oggpack_buffer *b);
extern unsigned char *oggpackB_get_buffer(oggpack_buffer *b);
/* Ogg BITSTREAM PRIMITIVES: encoding **************************/
extern int ogg_stream_packetin(ogg_stream_state *os, ogg_packet *op);
extern int ogg_stream_iovecin(ogg_stream_state *os, ogg_iovec_t *iov,
int count, long e_o_s, ogg_int64_t granulepos);
extern int ogg_stream_pageout(ogg_stream_state *os, ogg_page *og);
extern int ogg_stream_pageout_fill(ogg_stream_state *os, ogg_page *og, int nfill);
extern int ogg_stream_flush(ogg_stream_state *os, ogg_page *og);
extern int ogg_stream_flush_fill(ogg_stream_state *os, ogg_page *og, int nfill);
/* Ogg BITSTREAM PRIMITIVES: decoding **************************/
extern int ogg_sync_init(ogg_sync_state *oy);
extern int ogg_sync_clear(ogg_sync_state *oy);
extern int ogg_sync_reset(ogg_sync_state *oy);
extern int ogg_sync_destroy(ogg_sync_state *oy);
extern int ogg_sync_check(ogg_sync_state *oy);
extern char *ogg_sync_buffer(ogg_sync_state *oy, long size);
extern int ogg_sync_wrote(ogg_sync_state *oy, long bytes);
extern long ogg_sync_pageseek(ogg_sync_state *oy,ogg_page *og);
extern int ogg_sync_pageout(ogg_sync_state *oy, ogg_page *og);
extern int ogg_stream_pagein(ogg_stream_state *os, ogg_page *og);
extern int ogg_stream_packetout(ogg_stream_state *os,ogg_packet *op);
extern int ogg_stream_packetpeek(ogg_stream_state *os,ogg_packet *op);
/* Ogg BITSTREAM PRIMITIVES: general ***************************/
extern int ogg_stream_init(ogg_stream_state *os,int serialno);
extern int ogg_stream_clear(ogg_stream_state *os);
extern int ogg_stream_reset(ogg_stream_state *os);
extern int ogg_stream_reset_serialno(ogg_stream_state *os,int serialno);
extern int ogg_stream_destroy(ogg_stream_state *os);
extern int ogg_stream_check(ogg_stream_state *os);
extern int ogg_stream_eos(ogg_stream_state *os);
extern void ogg_page_checksum_set(ogg_page *og);
extern int ogg_page_version(const ogg_page *og);
extern int ogg_page_continued(const ogg_page *og);
extern int ogg_page_bos(const ogg_page *og);
extern int ogg_page_eos(const ogg_page *og);
extern ogg_int64_t ogg_page_granulepos(const ogg_page *og);
extern int ogg_page_serialno(const ogg_page *og);
extern long ogg_page_pageno(const ogg_page *og);
extern int ogg_page_packets(const ogg_page *og);
extern void ogg_packet_clear(ogg_packet *op);
#ifdef __cplusplus
}
#endif
#endif /* _OGG_H */

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/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: #ifdef jail to whip a few platforms into the UNIX ideal.
last mod: $Id: os_types.h 19098 2014-02-26 19:06:45Z giles $
********************************************************************/
#ifndef _OS_TYPES_H
#define _OS_TYPES_H
/* make it easy on the folks that want to compile the libs with a
different malloc than stdlib */
#define _ogg_malloc malloc
#define _ogg_calloc calloc
#define _ogg_realloc realloc
#define _ogg_free free
#if defined(_WIN32)
# if defined(__CYGWIN__)
# include <stdint.h>
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef uint64_t ogg_uint64_t;
# elif defined(__MINGW32__)
# include <sys/types.h>
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
# elif defined(__MWERKS__)
typedef long long ogg_int64_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
# else
/* MSVC/Borland */
typedef __int64 ogg_int64_t;
typedef __int32 ogg_int32_t;
typedef unsigned __int32 ogg_uint32_t;
typedef __int16 ogg_int16_t;
typedef unsigned __int16 ogg_uint16_t;
# endif
#elif defined(__MACOS__)
# include <sys/types.h>
typedef SInt16 ogg_int16_t;
typedef UInt16 ogg_uint16_t;
typedef SInt32 ogg_int32_t;
typedef UInt32 ogg_uint32_t;
typedef SInt64 ogg_int64_t;
#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
# include <inttypes.h>
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
#elif defined(__HAIKU__)
/* Haiku */
# include <sys/types.h>
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
#elif defined(__BEOS__)
/* Be */
# include <inttypes.h>
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
#elif defined (__EMX__)
/* OS/2 GCC */
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
#elif defined (DJGPP)
/* DJGPP */
typedef short ogg_int16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
#elif defined(R5900)
/* PS2 EE */
typedef long ogg_int64_t;
typedef int ogg_int32_t;
typedef unsigned ogg_uint32_t;
typedef short ogg_int16_t;
#elif defined(__SYMBIAN32__)
/* Symbian GCC */
typedef signed short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef signed int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long int ogg_int64_t;
#elif defined(__TMS320C6X__)
/* TI C64x compiler */
typedef signed short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef signed int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long int ogg_int64_t;
#else
# include <ogg/config_types.h>
#endif
#endif /* _OS_TYPES_H */

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/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2001 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
********************************************************************
function: libvorbis codec headers
last mod: $Id: codec.h 17021 2010-03-24 09:29:41Z xiphmont $
********************************************************************/
#ifndef _vorbis_codec_h_
#define _vorbis_codec_h_
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
#include <ogg/ogg.h>
typedef struct vorbis_info{
int version;
int channels;
long rate;
/* The below bitrate declarations are *hints*.
Combinations of the three values carry the following implications:
all three set to the same value:
implies a fixed rate bitstream
only nominal set:
implies a VBR stream that averages the nominal bitrate. No hard
upper/lower limit
upper and or lower set:
implies a VBR bitstream that obeys the bitrate limits. nominal
may also be set to give a nominal rate.
none set:
the coder does not care to speculate.
*/
long bitrate_upper;
long bitrate_nominal;
long bitrate_lower;
long bitrate_window;
void *codec_setup;
} vorbis_info;
/* vorbis_dsp_state buffers the current vorbis audio
analysis/synthesis state. The DSP state belongs to a specific
logical bitstream ****************************************************/
typedef struct vorbis_dsp_state{
int analysisp;
vorbis_info *vi;
float **pcm;
float **pcmret;
int pcm_storage;
int pcm_current;
int pcm_returned;
int preextrapolate;
int eofflag;
long lW;
long W;
long nW;
long centerW;
ogg_int64_t granulepos;
ogg_int64_t sequence;
ogg_int64_t glue_bits;
ogg_int64_t time_bits;
ogg_int64_t floor_bits;
ogg_int64_t res_bits;
void *backend_state;
} vorbis_dsp_state;
typedef struct vorbis_block{
/* necessary stream state for linking to the framing abstraction */
float **pcm; /* this is a pointer into local storage */
oggpack_buffer opb;
long lW;
long W;
long nW;
int pcmend;
int mode;
int eofflag;
ogg_int64_t granulepos;
ogg_int64_t sequence;
vorbis_dsp_state *vd; /* For read-only access of configuration */
/* local storage to avoid remallocing; it's up to the mapping to
structure it */
void *localstore;
long localtop;
long localalloc;
long totaluse;
struct alloc_chain *reap;
/* bitmetrics for the frame */
long glue_bits;
long time_bits;
long floor_bits;
long res_bits;
void *internal;
} vorbis_block;
/* vorbis_block is a single block of data to be processed as part of
the analysis/synthesis stream; it belongs to a specific logical
bitstream, but is independent from other vorbis_blocks belonging to
that logical bitstream. *************************************************/
struct alloc_chain{
void *ptr;
struct alloc_chain *next;
};
/* vorbis_info contains all the setup information specific to the
specific compression/decompression mode in progress (eg,
psychoacoustic settings, channel setup, options, codebook
etc). vorbis_info and substructures are in backends.h.
*********************************************************************/
/* the comments are not part of vorbis_info so that vorbis_info can be
static storage */
typedef struct vorbis_comment{
/* unlimited user comment fields. libvorbis writes 'libvorbis'
whatever vendor is set to in encode */
char **user_comments;
int *comment_lengths;
int comments;
char *vendor;
} vorbis_comment;
/* libvorbis encodes in two abstraction layers; first we perform DSP
and produce a packet (see docs/analysis.txt). The packet is then
coded into a framed OggSquish bitstream by the second layer (see
docs/framing.txt). Decode is the reverse process; we sync/frame
the bitstream and extract individual packets, then decode the
packet back into PCM audio.
The extra framing/packetizing is used in streaming formats, such as
files. Over the net (such as with UDP), the framing and
packetization aren't necessary as they're provided by the transport
and the streaming layer is not used */
/* Vorbis PRIMITIVES: general ***************************************/
extern void vorbis_info_init(vorbis_info *vi);
extern void vorbis_info_clear(vorbis_info *vi);
extern int vorbis_info_blocksize(vorbis_info *vi,int zo);
extern void vorbis_comment_init(vorbis_comment *vc);
extern void vorbis_comment_add(vorbis_comment *vc, const char *comment);
extern void vorbis_comment_add_tag(vorbis_comment *vc,
const char *tag, const char *contents);
extern char *vorbis_comment_query(vorbis_comment *vc, const char *tag, int count);
extern int vorbis_comment_query_count(vorbis_comment *vc, const char *tag);
extern void vorbis_comment_clear(vorbis_comment *vc);
extern int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb);
extern int vorbis_block_clear(vorbis_block *vb);
extern void vorbis_dsp_clear(vorbis_dsp_state *v);
extern double vorbis_granule_time(vorbis_dsp_state *v,
ogg_int64_t granulepos);
extern const char *vorbis_version_string(void);
/* Vorbis PRIMITIVES: analysis/DSP layer ****************************/
extern int vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi);
extern int vorbis_commentheader_out(vorbis_comment *vc, ogg_packet *op);
extern int vorbis_analysis_headerout(vorbis_dsp_state *v,
vorbis_comment *vc,
ogg_packet *op,
ogg_packet *op_comm,
ogg_packet *op_code);
extern float **vorbis_analysis_buffer(vorbis_dsp_state *v,int vals);
extern int vorbis_analysis_wrote(vorbis_dsp_state *v,int vals);
extern int vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb);
extern int vorbis_analysis(vorbis_block *vb,ogg_packet *op);
extern int vorbis_bitrate_addblock(vorbis_block *vb);
extern int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd,
ogg_packet *op);
/* Vorbis PRIMITIVES: synthesis layer *******************************/
extern int vorbis_synthesis_idheader(ogg_packet *op);
extern int vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc,
ogg_packet *op);
extern int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi);
extern int vorbis_synthesis_restart(vorbis_dsp_state *v);
extern int vorbis_synthesis(vorbis_block *vb,ogg_packet *op);
extern int vorbis_synthesis_trackonly(vorbis_block *vb,ogg_packet *op);
extern int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb);
extern int vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm);
extern int vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm);
extern int vorbis_synthesis_read(vorbis_dsp_state *v,int samples);
extern long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op);
extern int vorbis_synthesis_halfrate(vorbis_info *v,int flag);
extern int vorbis_synthesis_halfrate_p(vorbis_info *v);
/* Vorbis ERRORS and return codes ***********************************/
#define OV_FALSE -1
#define OV_EOF -2
#define OV_HOLE -3
#define OV_EREAD -128
#define OV_EFAULT -129
#define OV_EIMPL -130
#define OV_EINVAL -131
#define OV_ENOTVORBIS -132
#define OV_EBADHEADER -133
#define OV_EVERSION -134
#define OV_ENOTAUDIO -135
#define OV_EBADPACKET -136
#define OV_EBADLINK -137
#define OV_ENOSEEK -138
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif

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/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2001 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: vorbis encode-engine setup
last mod: $Id: vorbisenc.h 17021 2010-03-24 09:29:41Z xiphmont $
********************************************************************/
/** \file
* Libvorbisenc is a convenient API for setting up an encoding
* environment using libvorbis. Libvorbisenc encapsulates the
* actions needed to set up the encoder properly.
*/
#ifndef _OV_ENC_H_
#define _OV_ENC_H_
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
#include "codec.h"
/**
* This is the primary function within libvorbisenc for setting up managed
* bitrate modes.
*
* Before this function is called, the \ref vorbis_info
* struct should be initialized by using vorbis_info_init() from the libvorbis
* API. After encoding, vorbis_info_clear() should be called.
*
* The max_bitrate, nominal_bitrate, and min_bitrate settings are used to set
* constraints for the encoded file. This function uses these settings to
* select the appropriate encoding mode and set it up.
*
* \param vi Pointer to an initialized \ref vorbis_info struct.
* \param channels The number of channels to be encoded.
* \param rate The sampling rate of the source audio.
* \param max_bitrate Desired maximum bitrate (limit). -1 indicates unset.
* \param nominal_bitrate Desired average, or central, bitrate. -1 indicates unset.
* \param min_bitrate Desired minimum bitrate. -1 indicates unset.
*
* \return Zero for success, and negative values for failure.
*
* \retval 0 Success.
* \retval OV_EFAULT Internal logic fault; indicates a bug or heap/stack corruption.
* \retval OV_EINVAL Invalid setup request, eg, out of range argument.
* \retval OV_EIMPL Unimplemented mode; unable to comply with bitrate request.
*/
extern int vorbis_encode_init(vorbis_info *vi,
long channels,
long rate,
long max_bitrate,
long nominal_bitrate,
long min_bitrate);
/**
* This function performs step-one of a three-step bitrate-managed encode
* setup. It functions similarly to the one-step setup performed by \ref
* vorbis_encode_init but allows an application to make further encode setup
* tweaks using \ref vorbis_encode_ctl before finally calling \ref
* vorbis_encode_setup_init to complete the setup process.
*
* Before this function is called, the \ref vorbis_info struct should be
* initialized by using vorbis_info_init() from the libvorbis API. After
* encoding, vorbis_info_clear() should be called.
*
* The max_bitrate, nominal_bitrate, and min_bitrate settings are used to set
* constraints for the encoded file. This function uses these settings to
* select the appropriate encoding mode and set it up.
*
* \param vi Pointer to an initialized vorbis_info struct.
* \param channels The number of channels to be encoded.
* \param rate The sampling rate of the source audio.
* \param max_bitrate Desired maximum bitrate (limit). -1 indicates unset.
* \param nominal_bitrate Desired average, or central, bitrate. -1 indicates unset.
* \param min_bitrate Desired minimum bitrate. -1 indicates unset.
*
* \return Zero for success, and negative for failure.
*
* \retval 0 Success
* \retval OV_EFAULT Internal logic fault; indicates a bug or heap/stack corruption.
* \retval OV_EINVAL Invalid setup request, eg, out of range argument.
* \retval OV_EIMPL Unimplemented mode; unable to comply with bitrate request.
*/
extern int vorbis_encode_setup_managed(vorbis_info *vi,
long channels,
long rate,
long max_bitrate,
long nominal_bitrate,
long min_bitrate);
/**
* This function performs step-one of a three-step variable bitrate
* (quality-based) encode setup. It functions similarly to the one-step setup
* performed by \ref vorbis_encode_init_vbr() but allows an application to
* make further encode setup tweaks using \ref vorbis_encode_ctl() before
* finally calling \ref vorbis_encode_setup_init to complete the setup
* process.
*
* Before this function is called, the \ref vorbis_info struct should be
* initialized by using \ref vorbis_info_init() from the libvorbis API. After
* encoding, vorbis_info_clear() should be called.
*
* \param vi Pointer to an initialized vorbis_info struct.
* \param channels The number of channels to be encoded.
* \param rate The sampling rate of the source audio.
* \param quality Desired quality level, currently from -0.1 to 1.0 (lo to hi).
*
* \return Zero for success, and negative values for failure.
*
* \retval 0 Success
* \retval OV_EFAULT Internal logic fault; indicates a bug or heap/stack corruption.
* \retval OV_EINVAL Invalid setup request, eg, out of range argument.
* \retval OV_EIMPL Unimplemented mode; unable to comply with quality level request.
*/
extern int vorbis_encode_setup_vbr(vorbis_info *vi,
long channels,
long rate,
float quality
);
/**
* This is the primary function within libvorbisenc for setting up variable
* bitrate ("quality" based) modes.
*
*
* Before this function is called, the vorbis_info struct should be
* initialized by using vorbis_info_init() from the libvorbis API. After
* encoding, vorbis_info_clear() should be called.
*
* \param vi Pointer to an initialized vorbis_info struct.
* \param channels The number of channels to be encoded.
* \param rate The sampling rate of the source audio.
* \param base_quality Desired quality level, currently from -0.1 to 1.0 (lo to hi).
*
*
* \return Zero for success, or a negative number for failure.
*
* \retval 0 Success
* \retval OV_EFAULT Internal logic fault; indicates a bug or heap/stack corruption.
* \retval OV_EINVAL Invalid setup request, eg, out of range argument.
* \retval OV_EIMPL Unimplemented mode; unable to comply with quality level request.
*/
extern int vorbis_encode_init_vbr(vorbis_info *vi,
long channels,
long rate,
float base_quality
);
/**
* This function performs the last stage of three-step encoding setup, as
* described in the API overview under managed bitrate modes.
*
* Before this function is called, the \ref vorbis_info struct should be
* initialized by using vorbis_info_init() from the libvorbis API, one of
* \ref vorbis_encode_setup_managed() or \ref vorbis_encode_setup_vbr() called to
* initialize the high-level encoding setup, and \ref vorbis_encode_ctl()
* called if necessary to make encoding setup changes.
* vorbis_encode_setup_init() finalizes the highlevel encoding structure into
* a complete encoding setup after which the application may make no further
* setup changes.
*
* After encoding, vorbis_info_clear() should be called.
*
* \param vi Pointer to an initialized \ref vorbis_info struct.
*
* \return Zero for success, and negative values for failure.
*
* \retval 0 Success.
* \retval OV_EFAULT Internal logic fault; indicates a bug or heap/stack corruption.
*
* \retval OV_EINVAL Attempt to use vorbis_encode_setup_init() without first
* calling one of vorbis_encode_setup_managed() or vorbis_encode_setup_vbr() to
* initialize the high-level encoding setup
*
*/
extern int vorbis_encode_setup_init(vorbis_info *vi);
/**
* This function implements a generic interface to miscellaneous encoder
* settings similar to the classic UNIX 'ioctl()' system call. Applications
* may use vorbis_encode_ctl() to query or set bitrate management or quality
* mode details by using one of several \e request arguments detailed below.
* vorbis_encode_ctl() must be called after one of
* vorbis_encode_setup_managed() or vorbis_encode_setup_vbr(). When used
* to modify settings, \ref vorbis_encode_ctl() must be called before \ref
* vorbis_encode_setup_init().
*
* \param vi Pointer to an initialized vorbis_info struct.
*
* \param number Specifies the desired action; See \ref encctlcodes "the list
* of available requests".
*
* \param arg void * pointing to a data structure matching the request
* argument.
*
* \retval 0 Success. Any further return information (such as the result of a
* query) is placed into the storage pointed to by *arg.
*
* \retval OV_EINVAL Invalid argument, or an attempt to modify a setting after
* calling vorbis_encode_setup_init().
*
* \retval OV_EIMPL Unimplemented or unknown request
*/
extern int vorbis_encode_ctl(vorbis_info *vi,int number,void *arg);
/**
* \deprecated This is a deprecated interface. Please use vorbis_encode_ctl()
* with the \ref ovectl_ratemanage2_arg struct and \ref
* OV_ECTL_RATEMANAGE2_GET and \ref OV_ECTL_RATEMANAGE2_SET calls in new code.
*
* The \ref ovectl_ratemanage_arg structure is used with vorbis_encode_ctl()
* and the \ref OV_ECTL_RATEMANAGE_GET, \ref OV_ECTL_RATEMANAGE_SET, \ref
* OV_ECTL_RATEMANAGE_AVG, \ref OV_ECTL_RATEMANAGE_HARD calls in order to
* query and modify specifics of the encoder's bitrate management
* configuration.
*/
struct ovectl_ratemanage_arg {
int management_active; /**< nonzero if bitrate management is active*/
/** hard lower limit (in kilobits per second) below which the stream bitrate
will never be allowed for any given bitrate_hard_window seconds of time.*/
long bitrate_hard_min;
/** hard upper limit (in kilobits per second) above which the stream bitrate
will never be allowed for any given bitrate_hard_window seconds of time.*/
long bitrate_hard_max;
/** the window period (in seconds) used to regulate the hard bitrate minimum
and maximum*/
double bitrate_hard_window;
/** soft lower limit (in kilobits per second) below which the average bitrate
tracker will start nudging the bitrate higher.*/
long bitrate_av_lo;
/** soft upper limit (in kilobits per second) above which the average bitrate
tracker will start nudging the bitrate lower.*/
long bitrate_av_hi;
/** the window period (in seconds) used to regulate the average bitrate
minimum and maximum.*/
double bitrate_av_window;
/** Regulates the relative centering of the average and hard windows; in
libvorbis 1.0 and 1.0.1, the hard window regulation overlapped but
followed the average window regulation. In libvorbis 1.1 a bit-reservoir
interface replaces the old windowing interface; the older windowing
interface is simulated and this field has no effect.*/
double bitrate_av_window_center;
};
/**
* \name struct ovectl_ratemanage2_arg
*
* The ovectl_ratemanage2_arg structure is used with vorbis_encode_ctl() and
* the OV_ECTL_RATEMANAGE2_GET and OV_ECTL_RATEMANAGE2_SET calls in order to
* query and modify specifics of the encoder's bitrate management
* configuration.
*
*/
struct ovectl_ratemanage2_arg {
int management_active; /**< nonzero if bitrate management is active */
/** Lower allowed bitrate limit in kilobits per second */
long bitrate_limit_min_kbps;
/** Upper allowed bitrate limit in kilobits per second */
long bitrate_limit_max_kbps;
long bitrate_limit_reservoir_bits; /**<Size of the bitrate reservoir in bits */
/** Regulates the bitrate reservoir's preferred fill level in a range from 0.0
* to 1.0; 0.0 tries to bank bits to buffer against future bitrate spikes, 1.0
* buffers against future sudden drops in instantaneous bitrate. Default is
* 0.1
*/
double bitrate_limit_reservoir_bias;
/** Average bitrate setting in kilobits per second */
long bitrate_average_kbps;
/** Slew rate limit setting for average bitrate adjustment; sets the minimum
* time in seconds the bitrate tracker may swing from one extreme to the
* other when boosting or damping average bitrate.
*/
double bitrate_average_damping;
};
/**
* \name vorbis_encode_ctl() codes
*
* \anchor encctlcodes
*
* These values are passed as the \c number parameter of vorbis_encode_ctl().
* The type of the referent of that function's \c arg pointer depends on these
* codes.
*/
/*@{*/
/**
* Query the current encoder bitrate management setting.
*
*Argument: <tt>struct ovectl_ratemanage2_arg *</tt>
*
* Used to query the current encoder bitrate management setting. Also used to
* initialize fields of an ovectl_ratemanage2_arg structure for use with
* \ref OV_ECTL_RATEMANAGE2_SET.
*/
#define OV_ECTL_RATEMANAGE2_GET 0x14
/**
* Set the current encoder bitrate management settings.
*
* Argument: <tt>struct ovectl_ratemanage2_arg *</tt>
*
* Used to set the current encoder bitrate management settings to the values
* listed in the ovectl_ratemanage2_arg. Passing a NULL pointer will disable
* bitrate management.
*/
#define OV_ECTL_RATEMANAGE2_SET 0x15
/**
* Returns the current encoder hard-lowpass setting (kHz) in the double
* pointed to by arg.
*
* Argument: <tt>double *</tt>
*/
#define OV_ECTL_LOWPASS_GET 0x20
/**
* Sets the encoder hard-lowpass to the value (kHz) pointed to by arg. Valid
* lowpass settings range from 2 to 99.
*
* Argument: <tt>double *</tt>
*/
#define OV_ECTL_LOWPASS_SET 0x21
/**
* Returns the current encoder impulse block setting in the double pointed
* to by arg.
*
* Argument: <tt>double *</tt>
*/
#define OV_ECTL_IBLOCK_GET 0x30
/**
* Sets the impulse block bias to the the value pointed to by arg.
*
* Argument: <tt>double *</tt>
*
* Valid range is -15.0 to 0.0 [default]. A negative impulse block bias will
* direct to encoder to use more bits when incoding short blocks that contain
* strong impulses, thus improving the accuracy of impulse encoding.
*/
#define OV_ECTL_IBLOCK_SET 0x31
/**
* Returns the current encoder coupling setting in the int pointed
* to by arg.
*
* Argument: <tt>int *</tt>
*/
#define OV_ECTL_COUPLING_GET 0x40
/**
* Enables/disables channel coupling in multichannel encoding according to arg.
*
* Argument: <tt>int *</tt>
*
* Zero disables channel coupling for multichannel inputs, nonzer enables
* channel coupling. Setting has no effect on monophonic encoding or
* multichannel counts that do not offer coupling. At present, coupling is
* available for stereo and 5.1 encoding.
*/
#define OV_ECTL_COUPLING_SET 0x41
/* deprecated rate management supported only for compatibility */
/**
* Old interface to querying bitrate management settings.
*
* Deprecated after move to bit-reservoir style management in 1.1 rendered
* this interface partially obsolete.
* \deprecated Please use \ref OV_ECTL_RATEMANAGE2_GET instead.
*
* Argument: <tt>struct ovectl_ratemanage_arg *</tt>
*/
#define OV_ECTL_RATEMANAGE_GET 0x10
/**
* Old interface to modifying bitrate management settings.
*
* deprecated after move to bit-reservoir style management in 1.1 rendered
* this interface partially obsolete.
*
* \deprecated Please use \ref OV_ECTL_RATEMANAGE2_SET instead.
*
* Argument: <tt>struct ovectl_ratemanage_arg *</tt>
*/
#define OV_ECTL_RATEMANAGE_SET 0x11
/**
* Old interface to setting average-bitrate encoding mode.
*
* Deprecated after move to bit-reservoir style management in 1.1 rendered
* this interface partially obsolete.
*
* \deprecated Please use \ref OV_ECTL_RATEMANAGE2_SET instead.
*
* Argument: <tt>struct ovectl_ratemanage_arg *</tt>
*/
#define OV_ECTL_RATEMANAGE_AVG 0x12
/**
* Old interface to setting bounded-bitrate encoding modes.
*
* deprecated after move to bit-reservoir style management in 1.1 rendered
* this interface partially obsolete.
*
* \deprecated Please use \ref OV_ECTL_RATEMANAGE2_SET instead.
*
* Argument: <tt>struct ovectl_ratemanage_arg *</tt>
*/
#define OV_ECTL_RATEMANAGE_HARD 0x13
/*@}*/
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif

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/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: stdio-based convenience library for opening/seeking/decoding
last mod: $Id: vorbisfile.h 17182 2010-04-29 03:48:32Z xiphmont $
********************************************************************/
#ifndef _OV_FILE_H_
#define _OV_FILE_H_
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
#include <stdio.h>
#include "codec.h"
/* The function prototypes for the callbacks are basically the same as for
* the stdio functions fread, fseek, fclose, ftell.
* The one difference is that the FILE * arguments have been replaced with
* a void * - this is to be used as a pointer to whatever internal data these
* functions might need. In the stdio case, it's just a FILE * cast to a void *
*
* If you use other functions, check the docs for these functions and return
* the right values. For seek_func(), you *MUST* return -1 if the stream is
* unseekable
*/
typedef struct {
size_t (*read_func) (void *ptr, size_t size, size_t nmemb, void *datasource);
int (*seek_func) (void *datasource, ogg_int64_t offset, int whence);
int (*close_func) (void *datasource);
long (*tell_func) (void *datasource);
} ov_callbacks;
#ifndef OV_EXCLUDE_STATIC_CALLBACKS
/* a few sets of convenient callbacks, especially for use under
* Windows where ov_open_callbacks() should always be used instead of
* ov_open() to avoid problems with incompatible crt.o version linking
* issues. */
static int _ov_header_fseek_wrap(FILE *f,ogg_int64_t off,int whence){
if(f==NULL)return(-1);
#ifdef __MINGW32__
return fseeko64(f,off,whence);
#elif defined (_WIN32)
return _fseeki64(f,off,whence);
#else
return fseek(f,off,whence);
#endif
}
/* These structs below (OV_CALLBACKS_DEFAULT etc) are defined here as
* static data. That means that every file which includes this header
* will get its own copy of these structs whether it uses them or
* not unless it #defines OV_EXCLUDE_STATIC_CALLBACKS.
* These static symbols are essential on platforms such as Windows on
* which several different versions of stdio support may be linked to
* by different DLLs, and we need to be certain we know which one
* we're using (the same one as the main application).
*/
static ov_callbacks OV_CALLBACKS_DEFAULT = {
(size_t (*)(void *, size_t, size_t, void *)) fread,
(int (*)(void *, ogg_int64_t, int)) _ov_header_fseek_wrap,
(int (*)(void *)) fclose,
(long (*)(void *)) ftell
};
static ov_callbacks OV_CALLBACKS_NOCLOSE = {
(size_t (*)(void *, size_t, size_t, void *)) fread,
(int (*)(void *, ogg_int64_t, int)) _ov_header_fseek_wrap,
(int (*)(void *)) NULL,
(long (*)(void *)) ftell
};
static ov_callbacks OV_CALLBACKS_STREAMONLY = {
(size_t (*)(void *, size_t, size_t, void *)) fread,
(int (*)(void *, ogg_int64_t, int)) NULL,
(int (*)(void *)) fclose,
(long (*)(void *)) NULL
};
static ov_callbacks OV_CALLBACKS_STREAMONLY_NOCLOSE = {
(size_t (*)(void *, size_t, size_t, void *)) fread,
(int (*)(void *, ogg_int64_t, int)) NULL,
(int (*)(void *)) NULL,
(long (*)(void *)) NULL
};
#endif
#define NOTOPEN 0
#define PARTOPEN 1
#define OPENED 2
#define STREAMSET 3
#define INITSET 4
typedef struct OggVorbis_File {
void *datasource; /* Pointer to a FILE *, etc. */
int seekable;
ogg_int64_t offset;
ogg_int64_t end;
ogg_sync_state oy;
/* If the FILE handle isn't seekable (eg, a pipe), only the current
stream appears */
int links;
ogg_int64_t *offsets;
ogg_int64_t *dataoffsets;
long *serialnos;
ogg_int64_t *pcmlengths; /* overloaded to maintain binary
compatibility; x2 size, stores both
beginning and end values */
vorbis_info *vi;
vorbis_comment *vc;
/* Decoding working state local storage */
ogg_int64_t pcm_offset;
int ready_state;
long current_serialno;
int current_link;
double bittrack;
double samptrack;
ogg_stream_state os; /* take physical pages, weld into a logical
stream of packets */
vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
vorbis_block vb; /* local working space for packet->PCM decode */
ov_callbacks callbacks;
} OggVorbis_File;
extern int ov_clear(OggVorbis_File *vf);
extern int ov_fopen(const char *path,OggVorbis_File *vf);
extern int ov_open(FILE *f,OggVorbis_File *vf,const char *initial,long ibytes);
extern int ov_open_callbacks(void *datasource, OggVorbis_File *vf,
const char *initial, long ibytes, ov_callbacks callbacks);
extern int ov_test(FILE *f,OggVorbis_File *vf,const char *initial,long ibytes);
extern int ov_test_callbacks(void *datasource, OggVorbis_File *vf,
const char *initial, long ibytes, ov_callbacks callbacks);
extern int ov_test_open(OggVorbis_File *vf);
extern long ov_bitrate(OggVorbis_File *vf,int i);
extern long ov_bitrate_instant(OggVorbis_File *vf);
extern long ov_streams(OggVorbis_File *vf);
extern long ov_seekable(OggVorbis_File *vf);
extern long ov_serialnumber(OggVorbis_File *vf,int i);
extern ogg_int64_t ov_raw_total(OggVorbis_File *vf,int i);
extern ogg_int64_t ov_pcm_total(OggVorbis_File *vf,int i);
extern double ov_time_total(OggVorbis_File *vf,int i);
extern int ov_raw_seek(OggVorbis_File *vf,ogg_int64_t pos);
extern int ov_pcm_seek(OggVorbis_File *vf,ogg_int64_t pos);
extern int ov_pcm_seek_page(OggVorbis_File *vf,ogg_int64_t pos);
extern int ov_time_seek(OggVorbis_File *vf,double pos);
extern int ov_time_seek_page(OggVorbis_File *vf,double pos);
extern int ov_raw_seek_lap(OggVorbis_File *vf,ogg_int64_t pos);
extern int ov_pcm_seek_lap(OggVorbis_File *vf,ogg_int64_t pos);
extern int ov_pcm_seek_page_lap(OggVorbis_File *vf,ogg_int64_t pos);
extern int ov_time_seek_lap(OggVorbis_File *vf,double pos);
extern int ov_time_seek_page_lap(OggVorbis_File *vf,double pos);
extern ogg_int64_t ov_raw_tell(OggVorbis_File *vf);
extern ogg_int64_t ov_pcm_tell(OggVorbis_File *vf);
extern double ov_time_tell(OggVorbis_File *vf);
extern vorbis_info *ov_info(OggVorbis_File *vf,int link);
extern vorbis_comment *ov_comment(OggVorbis_File *vf,int link);
extern long ov_read_float(OggVorbis_File *vf,float ***pcm_channels,int samples,
int *bitstream);
extern long ov_read_filter(OggVorbis_File *vf,char *buffer,int length,
int bigendianp,int word,int sgned,int *bitstream,
void (*filter)(float **pcm,long channels,long samples,void *filter_param),void *filter_param);
extern long ov_read(OggVorbis_File *vf,char *buffer,int length,
int bigendianp,int word,int sgned,int *bitstream);
extern int ov_crosslap(OggVorbis_File *vf1,OggVorbis_File *vf2);
extern int ov_halfrate(OggVorbis_File *vf,int flag);
extern int ov_halfrate_p(OggVorbis_File *vf);
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif