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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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8
cmake/Modules/FindSFML.cmake
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@ -322,11 +322,13 @@ if(SFML_STATIC_LIBRARIES)
# find libraries
find_sfml_dependency(OPENAL_LIBRARY "OpenAL" openal openal32)
find_sfml_dependency(SNDFILE_LIBRARY "libsndfile" sndfile)
find_sfml_dependency(OGG_LIBRARY "Ogg" ogg)
find_sfml_dependency(VORBIS_LIBRARY "Vorbis" vorbis)
find_sfml_dependency(VORBISFILE_LIBRARY "VorbisFile" vorbisfile)
# update the list
set(SFML_AUDIO_DEPENDENCIES ${OPENAL_LIBRARY} ${SNDFILE_LIBRARY})
set(SFML_DEPENDENCIES ${SFML_AUDIO_DEPENDENCIES} ${SFML_DEPENDENCIES})
set(SFML_AUDIO_DEPENDENCIES ${OPENAL_LIBRARY} ${VORBISFILE_LIBRARY} ${VORBIS_LIBRARY} ${OGG_LIBRARY})
set(SFML_DEPENDENCIES ${SFML_DEPENDENCIES} ${SFML_AUDIO_DEPENDENCIES})
endif()
endif()

21
cmake/Modules/FindVorbis.cmake Normal file
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@ -0,0 +1,21 @@
#
# Try to find Ogg/Vorbis libraries and include paths.
# Once done this will define
#
# VORBIS_FOUND
# VORBIS_INCLUDE_DIRS
# VORBIS_LIBRARIES
#
find_path(OGG_INCLUDE_DIR ogg/ogg.h)
find_path(VORBIS_INCLUDE_DIR vorbis/vorbisfile.h)
find_library(OGG_LIBRARY NAMES ogg)
find_library(VORBIS_LIBRARY NAMES vorbis)
find_library(VORBISFILE_LIBRARY NAMES vorbisfile)
include(FindPackageHandleStandardArgs)
find_package_handle_standard_args(VORBIS DEFAULT_MSG VORBIS_LIBRARY VORBISFILE_LIBRARY OGG_LIBRARY VORBIS_INCLUDE_DIR OGG_INCLUDE_DIR)
set(VORBIS_INCLUDE_DIRS ${OGG_INCLUDE_DIR} ${VORBIS_INCLUDE_DIR})
set(VORBIS_LIBRARIES ${VORBISFILE_LIBRARY} ${VORBIS_LIBRARY} ${OGG_LIBRARY})

6
extlibs/Android.mk vendored
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@ -12,12 +12,6 @@ LOCAL_SRC_FILES := lib/$(TARGET_ARCH_ABI)/libjpeg.a
LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
include $(PREBUILT_STATIC_LIBRARY)
include $(CLEAR_VARS)
LOCAL_MODULE := sndfile
LOCAL_SRC_FILES := lib/$(TARGET_ARCH_ABI)/libsndfile.so
LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
include $(PREBUILT_SHARED_LIBRARY)
include $(CLEAR_VARS)
LOCAL_MODULE := openal
LOCAL_SRC_FILES := lib/$(TARGET_ARCH_ABI)/libopenal.so

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extlibs/bin/x64/libsndfile-1.dll vendored
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extlibs/bin/x86/libsndfile-1.dll vendored
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666
extlibs/headers/libsndfile/android/sndfile.h vendored
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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_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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@ -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_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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/********************************************************************
* *
* 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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@ -0,0 +1,206 @@
/********************************************************************
* *
* 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

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<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
<plist version="1.0">
<dict>
<key>CFBundleDevelopmentRegion</key>
<string>English</string>
<key>CFBundleExecutable</key>
<string>sndfile</string>
<key>CFBundleIdentifier</key>
<string>com.mega-nerd.sndfile</string>
<key>CFBundleInfoDictionaryVersion</key>
<string>6.0</string>
<key>CFBundlePackageType</key>
<string>FMWK</string>
<key>CFBundleSignature</key>
<string>????</string>
<key>CFBundleVersion</key>
<string>1.0</string>
</dict>
</plist>

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#include <SFML/Audio/Sound.hpp>
#include <SFML/Audio/SoundBuffer.hpp>
#include <SFML/Audio/SoundBufferRecorder.hpp>
#include <SFML/Audio/InputSoundFile.hpp>
#include <SFML/Audio/OutputSoundFile.hpp>
#include <SFML/Audio/SoundFileFactory.hpp>
#include <SFML/Audio/SoundFileReader.hpp>
#include <SFML/Audio/SoundFileWriter.hpp>
#include <SFML/Audio/SoundRecorder.hpp>
#include <SFML/Audio/SoundStream.hpp>

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//
////////////////////////////////////////////////////////////
#ifndef SFML_SOUNDFILE_HPP
#define SFML_SOUNDFILE_HPP
#ifndef SFML_INPUTSOUNDFILE_HPP
#define SFML_INPUTSOUNDFILE_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/Export.hpp>
#include <SFML/System/NonCopyable.hpp>
#include <SFML/System/Time.hpp>
#include <sndfile.h>
#include <string>
namespace sf
{
class InputStream;
class SoundFileReader;
namespace priv
{
////////////////////////////////////////////////////////////
/// \brief Provide read and write access to sound files
/// \brief Provide read access to sound files
///
////////////////////////////////////////////////////////////
class SoundFile : NonCopyable
class SFML_AUDIO_API InputSoundFile : NonCopyable
{
public :
@ -52,13 +51,62 @@ public:
/// \brief Default constructor
///
////////////////////////////////////////////////////////////
SoundFile();
InputSoundFile();
////////////////////////////////////////////////////////////
/// \brief Destructor
///
////////////////////////////////////////////////////////////
~SoundFile();
~InputSoundFile();
////////////////////////////////////////////////////////////
/// \brief Open a sound file from the disk for reading
///
/// The supported audio formats are: WAV, OGG/Vorbis, FLAC.
///
/// \param filename Path of the sound file to load
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
bool openFromFile(const std::string& filename);
////////////////////////////////////////////////////////////
/// \brief Open a sound file in memory for reading
///
/// The supported audio formats are: WAV, OGG/Vorbis, FLAC.
///
/// \param data Pointer to the file data in memory
/// \param sizeInBytes Size of the data to load, in bytes
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
bool openFromMemory(const void* data, std::size_t sizeInBytes);
////////////////////////////////////////////////////////////
/// \brief Open a sound file from a custom stream for reading
///
/// The supported audio formats are: WAV, OGG/Vorbis, FLAC.
///
/// \param stream Source stream to read from
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
bool openFromStream(InputStream& stream);
////////////////////////////////////////////////////////////
/// \brief Open the sound file from the disk for writing
///
/// \param filename Path of the sound file to write
/// \param channelCount Number of channels in the sound
/// \param sampleRate Sample rate of the sound
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
bool openForWriting(const std::string& filename, unsigned int channelCount, unsigned int sampleRate);
////////////////////////////////////////////////////////////
/// \brief Get the total number of audio samples in the file
@ -66,7 +114,7 @@ public:
/// \return Number of samples
///
////////////////////////////////////////////////////////////
std::size_t getSampleCount() const;
Uint64 getSampleCount() const;
////////////////////////////////////////////////////////////
/// \brief Get the number of channels used by the sound
@ -85,147 +133,118 @@ public:
unsigned int getSampleRate() const;
////////////////////////////////////////////////////////////
/// \brief Open a sound file for reading
/// \brief Get the total duration of the sound file
///
/// \param filename Path of the sound file to load
/// This function is provided for convenience, the duration is
/// deduced from the other sound file attributes.
///
/// \return True if the file was successfully opened
/// \return Duration of the sound file
///
////////////////////////////////////////////////////////////
bool openRead(const std::string& filename);
Time getDuration() const;
////////////////////////////////////////////////////////////
/// \brief Open a sound file in memory for reading
/// \brief Change the current read position to the given sample offset
///
/// \param data Pointer to the file data in memory
/// \param sizeInBytes Size of the data to load, in bytes
/// This function takes a sample offset to provide maximum
/// precision. If you need to jump to a given time, use the
/// other overload.
///
/// \return True if the file was successfully opened
/// If the given offset exceeds to total number of samples,
/// this function jumps to the end of the sound file.
///
/// \param sampleOffset Index of the sample to jump to, relative to the beginning
///
////////////////////////////////////////////////////////////
bool openRead(const void* data, std::size_t sizeInBytes);
void seek(Uint64 sampleOffset);
////////////////////////////////////////////////////////////
/// \brief Open a sound file from a custom stream for reading
/// \brief Change the current read position to the given time offset
///
/// \param stream Source stream to read from
/// Using a time offset is handy but unprecise. If you need an accurate
/// result, consider using the overload which takes a sample offset.
///
/// \return True if the file was successfully opened
/// If the given time exceeds to total duration, this function jumps
/// to the end of the sound file.
///
////////////////////////////////////////////////////////////
bool openRead(InputStream& stream);
////////////////////////////////////////////////////////////
/// \brief a the sound file for writing
///
/// \param filename Path of the sound file to write
/// \param channelCount Number of channels in the sound
/// \param sampleRate Sample rate of the sound
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
bool openWrite(const std::string& filename, unsigned int channelCount, unsigned int sampleRate);
////////////////////////////////////////////////////////////
/// \brief Read audio samples from the loaded sound
///
/// \param data Pointer to the sample array to fill
/// \param sampleCount Number of samples to read
///
/// \return Number of samples actually read (may be less than \a sampleCount)
///
////////////////////////////////////////////////////////////
std::size_t read(Int16* data, std::size_t sampleCount);
////////////////////////////////////////////////////////////
/// \brief Write audio samples to the file
///
/// \param data Pointer to the sample array to write
/// \param sampleCount Number of samples to write
///
////////////////////////////////////////////////////////////
void write(const Int16* data, std::size_t sampleCount);
////////////////////////////////////////////////////////////
/// \brief Change the current read position in the file
///
/// \param timeOffset New playing position, from the beginning of the file
/// \param timeOffset Time to jump to, relative to the beginning
///
////////////////////////////////////////////////////////////
void seek(Time timeOffset);
////////////////////////////////////////////////////////////
/// \brief Read audio samples from the open file
///
/// \param samples Pointer to the sample array to fill
/// \param maxCount Maximum number of samples to read
///
/// \return Number of samples actually read (may be less than \a maxCount)
///
////////////////////////////////////////////////////////////
Uint64 read(Int16* samples, Uint64 maxCount);
private:
////////////////////////////////////////////////////////////
/// \brief Initialize the internal state of the sound file
///
/// This function is called by all the openRead functions.
///
/// \param fileInfo Information about the loaded sound file
/// \brief Close the current file
///
////////////////////////////////////////////////////////////
void initialize(SF_INFO fileInfo);
////////////////////////////////////////////////////////////
/// \brief Get the internal format of an audio file according to
/// its filename extension
///
/// \param filename Filename to check
///
/// \return Internal format matching the filename (-1 if no match)
///
////////////////////////////////////////////////////////////
static int getFormatFromFilename(const std::string& filename);
////////////////////////////////////////////////////////////
/// \brief Data and callbacks for opening from memory
///
////////////////////////////////////////////////////////////
struct Memory
{
const char* begin;
const char* current;
sf_count_t size;
static sf_count_t getLength(void* user);
static sf_count_t read(void* ptr, sf_count_t count, void* user);
static sf_count_t seek(sf_count_t offset, int whence, void* user);
static sf_count_t tell(void* user);
};
////////////////////////////////////////////////////////////
/// \brief Data and callbacks for opening from stream
///
////////////////////////////////////////////////////////////
struct Stream
{
InputStream* source;
Int64 size;
static sf_count_t getLength(void* user);
static sf_count_t read(void* ptr, sf_count_t count, void* user);
static sf_count_t seek(sf_count_t offset, int whence, void* user);
static sf_count_t tell(void* user);
};
void close();
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
SNDFILE* m_file; ///< File descriptor
Memory m_memory; ///< Memory reading info
Stream m_stream; ///< Stream reading info
std::size_t m_sampleCount; ///< Total number of samples in the file
unsigned int m_channelCount; ///< Number of channels used by the sound
SoundFileReader* m_reader; ///< Reader that handles I/O on the file's format
InputStream* m_stream; ///< Input stream used to access the file's data
bool m_streamOwned; ///< Is the stream internal or external?
Uint64 m_sampleCount; ///< Total number of samples in the file
unsigned int m_channelCount; ///< Number of channels of the sound
unsigned int m_sampleRate; ///< Number of samples per second
#ifdef SFML_SYSTEM_ANDROID
void* m_resourceStream; ///< Asset file streamer (if loaded from file)
#endif
};
} // namespace priv
} // namespace sf
#endif // SFML_SOUNDFILE_HPP
#endif // SFML_INPUTSOUNDFILE_HPP
////////////////////////////////////////////////////////////
/// \class sf::InputSoundFile
/// \ingroup audio
///
/// This class decodes audio samples from a sound file. It is
/// used internally by higher-level classes such as sf::SoundBuffer
/// and sf::Music, but can also be useful if you want to process
/// or analyze audio files without playing them, or if you want to
/// implement your own version of sf::Music with more specific
/// features.
///
/// Usage example:
/// \code
/// // Open a sound file
/// sf::InputSoundFile file;
/// if (!file.openFromFile("music.ogg"))
/// /* error */;
///
/// // Print the sound attributes
/// std::cout << "duration: " << file.getDuration().asSeconds() << std::endl;
/// std::cout << "channels: " << file.getChannelCount() << std::endl;
/// std::cout << "sample rate: " << file.getSampleRate() << std::endl;
/// std::cout << "sample count: " << file.getSampleCount() << std::endl;
///
/// // Read and process batches of samples until the end of file is reached
/// sf::Int16 samples[1024];
/// sf::Uint64 count;
/// do
/// {
/// count = file.read(samples, 1024);
///
/// // process, analyse, play, convert, or whatever
/// // you want to do with the samples...
/// }
/// while (count > 0);
/// \endcode
///
/// \see sf::SoundFileReader, sf::OutputSoundFile
///
////////////////////////////////////////////////////////////

35
include/SFML/Audio/Music.hpp
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@ -30,6 +30,7 @@
////////////////////////////////////////////////////////////
#include <SFML/Audio/Export.hpp>
#include <SFML/Audio/SoundStream.hpp>
#include <SFML/Audio/InputSoundFile.hpp>
#include <SFML/System/Mutex.hpp>
#include <SFML/System/Time.hpp>
#include <string>
@ -38,11 +39,6 @@
namespace sf
{
namespace priv
{
class SoundFile;
}
class InputStream;
////////////////////////////////////////////////////////////
@ -70,9 +66,8 @@ public:
///
/// This function doesn't start playing the music (call play()
/// to do so).
/// Here is a complete list of all the supported audio formats:
/// ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc, ircam,
/// w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
/// See the documentation of sf::InputSoundFile for the list
/// of supported formats.
///
/// \param filename Path of the music file to open
///
@ -88,12 +83,11 @@ public:
///
/// This function doesn't start playing the music (call play()
/// to do so).
/// Here is a complete list of all the supported audio formats:
/// ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc, ircam,
/// w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
/// See the documentation of sf::InputSoundFile for the list
/// of supported formats.
/// Since the music is not loaded completely but rather streamed
/// continuously, the \a data must remain available as long as the
/// music is playing (i.e. you can't deallocate it right after calling
/// music is playing (ie. you can't deallocate it right after calling
/// this function).
///
/// \param data Pointer to the file data in memory
@ -111,12 +105,11 @@ public:
///
/// This function doesn't start playing the music (call play()
/// to do so).
/// Here is a complete list of all the supported audio formats:
/// ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc, ircam,
/// w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
/// See the documentation of sf::InputSoundFile for the list
/// of supported formats.
/// Since the music is not loaded completely but rather streamed
/// continuously, the \a stream must remain alive as long as the
/// music is playing (i.e. you can't destroy it right after calling
/// music is playing (ie. you can't destroy it right after calling
/// this function).
///
/// \param stream Source stream to read from
@ -170,7 +163,7 @@ private:
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
priv::SoundFile* m_file; ///< Sound file
InputSoundFile m_file; ///< The streamed music file
Time m_duration; ///< Music duration
std::vector<Int16> m_samples; ///< Temporary buffer of samples
Mutex m_mutex; ///< Mutex protecting the data
@ -186,13 +179,13 @@ private:
/// \class sf::Music
/// \ingroup audio
///
/// Music objects are sounds that are streamed rather than
/// completely loaded in memory. This is especially useful for
/// compressed music that usually takes hundreds of MB when it is
/// Musics are sounds that are streamed rather than completely
/// loaded in memory. This is especially useful for compressed
/// musics that usually take hundreds of MB when they are
/// uncompressed: by streaming it instead of loading it entirely,
/// you avoid saturating the memory and have almost no loading delay.
///
/// Apart from that, sf::Music has almost the same features as
/// Apart from that, a sf::Music has almost the same features as
/// the sf::SoundBuffer / sf::Sound pair: you can play/pause/stop
/// it, request its parameters (channels, sample rate), change
/// the way it is played (pitch, volume, 3D position, ...), etc.

133
include/SFML/Audio/OutputSoundFile.hpp Normal file
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@ -0,0 +1,133 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
#ifndef SFML_OUTPUTSOUNDFILE_HPP
#define SFML_OUTPUTSOUNDFILE_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/Export.hpp>
#include <SFML/System/NonCopyable.hpp>
#include <string>
namespace sf
{
class SoundFileWriter;
////////////////////////////////////////////////////////////
/// \brief Provide write access to sound files
///
////////////////////////////////////////////////////////////
class SFML_AUDIO_API OutputSoundFile : NonCopyable
{
public :
////////////////////////////////////////////////////////////
/// \brief Default constructor
///
////////////////////////////////////////////////////////////
OutputSoundFile();
////////////////////////////////////////////////////////////
/// \brief Destructor
///
/// Closes the file if it was still open.
///
////////////////////////////////////////////////////////////
~OutputSoundFile();
////////////////////////////////////////////////////////////
/// \brief Open the sound file from the disk for writing
///
/// The supported audio formats are: WAV, OGG/Vorbis, FLAC.
///
/// \param filename Path of the sound file to write
/// \param sampleRate Sample rate of the sound
/// \param channelCount Number of channels in the sound
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
bool openFromFile(const std::string& filename, unsigned int sampleRate, unsigned int channelCount);
////////////////////////////////////////////////////////////
/// \brief Write audio samples to the file
///
/// \param samples Pointer to the sample array to write
/// \param sampleCount Number of samples to write
///
////////////////////////////////////////////////////////////
void write(const Int16* samples, Uint64 count);
private:
////////////////////////////////////////////////////////////
/// \brief Close the current file
///
////////////////////////////////////////////////////////////
void close();
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
SoundFileWriter* m_writer; ///< Writer that handles I/O on the file's format
};
} // namespace sf
#endif // SFML_OUTPUTSOUNDFILE_HPP
////////////////////////////////////////////////////////////
/// \class sf::OutputSoundFile
/// \ingroup audio
///
/// This class encodes audio samples to a sound file. It is
/// used internally by higher-level classes such as sf::SoundBuffer,
/// but can also be useful if you want to create audio files from
/// custom data sources, like generated audio samples.
///
/// Usage example:
/// \code
/// // Create a sound file, ogg/vorbis format, 44100 Hz, stereo
/// sf::OutputSoundFile file;
/// if (!file.openFromFile("music.ogg", 44100, 2))
/// /* error */;
///
/// while (...)
/// {
/// // Read or generate audio samples from your custom source
/// std::vector<sf::Int16> samples = ...;
///
/// // Write them to the file
/// file.write(samples.data(), samples.size());
/// }
/// \endcode
///
/// \see sf::SoundFileWriter, sf::InputSoundFile
///
////////////////////////////////////////////////////////////

32
include/SFML/Audio/SoundBuffer.hpp
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@ -38,12 +38,8 @@
namespace sf
{
namespace priv
{
class SoundFile;
}
class Sound;
class InputSoundFile;
class InputStream;
////////////////////////////////////////////////////////////
@ -77,9 +73,8 @@ public:
////////////////////////////////////////////////////////////
/// \brief Load the sound buffer from a file
///
/// Here is a complete list of all the supported audio formats:
/// ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc, ircam,
/// w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
/// See the documentation of sf::InputSoundFile for the list
/// of supported formats.
///
/// \param filename Path of the sound file to load
///
@ -93,9 +88,8 @@ public:
////////////////////////////////////////////////////////////
/// \brief Load the sound buffer from a file in memory
///
/// Here is a complete list of all the supported audio formats:
/// ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc, ircam,
/// w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
/// See the documentation of sf::InputSoundFile for the list
/// of supported formats.
///
/// \param data Pointer to the file data in memory
/// \param sizeInBytes Size of the data to load, in bytes
@ -110,9 +104,8 @@ public:
////////////////////////////////////////////////////////////
/// \brief Load the sound buffer from a custom stream
///
/// Here is a complete list of all the supported audio formats:
/// ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc, ircam,
/// w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
/// See the documentation of sf::InputSoundFile for the list
/// of supported formats.
///
/// \param stream Source stream to read from
///
@ -139,14 +132,13 @@ public:
/// \see loadFromFile, loadFromMemory, saveToFile
///
////////////////////////////////////////////////////////////
bool loadFromSamples(const Int16* samples, std::size_t sampleCount, unsigned int channelCount, unsigned int sampleRate);
bool loadFromSamples(const Int16* samples, Uint64 sampleCount, unsigned int channelCount, unsigned int sampleRate);
////////////////////////////////////////////////////////////
/// \brief Save the sound buffer to an audio file
///
/// Here is a complete list of all the supported audio formats:
/// ogg, wav, flac, aiff, au, raw, paf, svx, nist, voc, ircam,
/// w64, mat4, mat5 pvf, htk, sds, avr, sd2, caf, wve, mpc2k, rf64.
/// See the documentation of sf::OutputSoundFile for the list
/// of supported formats.
///
/// \param filename Path of the sound file to write
///
@ -182,7 +174,7 @@ public:
/// \see getSamples
///
////////////////////////////////////////////////////////////
std::size_t getSampleCount() const;
Uint64 getSampleCount() const;
////////////////////////////////////////////////////////////
/// \brief Get the sample rate of the sound
@ -243,7 +235,7 @@ private:
/// \return True on succesful initialization, false on failure
///
////////////////////////////////////////////////////////////
bool initialize(priv::SoundFile& file);
bool initialize(InputSoundFile& file);
////////////////////////////////////////////////////////////
/// \brief Update the internal buffer with the cached audio samples

189
include/SFML/Audio/SoundFileFactory.hpp Normal file
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@ -0,0 +1,189 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
#ifndef SFML_SOUNDFILEFACTORY_HPP
#define SFML_SOUNDFILEFACTORY_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/Export.hpp>
#include <string>
#include <vector>
namespace sf
{
class InputStream;
class SoundFileReader;
class SoundFileWriter;
////////////////////////////////////////////////////////////
/// \brief Manages and instanciates sound file readers and writers
///
////////////////////////////////////////////////////////////
class SFML_AUDIO_API SoundFileFactory
{
public :
////////////////////////////////////////////////////////////
/// \brief Register a new reader
///
/// \see unregisterReader
///
////////////////////////////////////////////////////////////
template <typename T>
static void registerReader();
////////////////////////////////////////////////////////////
/// \brief Unregister a reader
///
/// \see registerReader
///
////////////////////////////////////////////////////////////
template <typename T>
static void unregisterReader();
////////////////////////////////////////////////////////////
/// \brief Register a new writer
///
/// \see unregisterWriter
///
////////////////////////////////////////////////////////////
template <typename T>
static void registerWriter();
////////////////////////////////////////////////////////////
/// \brief Unregister a writer
///
/// \see registerWriter
///
////////////////////////////////////////////////////////////
template <typename T>
static void unregisterWriter();
////////////////////////////////////////////////////////////
/// \brief Instanciate the right reader for the given file on disk
///
/// \param filename Path of the sound file
///
/// \return A new sound file reader that can read the given file, or null if no reader can handle it
///
/// \see createReaderFromMemory, createReaderFromStream
///
////////////////////////////////////////////////////////////
static SoundFileReader* createReaderFromFilename(const std::string& filename);
////////////////////////////////////////////////////////////
/// \brief Instanciate the right codec for the given file in memory
///
/// \param data Pointer to the file data in memory
/// \param sizeInBytes Total size of the file data, in bytes
///
/// \return A new sound file codec that can read the given file, or null if no codec can handle it
///
/// \see createReaderFromFilename, createReaderFromStream
///
////////////////////////////////////////////////////////////
static SoundFileReader* createReaderFromMemory(const void* data, std::size_t sizeInBytes);
////////////////////////////////////////////////////////////
/// \brief Instanciate the right codec for the given file in stream
///
/// \param stream Source stream to read from
///
/// \return A new sound file codec that can read the given file, or null if no codec can handle it
///
/// \see createReaderFromFilename, createReaderFromMemory
///
////////////////////////////////////////////////////////////
static SoundFileReader* createReaderFromStream(InputStream& stream);
////////////////////////////////////////////////////////////
/// \brief Instanciate the right writer for the given file on disk
///
/// \param filename Path of the sound file
///
/// \return A new sound file writer that can write given file, or null if no writer can handle it
///
////////////////////////////////////////////////////////////
static SoundFileWriter* createWriterFromFilename(const std::string& filename);
private:
////////////////////////////////////////////////////////////
// Types
////////////////////////////////////////////////////////////
struct ReaderFactory
{
bool (*check)(InputStream&);
SoundFileReader* (*create)();
};
typedef std::vector<ReaderFactory> ReaderFactoryArray;
struct WriterFactory
{
bool (*check)(const std::string&);
SoundFileWriter* (*create)();
};
typedef std::vector<WriterFactory> WriterFactoryArray;
////////////////////////////////////////////////////////////
// Static member data
////////////////////////////////////////////////////////////
static ReaderFactoryArray s_readers; ///< List of all registered readers
static WriterFactoryArray s_writers; ///< List of all registered writers
};
} // namespace sf
#include <SFML/Audio/SoundFileFactory.inl>
#endif // SFML_SOUNDFILEFACTORY_HPP
////////////////////////////////////////////////////////////
/// \class sf::SoundFileFactory
/// \ingroup audio
///
/// This class is where all the sound file readers and writers are
/// registered. You should normally only need to use its regitration
/// and unregistration functions; readers/writers creation and manipulation
/// are wrapped into the higher-level classes sf::InputSoundFile and
/// sf::OutputSoundFile.
///
/// To register a new reader (writer) use the sf::SoundFileFactory::registerReader
/// (registerWriter) static function. You don't have to call the unregisterReader
/// (unregisterWriter) function, unless you want to unregister a format before your
/// application ends (typically, when a plugin is unloaded).
///
/// Usage example:
/// \code
/// sf::SoundFileFactory::registerReader<MySoundFileReader>();
/// sf::SoundFileFactory::registerWriter<MySoundFileWriter>();
/// \endcode
///
/// \see sf::InputSoundFile, sf::OutputSoundFile, sf::SoundFileReader, sf::SoundFileWriter
///
////////////////////////////////////////////////////////////

100
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@ -0,0 +1,100 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
namespace sf
{
namespace priv
{
template <typename T> SoundFileReader* createReader() {return new T;}
template <typename T> SoundFileWriter* createWriter() {return new T;}
}
////////////////////////////////////////////////////////////
template <typename T>
void SoundFileFactory::registerReader()
{
// Make sure the same class won't be registered twice
unregisterReader<T>();
// Create a new factory with the functions provided by the class
ReaderFactory factory;
factory.check = &T::check;
factory.create = &priv::createReader<T>;
// Add it
s_readers.push_back(factory);
}
////////////////////////////////////////////////////////////
template <typename T>
void SoundFileFactory::unregisterReader()
{
// Remove the instance(s) of the reader from the array of factories
for (ReaderFactoryArray::iterator it = s_readers.begin(); it != s_readers.end(); )
{
if (it->create == &priv::createReader<T>)
it = s_readers.erase(it);
else
++it;
}
}
////////////////////////////////////////////////////////////
template <typename T>
void SoundFileFactory::registerWriter()
{
// Make sure the same class won't be registered twice
unregisterWriter<T>();
// Create a new factory with the functions provided by the class
WriterFactory factory;
factory.check = &T::check;
factory.create = &priv::createWriter<T>;
// Add it
s_writers.push_back(factory);
}
////////////////////////////////////////////////////////////
template <typename T>
void SoundFileFactory::unregisterWriter()
{
// Remove the instance(s) of the writer from the array of factories
for (WriterFactoryArray::iterator it = s_writers.begin(); it != s_writers.end(); )
{
if (it->create == &priv::createWriter<T>)
it = s_writers.erase(it);
else
++it;
}
}
} // namespace sf

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@ -0,0 +1,161 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
#ifndef SFML_SOUNDFILEREADER_HPP
#define SFML_SOUNDFILEREADER_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/Export.hpp>
#include <string>
namespace sf
{
class InputStream;
////////////////////////////////////////////////////////////
/// \brief Abstract base class for sound file decoding
///
////////////////////////////////////////////////////////////
class SFML_AUDIO_API SoundFileReader
{
public :
////////////////////////////////////////////////////////////
/// \brief Structure holding the audio properties of a sound file
///
////////////////////////////////////////////////////////////
struct Info
{
Uint64 sampleCount; ///< Total number of samples in the file
unsigned int channelCount; ///< Number of channels of the sound
unsigned int sampleRate; ///< Samples rate of the sound, in samples per second
};
////////////////////////////////////////////////////////////
/// \brief Virtual destructor
///
////////////////////////////////////////////////////////////
virtual ~SoundFileReader() {}
////////////////////////////////////////////////////////////
/// \brief Open a sound file for reading
///
/// The provided stream reference is valid as long as the
/// SoundFileReader is alive, so it is safe to use/store it
/// during the whole lifetime of the reader.
///
/// \param stream Source stream to read from
/// \param info Structure to fill with the properties of the loaded sound
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
virtual bool open(InputStream& stream, Info& info) = 0;
////////////////////////////////////////////////////////////
/// \brief Change the current read position to the given sample offset
///
/// If the given offset exceeds to total number of samples,
/// this function must jump to the end of the file.
///
/// \param sampleOffset Index of the sample to jump to, relative to the beginning
///
////////////////////////////////////////////////////////////
virtual void seek(Uint64 sampleOffset) = 0;
////////////////////////////////////////////////////////////
/// \brief Read audio samples from the open file
///
/// \param samples Pointer to the sample array to fill
/// \param maxCount Maximum number of samples to read
///
/// \return Number of samples actually read (may be less than \a maxCount)
///
////////////////////////////////////////////////////////////
virtual Uint64 read(Int16* samples, Uint64 maxCount) = 0;
};
} // namespace sf
#endif // SFML_SOUNDFILEREADER_HPP
////////////////////////////////////////////////////////////
/// \class sf::SoundFileReader
/// \ingroup audio
///
/// This class allows users to read audio file formats not natively
/// supported by SFML, and thus extend the set of supported readable
/// audio formats.
///
/// A valid sound file reader must override the open, seek and write functions,
/// as well as providing a static check function; the latter is used by
/// SFML to find a suitable writer for a given input file.
///
/// To register a new reader, use the sf::SoundFileFactory::registerReader
/// template function.
///
/// Usage example:
/// \code
/// class MySoundFileReader : public sf::SoundFileReader
/// {
/// public:
///
/// static bool check(sf::InputStream& stream)
/// {
/// // typically, read the first few header bytes and check fields that identify the format
/// // return true if the reader can handle the format
/// }
///
/// virtual bool open(sf::InputStream& stream, Info& info)
/// {
/// // read the sound file header and fill the sound attributes
/// // (channel count, sample count and sample rate)
/// // return true on success
/// }
///
/// virtual void seek(sf::Uint64 sampleOffset)
/// {
/// // advance to the sampleOffset-th sample from the beginning of the sound
/// }
///
/// virtual sf::Uint64 read(sf::Int16* samples, sf::Uint64 maxCount)
/// {
/// // read up to 'maxCount' samples into the 'samples' array,
/// // convert them (for example from normalized float) if they are not stored
/// // as 16-bits signed integers in the file
/// // return the actual number of samples read
/// }
/// };
///
/// sf::SoundFileFactory::registerReader<MySoundFileReader>();
/// \endcode
///
/// \see sf::InputSoundFile, sf::SoundFileFactory, sf::SoundFileWriter
///
////////////////////////////////////////////////////////////

125
include/SFML/Audio/SoundFileWriter.hpp Normal file
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@ -0,0 +1,125 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
#ifndef SFML_SOUNDFILEWRITER_HPP
#define SFML_SOUNDFILEWRITER_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/Export.hpp>
#include <string>
namespace sf
{
////////////////////////////////////////////////////////////
/// \brief Abstract base class for sound file encoding
///
////////////////////////////////////////////////////////////
class SFML_AUDIO_API SoundFileWriter
{
public :
////////////////////////////////////////////////////////////
/// \brief Virtual destructor
///
////////////////////////////////////////////////////////////
virtual ~SoundFileWriter() {}
////////////////////////////////////////////////////////////
/// \brief Open a sound file for writing
///
/// \param filename Path of the file to open
/// \param sampleRate Sample rate of the sound
/// \param channelCount Number of channels of the sound
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
virtual bool open(const std::string& filename, unsigned int sampleRate, unsigned int channelCount) = 0;
////////////////////////////////////////////////////////////
/// \brief Write audio samples to the open file
///
/// \param samples Pointer to the sample array to write
/// \param count Number of samples to write
///
////////////////////////////////////////////////////////////
virtual void write(const Int16* samples, Uint64 count) = 0;
};
} // namespace sf
#endif // SFML_SOUNDFILEWRITER_HPP
////////////////////////////////////////////////////////////
/// \class sf::SoundFileWriter
/// \ingroup audio
///
/// This class allows users to write audio file formats not natively
/// supported by SFML, and thus extend the set of supported writable
/// audio formats.
///
/// A valid sound file writer must override the open and write functions,
/// as well as providing a static check function; the latter is used by
/// SFML to find a suitable writer for a given filename.
///
/// To register a new writer, use the sf::SoundFileFactory::registerWriter
/// template function.
///
/// Usage example:
/// \code
/// class MySoundFileWriter : public sf::SoundFileWriter
/// {
/// public:
///
/// static bool check(const std::string& filename)
/// {
/// // typically, check the extension
/// // return true if the writer can handle the format
/// }
///
/// virtual bool open(const std::string& filename, unsigned int sampleRate, unsigned int channelCount)
/// {
/// // open the file 'filename' for writing,
/// // write the given sample rate and channel count to the file header
/// // return true on success
/// }
///
/// virtual void write(const sf::Int16* samples, sf::Uint64 count)
/// {
/// // write 'count' samples stored at address 'samples',
/// // convert them (for example to normalized float) if the format requires it
/// }
/// };
///
/// sf::SoundFileFactory::registerWriter<MySoundFileWriter>();
/// \endcode
///
/// \see sf::OutputSoundFile, sf::SoundFileFactory, sf::SoundFileReader
///
////////////////////////////////////////////////////////////

141
include/SFML/System/FileInputStream.hpp Normal file
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@ -0,0 +1,141 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
#ifndef SFML_FILEINPUTSTREAM_HPP
#define SFML_FILEINPUTSTREAM_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Config.hpp>
#include <SFML/System/InputStream.hpp>
#include <fstream>
namespace sf
{
////////////////////////////////////////////////////////////
/// \brief Implementation of input stream based on a file
///
////////////////////////////////////////////////////////////
class FileInputStream : public InputStream
{
public :
////////////////////////////////////////////////////////////
/// \brief Open the stream from a file path
///
/// \param filename Name of the file to open
///
/// \return True on success, false on error
///
////////////////////////////////////////////////////////////
bool open(const std::string& filename);
////////////////////////////////////////////////////////////
/// \brief Read data from the stream
///
/// After reading, the stream's reading position must be
/// advanced by the amount of bytes read.
///
/// \param data Buffer where to copy the read data
/// \param size Desired number of bytes to read
///
/// \return The number of bytes actually read, or -1 on error
///
////////////////////////////////////////////////////////////
virtual Int64 read(void* data, Int64 size);
////////////////////////////////////////////////////////////
/// \brief Change the current reading position
///
/// \param position The position to seek to, from the beginning
///
/// \return The position actually sought to, or -1 on error
///
////////////////////////////////////////////////////////////
virtual Int64 seek(Int64 position);
////////////////////////////////////////////////////////////
/// \brief Get the current reading position in the stream
///
/// \return The current position, or -1 on error.
///
////////////////////////////////////////////////////////////
virtual Int64 tell();
////////////////////////////////////////////////////////////
/// \brief Return the size of the stream
///
/// \return The total number of bytes available in the stream, or -1 on error
///
////////////////////////////////////////////////////////////
virtual Int64 getSize();
private:
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
std::ifstream m_file; ///< Standard file stream
};
} // namespace sf
#endif // SFML_FILEINPUTSTREAM_HPP
////////////////////////////////////////////////////////////
/// \class FileInputStream
/// \ingroup system
///
/// This class is a specialization of InputStream that
/// reads from a file on disk.
///
/// It wraps a file in the common InputStream interface
/// and therefore allows to use generic classes or functions
/// that accept such a stream, with a file on disk as the data
/// source.
///
/// In addition to the virtual functions inherited from
/// InputStream, FileInputStream adds a function to
/// specify the file to open.
///
/// SFML resource classes can usually be loaded directly from
/// a filename, so this class shouldn't be useful to you unless
/// you create your own algorithms that operate on a InputStream.
///
/// Usage example:
/// \code
/// void process(InputStream& stream);
///
/// FileStream stream;
/// if (stream.open("some_file.dat"))
/// process(stream);
/// \endcode
///
/// InputStream, MemoryStream
///
////////////////////////////////////////////////////////////

3
include/SFML/System/InputStream.hpp
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@ -29,6 +29,7 @@
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Config.hpp>
#include <SFML/System/Export.hpp>
namespace sf
@ -37,7 +38,7 @@ namespace sf
/// \brief Abstract class for custom file input streams
///
////////////////////////////////////////////////////////////
class InputStream
class SFML_SYSTEM_API InputStream
{
public:

147
include/SFML/System/MemoryInputStream.hpp Normal file
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@ -0,0 +1,147 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
#ifndef SFML_MEMORYINPUTSTREAM_HPP
#define SFML_MEMORYINPUTSTREAM_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Config.hpp>
#include <SFML/System/InputStream.hpp>
#include <cstdlib>
namespace sf
{
////////////////////////////////////////////////////////////
/// \brief Implementation of input stream based on a memory chunk
///
////////////////////////////////////////////////////////////
class MemoryInputStream : public InputStream
{
public :
////////////////////////////////////////////////////////////
/// \brief Default constructor
///
////////////////////////////////////////////////////////////
MemoryInputStream();
////////////////////////////////////////////////////////////
/// \brief Open the stream from its data
///
/// \param data Pointer to the data in memory
/// \param sizeInBytes Size of the data, in bytes
///
////////////////////////////////////////////////////////////
void open(const void* data, std::size_t sizeInBytes);
////////////////////////////////////////////////////////////
/// \brief Read data from the stream
///
/// After reading, the stream's reading position must be
/// advanced by the amount of bytes read.
///
/// \param data Buffer where to copy the read data
/// \param size Desired number of bytes to read
///
/// \return The number of bytes actually read, or -1 on error
///
////////////////////////////////////////////////////////////
virtual Int64 read(void* data, Int64 size);
////////////////////////////////////////////////////////////
/// \brief Change the current reading position
///
/// \param position The position to seek to, from the beginning
///
/// \return The position actually sought to, or -1 on error
///
////////////////////////////////////////////////////////////
virtual Int64 seek(Int64 position);
////////////////////////////////////////////////////////////
/// \brief Get the current reading position in the stream
///
/// \return The current position, or -1 on error.
///
////////////////////////////////////////////////////////////
virtual Int64 tell();
////////////////////////////////////////////////////////////
/// \brief Return the size of the stream
///
/// \return The total number of bytes available in the stream, or -1 on error
///
////////////////////////////////////////////////////////////
virtual Int64 getSize();
private:
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
const char* m_data; ///< Pointer to the data in memory
Int64 m_size; ///< Total size of the data
Int64 m_offset; ///< Current reading position
};
} // namespace sf
#endif // SFML_MEMORYINPUTSTREAM_HPP
////////////////////////////////////////////////////////////
/// \class MemoryeInputStream
/// \ingroup system
///
/// This class is a specialization of InputStream that
/// reads from data in memory.
///
/// It wraps a memory chunk in the common InputStream interface
/// and therefore allows to use generic classes or functions
/// that accept such a stream, with content already loaded in memory.
///
/// In addition to the virtual functions inherited from
/// InputStream, MemoryInputStream adds a function to
/// specify the pointer and size of the data in memory.
///
/// SFML resource classes can usually be loaded directly from
/// memory, so this class shouldn't be useful to you unless
/// you create your own algorithms that operate on a InputStream.
///
/// Usage example:
/// \code
/// void process(InputStream& stream);
///
/// MemoryStream stream;
/// stream.open(thePtr, theSize);
/// process(stream);
/// \endcode
///
/// InputStream, FileStream
///
////////////////////////////////////////////////////////////

2
license.txt
View File

@ -29,8 +29,8 @@ External libraries used by SFML
-------------------------------
* OpenAL-Soft is under the LGPL license
* libsndfile is under the LGPL license
* libjpeg is public domain
* stb_image and stb_image_write are public domain
* freetype is under the FreeType license or the GPL license
* GLEW is under the modified BSD License, the Mesa 3-D License (MIT License), and the Khronos License (MIT License)
* libogg and libvorbis are under the BSD license

38
src/SFML/Audio/CMakeLists.txt
View File

@ -21,8 +21,10 @@ set(SRC
${INCROOT}/SoundBuffer.hpp
${SRCROOT}/SoundBufferRecorder.cpp
${INCROOT}/SoundBufferRecorder.hpp
${SRCROOT}/SoundFile.cpp
${SRCROOT}/SoundFile.hpp
${SRCROOT}/InputSoundFile.cpp
${INCROOT}/InputSoundFile.hpp
${SRCROOT}/OutputSoundFile.cpp
${INCROOT}/OutputSoundFile.hpp
${SRCROOT}/SoundRecorder.cpp
${INCROOT}/SoundRecorder.hpp
${SRCROOT}/SoundSource.cpp
@ -32,36 +34,52 @@ set(SRC
)
source_group("" FILES ${SRC})
set(CODECS_SRC
${SRCROOT}/SoundFileFactory.cpp
${INCROOT}/SoundFileFactory.hpp
${INCROOT}/SoundFileFactory.inl
${INCROOT}/SoundFileReader.hpp
${SRCROOT}/SoundFileReaderOgg.hpp
${SRCROOT}/SoundFileReaderOgg.cpp
${SRCROOT}/SoundFileReaderWav.hpp
${SRCROOT}/SoundFileReaderWav.cpp
${INCROOT}/SoundFileWriter.hpp
${SRCROOT}/SoundFileWriterWav.hpp
${SRCROOT}/SoundFileWriterWav.cpp
${SRCROOT}/SoundFileWriterOgg.hpp
${SRCROOT}/SoundFileWriterOgg.cpp
)
source_group("codecs" FILES ${CODECS_SRC})
# let CMake know about our additional audio libraries paths (on Windows and OSX)
if(SFML_OS_WINDOWS)
set(CMAKE_INCLUDE_PATH ${CMAKE_INCLUDE_PATH} "${PROJECT_SOURCE_DIR}/extlibs/headers/AL")
set(CMAKE_INCLUDE_PATH ${CMAKE_INCLUDE_PATH} "${PROJECT_SOURCE_DIR}/extlibs/headers/libsndfile/windows")
elseif(SFML_OS_MACOSX)
set(CMAKE_INCLUDE_PATH ${CMAKE_INCLUDE_PATH} "${PROJECT_SOURCE_DIR}/extlibs/headers/libsndfile/osx")
set(CMAKE_LIBRARY_PATH ${CMAKE_LIBRARY_PATH} "${PROJECT_SOURCE_DIR}/extlibs/libs-osx/Frameworks")
elseif(SFML_OS_ANDROID)
set(CMAKE_INCLUDE_PATH ${CMAKE_INCLUDE_PATH} "${PROJECT_SOURCE_DIR}/extlibs/headers/AL")
set(CMAKE_INCLUDE_PATH ${CMAKE_INCLUDE_PATH} "${PROJECT_SOURCE_DIR}/extlibs/headers/libsndfile/android")
endif()
# find external libraries
if(NOT SFML_OS_ANDROID)
find_package(OpenAL REQUIRED)
find_package(Sndfile REQUIRED)
find_package(Vorbis REQUIRED)
else()
find_host_package(OpenAL REQUIRED)
find_host_package(Sndfile REQUIRED)
find_host_package(Vorbis REQUIRED)
endif()
include_directories(${OPENAL_INCLUDE_DIR} ${SNDFILE_INCLUDE_DIR})
include_directories(${OPENAL_INCLUDE_DIR})
include_directories(${VORBIS_INCLUDE_DIRS})
add_definitions(-DOV_EXCLUDE_STATIC_CALLBACKS) # avoids warnings in vorbisfile.h
# build the list of external libraries to link
if(SFML_OS_ANDROID)
list(APPEND AUDIO_EXT_LIBS android OpenSLES)
endif()
list(APPEND AUDIO_EXT_LIBS ${OPENAL_LIBRARY} ${SNDFILE_LIBRARY})
list(APPEND AUDIO_EXT_LIBS ${OPENAL_LIBRARY} ${VORBIS_LIBRARIES})
# define the sfml-audio target
sfml_add_library(sfml-audio
SOURCES ${SRC}
SOURCES ${SRC} ${CODECS_SRC}
DEPENDS sfml-system
EXTERNAL_LIBS ${AUDIO_EXT_LIBS})

257
src/SFML/Audio/InputSoundFile.cpp Normal file
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@ -0,0 +1,257 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/InputSoundFile.hpp>
#include <SFML/Audio/SoundFileReader.hpp>
#include <SFML/Audio/SoundFileFactory.hpp>
#include <SFML/System/InputStream.hpp>
#include <SFML/System/FileInputStream.hpp>
#include <SFML/System/MemoryInputStream.hpp>
#include <SFML/System/Err.hpp>
namespace sf
{
////////////////////////////////////////////////////////////
InputSoundFile::InputSoundFile() :
m_reader (NULL),
m_stream (NULL),
m_streamOwned (false),
m_sampleCount (0),
m_channelCount(0),
m_sampleRate (0)
{
}
////////////////////////////////////////////////////////////
InputSoundFile::~InputSoundFile()
{
// Close the file in case it was open
close();
}
////////////////////////////////////////////////////////////
bool InputSoundFile::openFromFile(const std::string& filename)
{
// If the file is already open, first close it
close();
// Find a suitable reader for the file type
m_reader = SoundFileFactory::createReaderFromFilename(filename);
if (!m_reader)
{
err() << "Failed to open sound file \"" << filename << "\" (format not supported)" << std::endl;
return false;
}
// Wrap the file into a stream
FileInputStream* file = new FileInputStream;
m_stream = file;
m_streamOwned = true;
// Open it
if (!file->open(filename))
{
close();
return false;
}
// Pass the stream to the reader
SoundFileReader::Info info;
if (!m_reader->open(*file, info))
{
close();
return false;
}
// Retrieve the attributes of the open sound file
m_sampleCount = info.sampleCount;
m_channelCount = info.channelCount;
m_sampleRate = info.sampleRate;
return true;
}
////////////////////////////////////////////////////////////
bool InputSoundFile::openFromMemory(const void* data, std::size_t sizeInBytes)
{
// If the file is already open, first close it
close();
// Find a suitable reader for the file type
m_reader = SoundFileFactory::createReaderFromMemory(data, sizeInBytes);
if (!m_reader)
{
err() << "Failed to open sound file from memory (format not supported)" << std::endl;
return false;
}
// Wrap the memory file into a stream
MemoryInputStream* memory = new MemoryInputStream;
m_stream = memory;
m_streamOwned = true;
// Open it
memory->open(data, sizeInBytes);
// Pass the stream to the reader
SoundFileReader::Info info;
if (!m_reader->open(*memory, info))
{
close();
return false;
}
// Retrieve the attributes of the open sound file
m_sampleCount = info.sampleCount;
m_channelCount = info.channelCount;
m_sampleRate = info.sampleRate;
return true;
}
////////////////////////////////////////////////////////////
bool InputSoundFile::openFromStream(InputStream& stream)
{
// If the file is already open, first close it
close();
// Find a suitable reader for the file type
m_reader = SoundFileFactory::createReaderFromStream(stream);
if (!m_reader)
{
err() << "Failed to open sound file from stream (format not supported)" << std::endl;
return false;
}
// store the stream
m_stream = &stream;
m_streamOwned = false;
// Don't forget to reset the stream to its beginning before re-opening it
if (stream.seek(0) != 0)
{
err() << "Failed to open sound file from stream (cannot restart stream)" << std::endl;
return false;
}
// Pass the stream to the reader
SoundFileReader::Info info;
if (!m_reader->open(stream, info))
{
close();
return false;
}
// Retrieve the attributes of the open sound file
m_sampleCount = info.sampleCount;
m_channelCount = info.channelCount;
m_sampleRate = info.sampleRate;
return true;
}
////////////////////////////////////////////////////////////
Uint64 InputSoundFile::getSampleCount() const
{
return m_sampleCount;
}
////////////////////////////////////////////////////////////
unsigned int InputSoundFile::getChannelCount() const
{
return m_channelCount;
}
////////////////////////////////////////////////////////////
unsigned int InputSoundFile::getSampleRate() const
{
return m_sampleRate;
}
////////////////////////////////////////////////////////////
Time InputSoundFile::getDuration() const
{
return seconds(static_cast<float>(m_sampleCount) / m_channelCount / m_sampleRate);
}
////////////////////////////////////////////////////////////
void InputSoundFile::seek(Uint64 sampleOffset)
{
if (m_reader)
m_reader->seek(sampleOffset);
}
////////////////////////////////////////////////////////////
void InputSoundFile::seek(Time timeOffset)
{
seek(static_cast<Uint64>(timeOffset.asSeconds() * m_sampleRate * m_channelCount));
}
////////////////////////////////////////////////////////////
Uint64 InputSoundFile::read(Int16* samples, Uint64 maxCount)
{
if (m_reader && samples && maxCount)
return m_reader->read(samples, maxCount);
else
return 0;
}
////////////////////////////////////////////////////////////
void InputSoundFile::close()
{
// Destroy the reader
delete m_reader;
m_reader = NULL;
// Destroy the stream if we own it
if (m_streamOwned)
{
delete m_stream;
m_stream = NULL;
m_streamOwned = false;
}
// Reset the sound file attributes
m_sampleCount = 0;
m_channelCount = 0;
m_sampleRate = 0;
}
} // namespace sf

21
src/SFML/Audio/Music.cpp
View File

@ -27,7 +27,6 @@
////////////////////////////////////////////////////////////
#include <SFML/Audio/Music.hpp>
#include <SFML/Audio/ALCheck.hpp>
#include <SFML/Audio/SoundFile.hpp>
#include <SFML/System/Lock.hpp>
#include <SFML/System/Err.hpp>
#include <fstream>
@ -37,7 +36,7 @@ namespace sf
{
////////////////////////////////////////////////////////////
Music::Music() :
m_file (new priv::SoundFile),
m_file (),
m_duration()
{
@ -49,8 +48,6 @@ Music::~Music()
{
// We must stop before destroying the file
stop();
delete m_file;
}
@ -61,7 +58,7 @@ bool Music::openFromFile(const std::string& filename)
stop();
// Open the underlying sound file
if (!m_file->openRead(filename))
if (!m_file.openFromFile(filename))
return false;
// Perform common initializations
@ -78,7 +75,7 @@ bool Music::openFromMemory(const void* data, std::size_t sizeInBytes)
stop();
// Open the underlying sound file
if (!m_file->openRead(data, sizeInBytes))
if (!m_file.openFromMemory(data, sizeInBytes))
return false;
// Perform common initializations
@ -95,7 +92,7 @@ bool Music::openFromStream(InputStream& stream)
stop();
// Open the underlying sound file
if (!m_file->openRead(stream))
if (!m_file.openFromStream(stream))
return false;
// Perform common initializations
@ -119,7 +116,7 @@ bool Music::onGetData(SoundStream::Chunk& data)
// Fill the chunk parameters
data.samples = &m_samples[0];
data.sampleCount = m_file->read(&m_samples[0], m_samples.size());
data.sampleCount = static_cast<std::size_t>(m_file.read(&m_samples[0], m_samples.size()));
// Check if we have reached the end of the audio file
return data.sampleCount == m_samples.size();
@ -131,7 +128,7 @@ void Music::onSeek(Time timeOffset)
{
Lock lock(m_mutex);
m_file->seek(timeOffset);
m_file.seek(timeOffset);
}
@ -139,13 +136,13 @@ void Music::onSeek(Time timeOffset)
void Music::initialize()
{
// Compute the music duration
m_duration = seconds(static_cast<float>(m_file->getSampleCount()) / m_file->getSampleRate() / m_file->getChannelCount());
m_duration = m_file.getDuration();
// Resize the internal buffer so that it can contain 1 second of audio samples
m_samples.resize(m_file->getSampleRate() * m_file->getChannelCount());
m_samples.resize(m_file.getSampleRate() * m_file.getChannelCount());
// Initialize the stream
SoundStream::initialize(m_file->getChannelCount(), m_file->getSampleRate());
SoundStream::initialize(m_file.getChannelCount(), m_file.getSampleRate());
}
} // namespace sf

92
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@ -0,0 +1,92 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/OutputSoundFile.hpp>
#include <SFML/Audio/SoundFileWriter.hpp>
#include <SFML/Audio/SoundFileFactory.hpp>
#include <SFML/System/Err.hpp>
namespace sf
{
////////////////////////////////////////////////////////////
OutputSoundFile::OutputSoundFile() :
m_writer(NULL)
{
}
////////////////////////////////////////////////////////////
OutputSoundFile::~OutputSoundFile()
{
// Close the file in case it was open
close();
}
////////////////////////////////////////////////////////////
bool OutputSoundFile::openFromFile(const std::string& filename, unsigned int sampleRate, unsigned int channelCount)
{
// If the file is already open, first close it
close();
// Find a suitable writer for the file type
m_writer = SoundFileFactory::createWriterFromFilename(filename);
if (!m_writer)
{
err() << "Failed to open sound file \"" << filename << "\" (format not supported)" << std::endl;
return false;
}
// Pass the stream to the reader
if (!m_writer->open(filename, sampleRate, channelCount))
{
close();
return false;
}
return true;
}
////////////////////////////////////////////////////////////
void OutputSoundFile::write(const Int16* samples, Uint64 count)
{
if (m_writer && samples && count)
m_writer->write(samples, count);
}
////////////////////////////////////////////////////////////
void OutputSoundFile::close()
{
// Destroy the reader
delete m_writer;
m_writer = NULL;
}
} // namespace sf

29
src/SFML/Audio/SoundBuffer.cpp
View File

@ -26,7 +26,8 @@
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundBuffer.hpp>
#include <SFML/Audio/SoundFile.hpp>
#include <SFML/Audio/InputSoundFile.hpp>
#include <SFML/Audio/OutputSoundFile.hpp>
#include <SFML/Audio/Sound.hpp>
#include <SFML/Audio/AudioDevice.hpp>
#include <SFML/Audio/ALCheck.hpp>
@ -77,8 +78,8 @@ SoundBuffer::~SoundBuffer()
////////////////////////////////////////////////////////////
bool SoundBuffer::loadFromFile(const std::string& filename)
{
priv::SoundFile file;
if (file.openRead(filename))
InputSoundFile file;
if (file.openFromFile(filename))
return initialize(file);
else
return false;
@ -88,8 +89,8 @@ bool SoundBuffer::loadFromFile(const std::string& filename)
////////////////////////////////////////////////////////////
bool SoundBuffer::loadFromMemory(const void* data, std::size_t sizeInBytes)
{
priv::SoundFile file;
if (file.openRead(data, sizeInBytes))
InputSoundFile file;
if (file.openFromMemory(data, sizeInBytes))
return initialize(file);
else
return false;
@ -99,8 +100,8 @@ bool SoundBuffer::loadFromMemory(const void* data, std::size_t sizeInBytes)
////////////////////////////////////////////////////////////
bool SoundBuffer::loadFromStream(InputStream& stream)
{
priv::SoundFile file;
if (file.openRead(stream))
InputSoundFile file;
if (file.openFromStream(stream))
return initialize(file);
else
return false;
@ -108,7 +109,7 @@ bool SoundBuffer::loadFromStream(InputStream& stream)
////////////////////////////////////////////////////////////
bool SoundBuffer::loadFromSamples(const Int16* samples, std::size_t sampleCount, unsigned int channelCount, unsigned int sampleRate)
bool SoundBuffer::loadFromSamples(const Int16* samples, Uint64 sampleCount, unsigned int channelCount, unsigned int sampleRate)
{
if (samples && sampleCount && channelCount && sampleRate)
{
@ -137,8 +138,8 @@ bool SoundBuffer::loadFromSamples(const Int16* samples, std::size_t sampleCount,
bool SoundBuffer::saveToFile(const std::string& filename) const
{
// Create the sound file in write mode
priv::SoundFile file;
if (file.openWrite(filename, getChannelCount(), getSampleRate()))
OutputSoundFile file;
if (file.openFromFile(filename, getSampleRate(), getChannelCount()))
{
// Write the samples to the opened file
file.write(&m_samples[0], m_samples.size());
@ -160,7 +161,7 @@ const Int16* SoundBuffer::getSamples() const
////////////////////////////////////////////////////////////
std::size_t SoundBuffer::getSampleCount() const
Uint64 SoundBuffer::getSampleCount() const
{
return m_samples.size();
}
@ -208,15 +209,15 @@ SoundBuffer& SoundBuffer::operator =(const SoundBuffer& right)
////////////////////////////////////////////////////////////
bool SoundBuffer::initialize(priv::SoundFile& file)
bool SoundBuffer::initialize(InputSoundFile& file)
{
// Retrieve the sound parameters
std::size_t sampleCount = file.getSampleCount();
Uint64 sampleCount = file.getSampleCount();
unsigned int channelCount = file.getChannelCount();
unsigned int sampleRate = file.getSampleRate();
// Read the samples from the provided file
m_samples.resize(sampleCount);
m_samples.resize(static_cast<std::size_t>(sampleCount));
if (file.read(&m_samples[0], sampleCount) == sampleCount)
{
// Update the internal buffer with the new samples

142
src/SFML/Audio/SoundFileFactory.cpp Normal file
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@ -0,0 +1,142 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundFileFactory.hpp>
#include <SFML/Audio/SoundFileReaderOgg.hpp>
#include <SFML/Audio/SoundFileWriterOgg.hpp>
#include <SFML/Audio/SoundFileReaderWav.hpp>
#include <SFML/Audio/SoundFileWriterWav.hpp>
#include <SFML/System/FileInputStream.hpp>
#include <SFML/System/MemoryInputStream.hpp>
namespace
{
// register all the built-in readers and writers if not already done
void ensureDefaultReadersWritersRegistered()
{
static bool registered = false;
if (!registered)
{
sf::SoundFileFactory::registerReader<sf::priv::SoundFileReaderOgg>();
sf::SoundFileFactory::registerWriter<sf::priv::SoundFileWriterOgg>();
sf::SoundFileFactory::registerReader<sf::priv::SoundFileReaderWav>();
sf::SoundFileFactory::registerWriter<sf::priv::SoundFileWriterWav>();
registered = true;
}
}
}
namespace sf
{
SoundFileFactory::ReaderFactoryArray SoundFileFactory::s_readers;
SoundFileFactory::WriterFactoryArray SoundFileFactory::s_writers;
////////////////////////////////////////////////////////////
SoundFileReader* SoundFileFactory::createReaderFromFilename(const std::string& filename)
{
// Register the built-in readers/writers on first call
ensureDefaultReadersWritersRegistered();
// Wrap the input file into a file stream
FileInputStream stream;
if (!stream.open(filename))
return NULL;
// Test the filename in all the registered factories
for (ReaderFactoryArray::const_iterator it = s_readers.begin(); it != s_readers.end(); ++it)
{
stream.seek(0);
if (it->check(stream))
return it->create();
}
// No suitable reader found
return NULL;
}
////////////////////////////////////////////////////////////
SoundFileReader* SoundFileFactory::createReaderFromMemory(const void* data, std::size_t sizeInBytes)
{
// Register the built-in readers/writers on first call
ensureDefaultReadersWritersRegistered();
// Wrap the memory file into a file stream
MemoryInputStream stream;
stream.open(data, sizeInBytes);
// Test the filename in all the registered factories
for (ReaderFactoryArray::const_iterator it = s_readers.begin(); it != s_readers.end(); ++it)
{
stream.seek(0);
if (it->check(stream))
return it->create();
}
// No suitable reader found
return NULL;
}
////////////////////////////////////////////////////////////
SoundFileReader* SoundFileFactory::createReaderFromStream(InputStream& stream)
{
// Register the built-in readers/writers on first call
ensureDefaultReadersWritersRegistered();
// Test the filename in all the registered factories
for (ReaderFactoryArray::const_iterator it = s_readers.begin(); it != s_readers.end(); ++it)
{
if (it->check(stream))
return it->create();
}
// No suitable reader found
return NULL;
}
////////////////////////////////////////////////////////////
SoundFileWriter* SoundFileFactory::createWriterFromFilename(const std::string& filename)
{
// Register the built-in readers/writers on first call
ensureDefaultReadersWritersRegistered();
// Test the filename in all the registered factories
for (WriterFactoryArray::const_iterator it = s_writers.begin(); it != s_writers.end(); ++it)
{
if (it->check(filename))
return it->create();
}
// No suitable writer found
return NULL;
}
} // namespace sf

179
src/SFML/Audio/SoundFileReaderOgg.cpp Normal file
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@ -0,0 +1,179 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundFileReaderOgg.hpp>
#include <SFML/System/MemoryInputStream.hpp>
#include <SFML/System/Err.hpp>
#include <algorithm>
#include <cctype>
#include <cassert>
namespace
{
size_t read(void* ptr, size_t size, size_t nmemb, void* data)
{
sf::InputStream* stream = static_cast<sf::InputStream*>(data);
return static_cast<std::size_t>(stream->read(ptr, size * nmemb));
}
int seek(void* data, ogg_int64_t offset, int whence)
{
sf::InputStream* stream = static_cast<sf::InputStream*>(data);
switch (whence)
{
case SEEK_SET:
break;
case SEEK_CUR:
offset += stream->tell();
break;
case SEEK_END:
offset = stream->getSize() - offset;
}
return static_cast<int>(stream->seek(offset));
}
long tell(void* data)
{
sf::InputStream* stream = static_cast<sf::InputStream*>(data);
return static_cast<long>(stream->tell());
}
static ov_callbacks callbacks = {&read, &seek, NULL, &tell};
}
namespace sf
{
namespace priv
{
////////////////////////////////////////////////////////////
bool SoundFileReaderOgg::check(InputStream& stream)
{
OggVorbis_File file;
if (ov_test_callbacks(&stream, &file, NULL, 0, callbacks) == 0)
{
ov_clear(&file);
return true;
}
else
{
return false;
}
}
////////////////////////////////////////////////////////////
SoundFileReaderOgg::SoundFileReaderOgg() :
m_vorbis (),
m_channelCount(0)
{
m_vorbis.datasource = NULL;
}
////////////////////////////////////////////////////////////
SoundFileReaderOgg::~SoundFileReaderOgg()
{
close();
}
////////////////////////////////////////////////////////////
bool SoundFileReaderOgg::open(InputStream& stream, Info& info)
{
// Open the Vorbis stream
int status = ov_open_callbacks(&stream, &m_vorbis, NULL, 0, callbacks);
if (status < 0)
{
err() << "Failed to open Vorbis file for reading" << std::endl;
return false;
}
// Retrieve the music attributes
vorbis_info* vorbisInfo = ov_info(&m_vorbis, -1);
info.channelCount = vorbisInfo->channels;
info.sampleRate = vorbisInfo->rate;
info.sampleCount = static_cast<std::size_t>(ov_pcm_total(&m_vorbis, -1) * vorbisInfo->channels);
// We must keep the channel count for the seek function
m_channelCount = info.channelCount;
return true;
}
////////////////////////////////////////////////////////////
void SoundFileReaderOgg::seek(Uint64 sampleOffset)
{
assert(m_vorbis.datasource);
ov_pcm_seek(&m_vorbis, sampleOffset / m_channelCount);
}
////////////////////////////////////////////////////////////
Uint64 SoundFileReaderOgg::read(Int16* samples, Uint64 maxCount)
{
assert(m_vorbis.datasource);
// Try to read the requested number of samples, stop only on error or end of file
Uint64 count = 0;
while (count < maxCount)
{
int bytesToRead = static_cast<int>(maxCount - count) * sizeof(Int16);
long bytesRead = ov_read(&m_vorbis, reinterpret_cast<char*>(samples), bytesToRead, 0, 2, 1, NULL);
if (bytesRead > 0)
{
long samplesRead = bytesRead / sizeof(Int16);
count += samplesRead;
samples += samplesRead;
}
else
{
// error or end of file
break;
}
}
return count;
}
////////////////////////////////////////////////////////////
void SoundFileReaderOgg::close()
{
if (m_vorbis.datasource)
{
ov_clear(&m_vorbis);
m_vorbis.datasource = NULL;
m_channelCount = 0;
}
}
} // namespace priv
} // namespace sf

124
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@ -0,0 +1,124 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
#ifndef SFML_SOUNDFILEREADEROGG_HPP
#define SFML_SOUNDFILEREADEROGG_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundFileReader.hpp>
#include <vorbis/vorbisfile.h>
namespace sf
{
namespace priv
{
////////////////////////////////////////////////////////////
/// \brief Implementation of sound file reader that handles ogg/vorbis files
///
////////////////////////////////////////////////////////////
class SoundFileReaderOgg : public SoundFileReader
{
public:
////////////////////////////////////////////////////////////
/// \brief Check if this reader can handle a file given by an input stream
///
/// \param stream Source stream to check
///
/// \return True if the file is supported by this reader
///
////////////////////////////////////////////////////////////
static bool check(InputStream& stream);
public :
////////////////////////////////////////////////////////////
/// \brief Default constructor
///
////////////////////////////////////////////////////////////
SoundFileReaderOgg();
////////////////////////////////////////////////////////////
/// \brief Destructor
///
////////////////////////////////////////////////////////////
~SoundFileReaderOgg();
////////////////////////////////////////////////////////////
/// \brief Open a sound file for reading
///
/// \param stream Source stream to read from
/// \param info Structure to fill with the properties of the loaded sound
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
virtual bool open(InputStream& stream, Info& info);
////////////////////////////////////////////////////////////
/// \brief Change the current read position to the given sample offset
///
/// If the given offset exceeds to total number of samples,
/// this function must jump to the end of the file.
///
/// \param sampleOffset Index of the sample to jump to, relative to the beginning
///
////////////////////////////////////////////////////////////
virtual void seek(Uint64 sampleOffset);
////////////////////////////////////////////////////////////
/// \brief Read audio samples from the open file
///
/// \param samples Pointer to the sample array to fill
/// \param maxCount Maximum number of samples to read
///
/// \return Number of samples actually read (may be less than \a maxCount)
///
////////////////////////////////////////////////////////////
virtual Uint64 read(Int16* samples, Uint64 maxCount);
private:
////////////////////////////////////////////////////////////
/// \brief Close the open Vorbis file
///
////////////////////////////////////////////////////////////
void close();
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
OggVorbis_File m_vorbis; // ogg/vorbis file handle
unsigned int m_channelCount; // number of channels of the open sound file
};
} // namespace priv
} // namespace sf
#endif // SFML_SOUNDFILEREADEROGG_HPP

266
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@ -0,0 +1,266 @@
////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundFileReaderWav.hpp>
#include <SFML/System/InputStream.hpp>
#include <SFML/System/Err.hpp>
#include <algorithm>
#include <cctype>
#include <cassert>
namespace
{
// The following functions read integers as little endian and
// return them in the host byte order
bool decode(sf::InputStream& stream, sf::Uint8& value)
{
return stream.read(&value, sizeof(value)) == sizeof(value);
}
bool decode(sf::InputStream& stream, sf::Int16& value)
{
unsigned char bytes[sizeof(value)];
if (stream.read(bytes, sizeof(bytes)) != sizeof(bytes))
return false;
value = bytes[0] | (bytes[1] << 8);
return true;
}
bool decode(sf::InputStream& stream, sf::Uint16& value)
{
unsigned char bytes[sizeof(value)];
if (stream.read(bytes, sizeof(bytes)) != sizeof(bytes))
return false;
value = bytes[0] | (bytes[1] << 8);
return true;
}
bool decode(sf::InputStream& stream, sf::Uint32& value)
{
unsigned char bytes[sizeof(value)];
if (stream.read(bytes, sizeof(bytes)) != sizeof(bytes))
return false;
value = bytes[0] | (bytes[1] << 8) | (bytes[2] << 16) | (bytes[3] << 24);
return true;
}
const sf::Uint64 mainChunkSize = 12;
}
namespace sf
{
namespace priv
{
////////////////////////////////////////////////////////////
bool SoundFileReaderWav::check(InputStream& stream)
{
char header[mainChunkSize];
if (stream.read(header, sizeof(header)) < sizeof(header))
return false;
return (header[0] == 'R') && (header[1] == 'I') && (header[2] == 'F') && (header[3] == 'F')
&& (header[8] == 'W') && (header[9] == 'A') && (header[10] == 'V') && (header[11] == 'E');
}
////////////////////////////////////////////////////////////
SoundFileReaderWav::SoundFileReaderWav() :
m_stream (NULL),
m_bytesPerSample(0),
m_dataStart (0)
{
}
////////////////////////////////////////////////////////////
bool SoundFileReaderWav::open(InputStream& stream, Info& info)
{
m_stream = &stream;
if (!parseHeader(info))
{
err() << "Failed to open WAV sound file (invalid or unsupported file)" << std::endl;
return false;
}
return true;
}
////////////////////////////////////////////////////////////
void SoundFileReaderWav::seek(Uint64 sampleOffset)
{
assert(m_stream);
m_stream->seek(m_dataStart + sampleOffset * m_bytesPerSample);
}
////////////////////////////////////////////////////////////
Uint64 SoundFileReaderWav::read(Int16* samples, Uint64 maxCount)
{
assert(m_stream);
Uint64 count = 0;
while (count < maxCount)
{
switch (m_bytesPerSample)
{
case 1:
{
Uint8 sample = 0;
if (decode(*m_stream, sample))
*samples++ = (static_cast<Int16>(sample) - 128) << 8;
else
return count;
break;
}
case 2:
{
Int16 sample = 0;
if (decode(*m_stream, sample))
*samples++ = sample;
else
return count;
break;
}
}
++count;
}
return count;
}
////////////////////////////////////////////////////////////
bool SoundFileReaderWav::parseHeader(Info& info)
{
assert(m_stream);
// If we are here, it means that the first part of the header
// (the format) has already been checked
char mainChunk[mainChunkSize];
if (m_stream->read(mainChunk, sizeof(mainChunk)) != sizeof(mainChunk))
return false;
// Parse all the sub-chunks
bool dataChunkFound = false;
while (!dataChunkFound)
{
// Parse the sub-chunk id and size
char subChunkId[4];
if (m_stream->read(subChunkId, sizeof(subChunkId)) != sizeof(subChunkId))
return false;
Uint32 subChunkSize = 0;
if (!decode(*m_stream, subChunkSize))
return false;
// Check which chunk it is
if ((subChunkId[0] == 'f') && (subChunkId[1] == 'm') && (subChunkId[2] == 't') && (subChunkId[3] == ' '))
{
// "fmt" chunk
// Audio format
Uint16 format = 0;
if (!decode(*m_stream, format))
return false;
if (format != 1) // PCM
return false;
// Channel count
Uint16 channelCount = 0;
if (!decode(*m_stream, channelCount))
return false;
info.channelCount = channelCount;
// Sample rate
Uint32 sampleRate = 0;
if (!decode(*m_stream, sampleRate))
return false;
info.sampleRate = sampleRate;
// Byte rate
Uint32 byteRate = 0;
if (!decode(*m_stream, byteRate))
return false;
// Block align
Uint16 blockAlign = 0;
if (!decode(*m_stream, blockAlign))
return false;
// Bits per sample
Uint16 bitsPerSample = 0;
if (!decode(*m_stream, bitsPerSample))
return false;
m_bytesPerSample = bitsPerSample / 8;
// Skip potential extra information (should not exist for PCM)
if (subChunkSize > 16)
{
char buffer[1024];
if (m_stream->read(buffer, subChunkSize - 16) != subChunkSize - 16)
return false;
}
}
else if ((subChunkId[0] == 'd') && (subChunkId[1] == 'a') && (subChunkId[2] == 't') && (subChunkId[3] == 'a'))
{
// "data" chunk
// Compute the total number of samples
info.sampleCount = subChunkSize / info.channelCount / m_bytesPerSample;
// Store the starting position of samples in the file
m_dataStart = m_stream->tell();
dataChunkFound = true;
}
else
{
// unknown chunk, skip it
char buffer[1024];
if (m_stream->read(buffer, subChunkSize) != subChunkSize)
return false;
}
}
return true;
}
} // namespace priv
} // namespace sf

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////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
#ifndef SFML_SOUNDFILEREADERWAV_HPP
#define SFML_SOUNDFILEREADERWAV_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundFileReader.hpp>
#include <string>
namespace sf
{
namespace priv
{
////////////////////////////////////////////////////////////
/// \brief Implementation of sound file reader that handles wav files
///
////////////////////////////////////////////////////////////
class SoundFileReaderWav : public SoundFileReader
{
public:
////////////////////////////////////////////////////////////
/// \brief Check if this reader can handle a file given by an input stream
///
/// \param stream Source stream to check
///
/// \return True if the file is supported by this reader
///
////////////////////////////////////////////////////////////
static bool check(InputStream& stream);
public :
////////////////////////////////////////////////////////////
/// \brief Default constructor
///
////////////////////////////////////////////////////////////
SoundFileReaderWav();
////////////////////////////////////////////////////////////
/// \brief Open a sound file for reading
///
/// \param stream Stream to open
/// \param info Structure to fill with the attributes of the loaded sound
///
////////////////////////////////////////////////////////////
virtual bool open(sf::InputStream& stream, Info& info);
////////////////////////////////////////////////////////////
/// \brief Change the current read position to the given sample offset
///
/// If the given offset exceeds to total number of samples,
/// this function must jump to the end of the file.
///
/// \param sampleOffset Index of the sample to jump to, relative to the beginning
///
////////////////////////////////////////////////////////////
virtual void seek(Uint64 sampleOffset);
////////////////////////////////////////////////////////////
/// \brief Read audio samples from the open file
///
/// \param samples Pointer to the sample array to fill
/// \param maxCount Maximum number of samples to read
///
/// \return Number of samples actually read (may be less than \a maxCount)
///
////////////////////////////////////////////////////////////
virtual Uint64 read(Int16* samples, Uint64 maxCount);
private:
////////////////////////////////////////////////////////////
/// \brief Read the header of the open file
///
/// \param info Attributes of the sound file
///
/// \return True on success, false on error
///
////////////////////////////////////////////////////////////
bool parseHeader(Info& info);
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
InputStream* m_stream; ///< Source stream to read from
unsigned int m_bytesPerSample; ///< Size of a sample, in bytes
Uint64 m_dataStart; ///< Starting position of the audio data in the open file
};
} // namespace priv
} // namespace sf
#endif // SFML_SOUNDFILEREADERWAV_HPP

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////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundFileWriterOgg.hpp>
#include <SFML/System/Err.hpp>
#include <algorithm>
#include <cctype>
#include <cstdlib>
#include <cassert>
namespace sf
{
namespace priv
{
////////////////////////////////////////////////////////////
bool SoundFileWriterOgg::check(const std::string& filename)
{
std::string extension = filename.substr(filename.find_last_of(".") + 1);
std::transform(extension.begin(), extension.end(), extension.begin(), ::tolower);
return extension == "ogg";
}
////////////////////////////////////////////////////////////
SoundFileWriterOgg::SoundFileWriterOgg() :
m_channelCount(0),
m_file (),
m_ogg (),
m_vorbis (),
m_state ()
{
}
////////////////////////////////////////////////////////////
SoundFileWriterOgg::~SoundFileWriterOgg()
{
close();
}
////////////////////////////////////////////////////////////
bool SoundFileWriterOgg::open(const std::string& filename, unsigned int sampleRate, unsigned int channelCount)
{
// Save the channel count
m_channelCount = channelCount;
// Initialize the ogg/vorbis stream
ogg_stream_init(&m_ogg, std::rand());
vorbis_info_init(&m_vorbis);
// Setup the encoder: VBR, automatic bitrate management
// Quality is in range [-1 .. 1], 0.4 gives ~128 kbps for a 44 KHz stereo sound
int status = vorbis_encode_init_vbr(&m_vorbis, channelCount, sampleRate, 0.4f);
if (status < 0)
{
err() << "Failed to write ogg/vorbis file \"" << filename << "\" (unsupported bitrate)" << std::endl;
close();
return false;
}
vorbis_analysis_init(&m_state, &m_vorbis);
// Open the file after the vorbis setup is ok
m_file.open(filename.c_str(), std::ios::binary);
if (!m_file)
{
err() << "Failed to write ogg/vorbis file \"" << filename << "\" (cannot open file)" << std::endl;
close();
return false;
}
// Generate header metadata (leave it empty)
vorbis_comment comment;
vorbis_comment_init(&comment);
// Generate the header packets
ogg_packet header, headerComm, headerCode;
status = vorbis_analysis_headerout(&m_state, &comment, &header, &headerComm, &headerCode);
vorbis_comment_clear(&comment);
if (status < 0)
{
err() << "Failed to write ogg/vorbis file \"" << filename << "\" (cannot generate the headers)" << std::endl;
close();
return false;
}
// Write the header packets to the ogg stream
ogg_stream_packetin(&m_ogg, &header);
ogg_stream_packetin(&m_ogg, &headerComm);
ogg_stream_packetin(&m_ogg, &headerCode);
// This ensures the actual audio data will start on a new page, as per spec
ogg_page page;
while (ogg_stream_flush(&m_ogg, &page) > 0)
{
m_file.write(reinterpret_cast<const char*>(page.header), page.header_len);
m_file.write(reinterpret_cast<const char*>(page.body), page.body_len);
}
return true;
}
////////////////////////////////////////////////////////////
void SoundFileWriterOgg::write(const Int16* samples, Uint64 count)
{
// Prepare a buffer to hold our samples
int frameCount = static_cast<int>(count / m_channelCount);
float** buffer = vorbis_analysis_buffer(&m_state, frameCount);
assert(buffer);
// Write the samples to the buffer, converted to float
for (int i = 0; i < frameCount; ++i)
for (unsigned int j = 0; j < m_channelCount; ++j)
buffer[j][i] = *samples++ / 32767.0f;
// Tell the library how many samples we've written
vorbis_analysis_wrote(&m_state, frameCount);
// Flush any produced block
flushBlocks();
}
////////////////////////////////////////////////////////////
void SoundFileWriterOgg::flushBlocks()
{
// Let the library divide uncompressed data into blocks, and process them
vorbis_block block;
vorbis_block_init(&m_state, &block);
while (vorbis_analysis_blockout(&m_state, &block) == 1)
{
// Let the automatic bitrate management do its job
vorbis_analysis(&block, NULL);
vorbis_bitrate_addblock(&block);
// Get new packets from the bitrate management engine
ogg_packet packet;
while (vorbis_bitrate_flushpacket(&m_state, &packet))
{
// Write the packet to the ogg stream
ogg_stream_packetin(&m_ogg, &packet);
// If the stream produced new pages, write them to the output file
ogg_page page;
while (ogg_stream_flush(&m_ogg, &page) > 0)
{
m_file.write(reinterpret_cast<const char*>(page.header), page.header_len);
m_file.write(reinterpret_cast<const char*>(page.body), page.body_len);
}
}
}
// Clear the allocated block
vorbis_block_clear(&block);
}
////////////////////////////////////////////////////////////
void SoundFileWriterOgg::close()
{
if (m_file.is_open())
{
// Submit an empty packet to mark the end of stream
vorbis_analysis_wrote(&m_state, 0);
flushBlocks();
// Close the file
m_file.close();
}
// Clear all the ogg/vorbis structures
ogg_stream_clear(&m_ogg);
vorbis_dsp_clear(&m_state);
vorbis_info_clear(&m_vorbis);
}
} // namespace priv
} // namespace sf

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////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
#ifndef SFML_SOUNDFILEWRITEROGG_HPP
#define SFML_SOUNDFILEWRITEROGG_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundFileWriter.hpp>
#include <vorbis/vorbisenc.h>
#include <fstream>
namespace sf
{
namespace priv
{
////////////////////////////////////////////////////////////
/// \brief Implementation of sound file writer that handles ogg files
///
////////////////////////////////////////////////////////////
class SoundFileWriterOgg : public SoundFileWriter
{
public:
////////////////////////////////////////////////////////////
/// \brief Check if this writer can handle a file on disk
///
/// \param filename Path of the sound file to check
///
/// \return True if the file can be written by this writer
///
////////////////////////////////////////////////////////////
static bool check(const std::string& filename);
public :
////////////////////////////////////////////////////////////
/// \brief Default constructor
///
////////////////////////////////////////////////////////////
SoundFileWriterOgg();
////////////////////////////////////////////////////////////
/// \brief Destructor
///
////////////////////////////////////////////////////////////
~SoundFileWriterOgg();
////////////////////////////////////////////////////////////
/// \brief Open a sound file for writing
///
/// \param filename Path of the file to open
/// \param sampleRate Sample rate of the sound
/// \param channelCount Number of channels of the sound
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
virtual bool open(const std::string& filename, unsigned int sampleRate, unsigned int channelCount);
////////////////////////////////////////////////////////////
/// \brief Write audio samples to the open file
///
/// \param samples Pointer to the sample array to write
/// \param count Number of samples to write
///
////////////////////////////////////////////////////////////
virtual void write(const Int16* samples, Uint64 count);
private:
////////////////////////////////////////////////////////////
/// \brief Flush blocks produced by the ogg stream, if any
///
////////////////////////////////////////////////////////////
void flushBlocks();
////////////////////////////////////////////////////////////
/// \brief Close the file
///
////////////////////////////////////////////////////////////
void close();
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
unsigned int m_channelCount; // channel count of the sound being written
std::ofstream m_file; // output file
ogg_stream_state m_ogg; // ogg stream
vorbis_info m_vorbis; // vorbis handle
vorbis_dsp_state m_state; // current encoding state
};
} // namespace priv
} // namespace sf
#endif // SFML_SOUNDFILEWRITEROGG_HPP

207
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////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundFileWriterWav.hpp>
#include <SFML/System/Err.hpp>
#include <algorithm>
#include <cctype>
#include <cassert>
namespace
{
// The following functions takes integers in host byte order
// and writes them to a stream as little endian
void encode(std::ostream& stream, sf::Int16 value)
{
unsigned char bytes[] =
{
static_cast<unsigned char>(value & 0xFF),
static_cast<unsigned char>(value >> 8)
};
stream.write(reinterpret_cast<const char*>(bytes), sizeof(bytes));
}
void encode(std::ostream& stream, sf::Uint16 value)
{
unsigned char bytes[] =
{
static_cast<unsigned char>(value & 0xFF),
static_cast<unsigned char>(value >> 8)
};
stream.write(reinterpret_cast<const char*>(bytes), sizeof(bytes));
}
void encode(std::ostream& stream, sf::Uint32 value)
{
unsigned char bytes[] =
{
static_cast<unsigned char>(value & 0x000000FF),
static_cast<unsigned char>((value & 0x0000FF00) >> 8),
static_cast<unsigned char>((value & 0x00FF0000) >> 16),
static_cast<unsigned char>((value & 0xFF000000) >> 24),
};
stream.write(reinterpret_cast<const char*>(bytes), sizeof(bytes));
}
}
namespace sf
{
namespace priv
{
////////////////////////////////////////////////////////////
bool SoundFileWriterWav::check(const std::string& filename)
{
std::string extension = filename.substr(filename.find_last_of(".") + 1);
std::transform(extension.begin(), extension.end(), extension.begin(), ::tolower);
return extension == "wav";
}
////////////////////////////////////////////////////////////
SoundFileWriterWav::SoundFileWriterWav() :
m_file (),
m_sampleCount (0),
m_channelCount(0)
{
}
////////////////////////////////////////////////////////////
SoundFileWriterWav::~SoundFileWriterWav()
{
close();
}
////////////////////////////////////////////////////////////
bool SoundFileWriterWav::open(const std::string& filename, unsigned int sampleRate, unsigned int channelCount)
{
// Open the file
m_file.open(filename.c_str(), std::ios::binary);
if (!m_file)
{
err() << "Failed to open WAV sound file \"" << filename << "\" for writing" << std::endl;
return false;
}
// Write the header
if (!writeHeader(sampleRate, channelCount))
{
err() << "Failed to write header of WAV sound file \"" << filename << "\"" << std::endl;
return false;
}
// Save the channel count
m_channelCount = channelCount;
return true;
}
////////////////////////////////////////////////////////////
void SoundFileWriterWav::write(const Int16* samples, Uint64 count)
{
assert(m_file.good());
m_sampleCount += count;
while (count--)
encode(m_file, *samples++);
}
////////////////////////////////////////////////////////////
bool SoundFileWriterWav::writeHeader(unsigned int sampleRate, unsigned int channelCount)
{
assert(m_file.good());
// Write the main chunk ID
char mainChunkId[4] = {'R', 'I', 'F', 'F'};
m_file.write(mainChunkId, sizeof(mainChunkId));
// Write the main chunk header
Uint32 mainChunkSize = 0; // placeholder, will be written later
encode(m_file, mainChunkSize);
char mainChunkFormat[4] = {'W', 'A', 'V', 'E'};
m_file.write(mainChunkFormat, sizeof(mainChunkFormat));
// Write the sub-chunk 1 ("format") id and size
char fmtChunkId[4] = {'f', 'm', 't', ' '};
m_file.write(fmtChunkId, sizeof(fmtChunkId));
Uint32 fmtChunkSize = 16;
encode(m_file, fmtChunkSize);
// Write the format (PCM)
Uint16 format = 1;
encode(m_file, format);
// Write the sound attributes
encode(m_file, static_cast<Uint16>(channelCount));
encode(m_file, static_cast<Uint32>(sampleRate));
Uint32 byteRate = sampleRate * channelCount * 2;
encode(m_file, byteRate);
Uint16 blockAlign = channelCount * 2;
encode(m_file, blockAlign);
Uint16 bitsPerSample = 16;
encode(m_file, bitsPerSample);
// Write the sub-chunk 2 ("data") id and size
char dataChunkId[4] = {'d', 'a', 't', 'a'};
m_file.write(dataChunkId, sizeof(dataChunkId));
Uint32 dataChunkSize = 0; // placeholder, will be written later
encode(m_file, dataChunkSize);
return true;
}
////////////////////////////////////////////////////////////
void SoundFileWriterWav::close()
{
// If the file is open, finalize the header and close it
if (m_file.is_open())
{
m_file.flush();
// Update the main chunk size and data sub-chunk size
Uint32 dataChunkSize = static_cast<Uint32>(m_sampleCount * m_channelCount * 2);
Uint32 mainChunkSize = dataChunkSize + 36;
m_file.seekp(4);
encode(m_file, mainChunkSize);
m_file.seekp(40);
encode(m_file, dataChunkSize);
m_file.close();
}
}
} // namespace priv
} // namespace sf

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////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
#ifndef SFML_SOUNDFILEWRITERWAV_HPP
#define SFML_SOUNDFILEWRITERWAV_HPP
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/Audio/SoundFileWriter.hpp>
#include <fstream>
#include <string>
namespace sf
{
namespace priv
{
////////////////////////////////////////////////////////////
/// \brief Implementation of sound file writer that handles wav files
///
////////////////////////////////////////////////////////////
class SoundFileWriterWav : public SoundFileWriter
{
public:
////////////////////////////////////////////////////////////
/// \brief Check if this writer can handle a file on disk
///
/// \param filename Path of the sound file to check
///
/// \return True if the file can be written by this writer
///
////////////////////////////////////////////////////////////
static bool check(const std::string& filename);
public :
////////////////////////////////////////////////////////////
/// \brief Default constructor
///
////////////////////////////////////////////////////////////
SoundFileWriterWav();
////////////////////////////////////////////////////////////
/// \brief Destructor
///
////////////////////////////////////////////////////////////
~SoundFileWriterWav();
////////////////////////////////////////////////////////////
/// \brief Open a sound file for writing
///
/// \param filename Path of the file to open
/// \param sampleRate Sample rate of the sound
/// \param channelCount Number of channels of the sound
///
/// \return True if the file was successfully opened
///
////////////////////////////////////////////////////////////
virtual bool open(const std::string& filename, unsigned int sampleRate, unsigned int channelCount);
////////////////////////////////////////////////////////////
/// \brief Write audio samples to the open file
///
/// \param samples Pointer to the sample array to write
/// \param count Number of samples to write
///
////////////////////////////////////////////////////////////
virtual void write(const Int16* samples, Uint64 count);
private:
////////////////////////////////////////////////////////////
/// \brief Write the header of the open file
///
/// \param sampleRate Sample rate of the sound
/// \param channelCount Number of channels of the sound
///
/// \return True on success, false on error
///
////////////////////////////////////////////////////////////
bool writeHeader(unsigned int sampleRate, unsigned int channelCount);
////////////////////////////////////////////////////////////
/// \brief Close the file
///
////////////////////////////////////////////////////////////
void close();
////////////////////////////////////////////////////////////
// Member data
////////////////////////////////////////////////////////////
std::ofstream m_file; ///< File stream to write to
Uint64 m_sampleCount; ///< Total number of samples written to the file
unsigned int m_channelCount; ///< Number of channels of the sound
};
} // namespace priv
} // namespace sf
#endif // SFML_SOUNDFILEWRITERWAV_HPP

4
src/SFML/System/CMakeLists.txt
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@ -35,6 +35,10 @@ set(SRC
${INCROOT}/Vector2.inl
${INCROOT}/Vector3.hpp
${INCROOT}/Vector3.inl
${SRCROOT}/FileInputStream.cpp
${INCROOT}/FileInputStream.hpp
${SRCROOT}/MemoryInputStream.cpp
${INCROOT}/MemoryInputStream.hpp
)
source_group("" FILES ${SRC})

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////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/System/FileInputStream.hpp>
namespace sf
{
////////////////////////////////////////////////////////////
bool FileInputStream::open(const std::string& filename)
{
m_file.open(filename.c_str(), std::ios::binary);
return m_file.good();
}
////////////////////////////////////////////////////////////
Int64 FileInputStream::read(void* data, Int64 size)
{
m_file.read(static_cast<char*>(data), size);
return m_file.gcount();
}
////////////////////////////////////////////////////////////
Int64 FileInputStream::seek(Int64 position)
{
if (m_file.eof())
m_file.clear();
m_file.seekg(position);
return tell();
}
////////////////////////////////////////////////////////////
Int64 FileInputStream::tell()
{
return m_file.tellg();
}
////////////////////////////////////////////////////////////
Int64 FileInputStream::getSize()
{
std::ifstream::pos_type pos = m_file.tellg();
m_file.seekg(0, std::ios::end);
std::ifstream::pos_type size = m_file.tellg();
m_file.seekg(pos);
return size;
}
} // namespace sf

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////////////////////////////////////////////////////////////
//
// SFML - Simple and Fast Multimedia Library
// Copyright (C) 2007-2014 Laurent Gomila (laurent.gom@gmail.com)
//
// This software is provided 'as-is', without any express or implied warranty.
// In no event will the authors be held liable for any damages arising from the use of this software.
//
// Permission is granted to anyone to use this software for any purpose,
// including commercial applications, and to alter it and redistribute it freely,
// subject to the following restrictions:
//
// 1. The origin of this software must not be misrepresented;
// you must not claim that you wrote the original software.
// If you use this software in a product, an acknowledgment
// in the product documentation would be appreciated but is not required.
//
// 2. Altered source versions must be plainly marked as such,
// and must not be misrepresented as being the original software.
//
// 3. This notice may not be removed or altered from any source distribution.
//
////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////
// Headers
////////////////////////////////////////////////////////////
#include <SFML/System/MemoryInputStream.hpp>
#include <cstring>
namespace sf
{
////////////////////////////////////////////////////////////
MemoryInputStream::MemoryInputStream() :
m_data (NULL),
m_size (0),
m_offset(0)
{
}
////////////////////////////////////////////////////////////
void MemoryInputStream::open(const void* data, std::size_t sizeInBytes)
{
m_data = static_cast<const char*>(data);
m_size = sizeInBytes;
m_offset = 0;
}
////////////////////////////////////////////////////////////
Int64 MemoryInputStream::read(void* data, Int64 size)
{
if (!m_data)
return -1;
Int64 endPosition = m_offset + size;
Int64 count = endPosition <= m_size ? size : m_size - m_offset;
if (count > 0)
{
std::memcpy(data, m_data + m_offset, static_cast<std::size_t>(count));
m_offset += count;
}
return count;
}
////////////////////////////////////////////////////////////
Int64 MemoryInputStream::seek(Int64 position)
{
if (!m_data)
return -1;
m_offset = position < m_size ? position : m_size;
return m_offset;
}
////////////////////////////////////////////////////////////
Int64 MemoryInputStream::tell()
{
if (!m_data)
return -1;
return m_offset;
}
////////////////////////////////////////////////////////////
Int64 MemoryInputStream::getSize()
{
if (!m_data)
return -1;
return m_size;
}
} // namespace sf