diff --git a/build/codeblocks/SFML.workspace b/build/codeblocks/SFML.workspace index 56e3f261b..a55cc15eb 100644 --- a/build/codeblocks/SFML.workspace +++ b/build/codeblocks/SFML.workspace @@ -1,11 +1,11 @@ - + - + diff --git a/build/codeblocks/sfml-audio.cbp b/build/codeblocks/sfml-audio.cbp index f51722c33..05b989656 100644 --- a/build/codeblocks/sfml-audio.cbp +++ b/build/codeblocks/sfml-audio.cbp @@ -129,16 +129,8 @@ - - - - - - - diff --git a/build/vc2005/sfml-audio.vcproj b/build/vc2005/sfml-audio.vcproj index 857d80a4a..7e3bb55d6 100644 --- a/build/vc2005/sfml-audio.vcproj +++ b/build/vc2005/sfml-audio.vcproj @@ -343,52 +343,6 @@ - - - - - - - - - - - - - - - - - - @@ -457,22 +411,6 @@ RelativePath="..\..\src\SFML\Audio\SoundFile.hpp" > - - - - - - - - diff --git a/build/vc2008/sfml-audio.vcproj b/build/vc2008/sfml-audio.vcproj index ffad02ee5..970eb9a5a 100644 --- a/build/vc2008/sfml-audio.vcproj +++ b/build/vc2008/sfml-audio.vcproj @@ -339,52 +339,6 @@ - - - - - - - - - - - - - - - - - - @@ -453,22 +407,6 @@ RelativePath="..\..\src\SFML\Audio\SoundFile.hpp" > - - - - - - - - diff --git a/dotnet/extlibs/libsndfile-1.dll b/dotnet/extlibs/libsndfile-1.dll index f112de2bc..8d9d7ec36 100644 Binary files a/dotnet/extlibs/libsndfile-1.dll and b/dotnet/extlibs/libsndfile-1.dll differ diff --git a/extlibs/bin/libsndfile-1.dll b/extlibs/bin/libsndfile-1.dll index f112de2bc..8d9d7ec36 100644 Binary files a/extlibs/bin/libsndfile-1.dll and b/extlibs/bin/libsndfile-1.dll differ diff --git a/extlibs/headers/sndfile.h b/extlibs/headers/sndfile.h index 46eb18844..08e63d2f2 100644 --- a/extlibs/headers/sndfile.h +++ b/extlibs/headers/sndfile.h @@ -1,561 +1,617 @@ -/* -** Copyright (C) 1999-2006 Erik de Castro Lopo -** -** 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 -#ifdef __APPLE__ - #include -#endif - -/* For the Metrowerks CodeWarrior Pro Compiler (mainly MacOS) */ - -#if (defined (__MWERKS__)) -#include -#else -#include -#endif - -#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 */ - - /* 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) */ - - /* 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_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_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_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, - - /* 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 -} ; - -/* -** 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_DATE - -enum -{ /* True and false */ - SF_FALSE = 0, - SF_TRUE = 1, - - /* Modes for opening files. */ - SFM_READ = 0x10, - SFM_WRITE = 0x20, - SFM_RDWR = 0x30 -} ; - -/* 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 -} ; - -/* 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 can be one of loff_t (Linux), off_t (*BSD), -** off64_t (Solaris), __int64_t (Win32) etc. -*/ - -#ifdef __APPLE__ -typedef int64_t sf_count_t ; -#else -typedef __int64 sf_count_t ; -#endif - -#define SF_COUNT_MAX 0x7FFFFFFFFFFFFFFFLL - -/* A pointer to a SF_INFO structure is passed to sf_open_read () and filled in. -** On write, the SF_INFO structure is filled in by the user and passed into -** sf_open_write (). -*/ - -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. -*/ -typedef struct -{ char description [256] ; - char originator [32] ; - char originator_reference [32] ; - char origination_date [10] ; - char origination_time [8] ; - int time_reference_low ; - int time_reference_high ; - short version ; - char umid [64] ; - char reserved [190] ; - unsigned int coding_history_size ; - char coding_history [256] ; -} SF_BROADCAST_INFO ; - -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_perror () or sf_error_str (). -** 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_perror () or sf_error_str (). -** 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 three 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) ; - -/* 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) ; - -#ifdef __cplusplus -} /* extern "C" */ -#endif /* __cplusplus */ - -#endif /* SNDFILE_H */ +/* +** Copyright (C) 1999-2009 Erik de Castro Lopo +** +** 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 +#include + +#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 +} ; + +/* +** 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, + SF_CHANNEL_MAP_RIGHT, + SF_CHANNEL_MAP_CENTER, + SF_CHANNEL_MAP_FRONT_LEFT, + SF_CHANNEL_MAP_FRONT_RIGHT, + SF_CHANNEL_MAP_FRONT_CENTER, + SF_CHANNEL_MAP_REAR_CENTER, + SF_CHANNEL_MAP_REAR_LEFT, + SF_CHANNEL_MAP_REAR_RIGHT, + SF_CHANNEL_MAP_LFE, + SF_CHANNEL_MAP_FRONT_LEFT_OF_CENTER, + SF_CHANNEL_MAP_FRONT_RIGHT_OF_CENTER, + SF_CHANNEL_MAP_SIDE_LEFT, + SF_CHANNEL_MAP_SIDE_RIGHT, + SF_CHANNEL_MAP_TOP_CENTER, + SF_CHANNEL_MAP_TOP_FRONT_LEFT, + SF_CHANNEL_MAP_TOP_FRONT_RIGHT, + SF_CHANNEL_MAP_TOP_FRONT_CENTER, + SF_CHANNEL_MAP_TOP_REAR_LEFT, + SF_CHANNEL_MAP_TOP_REAR_RIGHT, + SF_CHANNEL_MAP_TOP_REAR_CENTER +} ; + + +/* 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 can be one of loff_t (Linux), off_t (*BSD), off64_t +** (Solaris), __int64_t (Win32) etc. On windows, we need to allow the same +** header file to be compiler by both GCC and the microsoft compiler. +*/ + +#ifdef _MSCVER +typedef __int64_t 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_read () and filled in. +** On write, the SF_INFO structure is filled in by the user and passed into +** sf_open_write (). +*/ + +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) ; + +/* 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) ; + +#ifdef __cplusplus +} /* extern "C" */ +#endif /* __cplusplus */ + +#endif /* SNDFILE_H */ diff --git a/extlibs/libs-mingw/libsndfile.a b/extlibs/libs-mingw/libsndfile.a index 1667abcd1..2604d2128 100644 Binary files a/extlibs/libs-mingw/libsndfile.a and b/extlibs/libs-mingw/libsndfile.a differ diff --git a/extlibs/libs-vc2005/sndfile.lib b/extlibs/libs-vc2005/sndfile.lib index be79d731a..947fc396d 100644 Binary files a/extlibs/libs-vc2005/sndfile.lib and b/extlibs/libs-vc2005/sndfile.lib differ diff --git a/samples/bin/libsndfile-1.dll b/samples/bin/libsndfile-1.dll index f112de2bc..8d9d7ec36 100644 Binary files a/samples/bin/libsndfile-1.dll and b/samples/bin/libsndfile-1.dll differ diff --git a/src/SFML/Audio/Makefile b/src/SFML/Audio/Makefile index 109a00ad4..c9825bb54 100644 --- a/src/SFML/Audio/Makefile +++ b/src/SFML/Audio/Makefile @@ -1,7 +1,5 @@ -SRC = $(wildcard *.cpp) -SRCVORBIS = $(wildcard ./stb_vorbis/*.c) -OBJ = $(SRC:.cpp=.o) -OBJVORBIS = $(SRCVORBIS:.c=.o) +SRC = $(wildcard *.cpp) +OBJ = $(SRC:.cpp=.o) ifeq ($(STATIC), yes) LIB = libsfml-audio-s.a @@ -15,22 +13,19 @@ endif all: $(LIB) -libsfml-audio-s.a: $(OBJ) $(OBJVORBIS) - $(AR) $(ARFLAGS) $(LIBNAME) $(OBJ) $(OBJVORBIS) +libsfml-audio-s.a: $(OBJ) + $(AR) $(ARFLAGS) $(LIBNAME) $(OBJ) libsfml-audio.so: $(OBJ) $(OBJVORBIS) - $(CPP) $(LDFLAGS) -Wl,-soname,$(LIB).$(VERSION) -o $(LIBNAME) $(OBJ) $(OBJVORBIS) -lsndfile -lopenal + $(CPP) $(LDFLAGS) -Wl,-soname,$(LIB).$(VERSION) -o $(LIBNAME) $(OBJ) -lsndfile -lopenal $(OBJ): %.o: %.cpp $(CPP) -o $@ -c $< $(CFLAGS) -$(OBJVORBIS): %.o: %.c - $(CC) -o $@ -c $< $(CFLAGSEXT) - .PHONY: clean mrproper clean: - @rm -rf $(OBJ) $(OBJVORBIS) + @rm -rf $(OBJ) mrproper: clean @rm -rf $(LIBNAME) diff --git a/src/SFML/Audio/Music.cpp b/src/SFML/Audio/Music.cpp index 5023ff50d..ef495d79a 100644 --- a/src/SFML/Audio/Music.cpp +++ b/src/SFML/Audio/Music.cpp @@ -38,7 +38,7 @@ namespace sf /// Construct the music with a buffer size //////////////////////////////////////////////////////////// Music::Music(std::size_t BufferSize) : -myFile (NULL), +myFile (new priv::SoundFile), myDuration(0.f), mySamples (BufferSize) { @@ -67,9 +67,7 @@ bool Music::OpenFromFile(const std::string& Filename) Stop(); // Create the sound file implementation, and open it in read mode - delete myFile; - myFile = priv::SoundFile::CreateRead(Filename); - if (!myFile) + if (!myFile->OpenRead(Filename)) { std::cerr << "Failed to open \"" << Filename << "\" for reading" << std::endl; return false; @@ -94,9 +92,7 @@ bool Music::OpenFromMemory(const char* Data, std::size_t SizeInBytes) Stop(); // Create the sound file implementation, and open it in read mode - delete myFile; - myFile = priv::SoundFile::CreateRead(Data, SizeInBytes); - if (!myFile) + if (!myFile->OpenRead(Data, SizeInBytes)) { std::cerr << "Failed to open music from memory for reading" << std::endl; return false; diff --git a/src/SFML/Audio/SoundBuffer.cpp b/src/SFML/Audio/SoundBuffer.cpp index 7636a01db..7294e0acb 100644 --- a/src/SFML/Audio/SoundBuffer.cpp +++ b/src/SFML/Audio/SoundBuffer.cpp @@ -80,37 +80,29 @@ SoundBuffer::~SoundBuffer() //////////////////////////////////////////////////////////// bool SoundBuffer::LoadFromFile(const std::string& Filename) { - // Create the sound file - std::auto_ptr File(priv::SoundFile::CreateRead(Filename)); - // Open the sound file - if (File.get()) + priv::SoundFile File; + if (File.OpenRead(Filename)) { // Get the sound parameters - std::size_t NbSamples = File->GetSamplesCount(); - unsigned int ChannelsCount = File->GetChannelsCount(); - unsigned int SampleRate = File->GetSampleRate(); + std::size_t NbSamples = File.GetSamplesCount(); + unsigned int ChannelsCount = File.GetChannelsCount(); + unsigned int SampleRate = File.GetSampleRate(); // Read the samples from the opened file mySamples.resize(NbSamples); - if (File->Read(&mySamples[0], NbSamples) == NbSamples) + if (File.Read(&mySamples[0], NbSamples) == NbSamples) { // Update the internal buffer with the new samples return Update(ChannelsCount, SampleRate); } else { - // Error... - std::cerr << "Failed to read audio data from file \"" << Filename << "\"" << std::endl; - return false; } } else { - // Error... - std::cerr << "Failed to load sound buffer from file \"" << Filename << "\"" << std::endl; - return false; } } @@ -121,37 +113,29 @@ bool SoundBuffer::LoadFromFile(const std::string& Filename) //////////////////////////////////////////////////////////// bool SoundBuffer::LoadFromMemory(const char* Data, std::size_t SizeInBytes) { - // Create the sound file - std::auto_ptr File(priv::SoundFile::CreateRead(Data, SizeInBytes)); - // Open the sound file - if (File.get()) + priv::SoundFile File; + if (File.OpenRead(Data, SizeInBytes)) { // Get the sound parameters - std::size_t NbSamples = File->GetSamplesCount(); - unsigned int ChannelsCount = File->GetChannelsCount(); - unsigned int SampleRate = File->GetSampleRate(); + std::size_t NbSamples = File.GetSamplesCount(); + unsigned int ChannelsCount = File.GetChannelsCount(); + unsigned int SampleRate = File.GetSampleRate(); // Read the samples from the opened file mySamples.resize(NbSamples); - if (File->Read(&mySamples[0], NbSamples) == NbSamples) + if (File.Read(&mySamples[0], NbSamples) == NbSamples) { // Update the internal buffer with the new samples return Update(ChannelsCount, SampleRate); } else { - // Error... - std::cerr << "Failed to read audio data from file in memory" << std::endl; - return false; } } else { - // Error... - std::cerr << "Failed to load sound buffer from file in memory" << std::endl; - return false; } } @@ -192,19 +176,16 @@ bool SoundBuffer::LoadFromSamples(const Int16* Samples, std::size_t SamplesCount bool SoundBuffer::SaveToFile(const std::string& Filename) const { // Create the sound file in write mode - std::auto_ptr File(priv::SoundFile::CreateWrite(Filename, GetChannelsCount(), GetSampleRate())); - if (File.get()) + priv::SoundFile File; + if (File.OpenWrite(Filename, GetChannelsCount(), GetSampleRate())) { // Write the samples to the opened file - File->Write(&mySamples[0], mySamples.size()); + File.Write(&mySamples[0], mySamples.size()); return true; } else { - // Error... - std::cerr << "Failed to save sound buffer to file \"" << Filename << "\"" << std::endl; - return false; } } diff --git a/src/SFML/Audio/SoundFile.cpp b/src/SFML/Audio/SoundFile.cpp index 0554d8b14..c90ba7c5f 100644 --- a/src/SFML/Audio/SoundFile.cpp +++ b/src/SFML/Audio/SoundFile.cpp @@ -26,126 +26,19 @@ // Headers //////////////////////////////////////////////////////////// #include -#include -#include #include +#include namespace sf { namespace priv { -//////////////////////////////////////////////////////////// -/// Create a new sound from a file, for reading -//////////////////////////////////////////////////////////// -SoundFile* SoundFile::CreateRead(const std::string& Filename) -{ - // Create the file according to its type - SoundFile* File = NULL; - if (SoundFileOgg::IsFileSupported(Filename, true)) File = new SoundFileOgg; - else if (SoundFileDefault::IsFileSupported(Filename, true)) File = new SoundFileDefault; - - // Open it for reading - if (File) - { - std::size_t SamplesCount; - unsigned int ChannelsCount; - unsigned int SampleRate; - - if (File->OpenRead(Filename, SamplesCount, ChannelsCount, SampleRate)) - { - File->myFilename = Filename; - File->myData = NULL; - File->mySize = 0; - File->myNbSamples = SamplesCount; - File->myChannelsCount = ChannelsCount; - File->mySampleRate = SampleRate; - } - else - { - delete File; - File = NULL; - } - } - - return File; -} - - -//////////////////////////////////////////////////////////// -/// Create a new sound from a file in memory, for reading -//////////////////////////////////////////////////////////// -SoundFile* SoundFile::CreateRead(const char* Data, std::size_t SizeInMemory) -{ - // Create the file according to its type - SoundFile* File = NULL; - if (SoundFileOgg::IsFileSupported(Data, SizeInMemory)) File = new SoundFileOgg; - else if (SoundFileDefault::IsFileSupported(Data, SizeInMemory)) File = new SoundFileDefault; - - // Open it for reading - if (File) - { - std::size_t SamplesCount; - unsigned int ChannelsCount; - unsigned int SampleRate; - - if (File->OpenRead(Data, SizeInMemory, SamplesCount, ChannelsCount, SampleRate)) - { - File->myFilename = ""; - File->myData = Data; - File->mySize = SizeInMemory; - File->myNbSamples = SamplesCount; - File->myChannelsCount = ChannelsCount; - File->mySampleRate = SampleRate; - } - else - { - delete File; - File = NULL; - } - } - - return File; -} - - -//////////////////////////////////////////////////////////// -/// Create a new sound from a file, for writing -//////////////////////////////////////////////////////////// -SoundFile* SoundFile::CreateWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate) -{ - // Create the file according to its type - SoundFile* File = NULL; - if (SoundFileOgg::IsFileSupported(Filename, false)) File = new SoundFileOgg; - else if (SoundFileDefault::IsFileSupported(Filename, false)) File = new SoundFileDefault; - - // Open it for writing - if (File) - { - if (File->OpenWrite(Filename, ChannelsCount, SampleRate)) - { - File->myFilename = ""; - File->myData = NULL; - File->mySize = 0; - File->myNbSamples = 0; - File->myChannelsCount = ChannelsCount; - File->mySampleRate = SampleRate; - } - else - { - delete File; - File = NULL; - } - } - - return File; -} - - //////////////////////////////////////////////////////////// /// Default constructor //////////////////////////////////////////////////////////// SoundFile::SoundFile() : +myFile (NULL), myNbSamples (0), myChannelsCount(0), mySampleRate (0) @@ -155,11 +48,12 @@ mySampleRate (0) //////////////////////////////////////////////////////////// -/// Virtual destructor +/// Destructor //////////////////////////////////////////////////////////// SoundFile::~SoundFile() { - // Nothing to do + if (myFile) + sf_close(myFile); } @@ -198,12 +92,12 @@ bool SoundFile::Restart() if (myData) { // Reopen from memory - return OpenRead(myData, mySize, myNbSamples, myChannelsCount, mySampleRate); + return OpenRead(myData, mySize); } else if (myFilename != "") { // Reopen from file - return OpenRead(myFilename, myNbSamples, myChannelsCount, mySampleRate); + return OpenRead(myFilename); } else { @@ -217,53 +111,257 @@ bool SoundFile::Restart() //////////////////////////////////////////////////////////// /// Open the sound file for reading //////////////////////////////////////////////////////////// -bool SoundFile::OpenRead(const std::string& Filename, std::size_t&, unsigned int&, unsigned int&) +bool SoundFile::OpenRead(const std::string& Filename) { - std::cerr << "Failed to open sound file \"" << Filename << "\", format is not supported by SFML" << std::endl; + // If the file is already opened, first close it + if (myFile) + sf_close(myFile); - return false; + // Open the sound file + SF_INFO FileInfos; + myFile = sf_open(Filename.c_str(), SFM_READ, &FileInfos); + if (!myFile) + { + std::cerr << "Failed to read sound file \"" << Filename << "\" (" << sf_strerror(myFile) << ")" << std::endl; + return false; + } + + // Set the sound parameters + myChannelsCount = FileInfos.channels; + mySampleRate = FileInfos.samplerate; + myNbSamples = static_cast(FileInfos.frames) * myChannelsCount; + myFilename = Filename; + myData = NULL; + mySize = 0; + + return true; } //////////////////////////////////////////////////////////// /// Open the sound file in memory for reading //////////////////////////////////////////////////////////// -bool SoundFile::OpenRead(const char*, std::size_t, std::size_t&, unsigned int&, unsigned int&) +bool SoundFile::OpenRead(const char* Data, std::size_t SizeInBytes) { - std::cerr << "Failed to open sound file from memory, format is not supported by SFML" << std::endl; + // If the file is already opened, first close it + if (myFile) + sf_close(myFile); - return false; + // Define the I/O custom functions for reading from memory + SF_VIRTUAL_IO VirtualIO; + VirtualIO.get_filelen = &SoundFile::MemoryGetLength; + VirtualIO.read = &SoundFile::MemoryRead; + VirtualIO.seek = &SoundFile::MemorySeek; + VirtualIO.tell = &SoundFile::MemoryTell; + VirtualIO.write = &SoundFile::MemoryWrite; + + // Initialize the memory data + myMemory.DataStart = Data; + myMemory.DataPtr = Data; + myMemory.TotalSize = SizeInBytes; + + // Open the sound file + SF_INFO FileInfos; + myFile = sf_open_virtual(&VirtualIO, SFM_READ, &FileInfos, &myMemory); + if (!myFile) + { + std::cerr << "Failed to read sound file from memory (" << sf_strerror(myFile) << ")" << std::endl; + return false; + } + + // Set the sound parameters + myChannelsCount = FileInfos.channels; + mySampleRate = FileInfos.samplerate; + myNbSamples = static_cast(FileInfos.frames) * myChannelsCount; + myFilename = ""; + myData = Data; + mySize = SizeInBytes; + + return true; } //////////////////////////////////////////////////////////// /// Open the sound file for writing //////////////////////////////////////////////////////////// -bool SoundFile::OpenWrite(const std::string& Filename, unsigned int, unsigned int) +bool SoundFile::OpenWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate) { - std::cerr << "Failed to open sound file \"" << Filename << "\", format is not supported by SFML" << std::endl; + // If the file is already opened, first close it + if (myFile) + sf_close(myFile); - return false; + // Find the right format according to the file extension + int Format = GetFormatFromFilename(Filename); + if (Format == -1) + { + // Error : unrecognized extension + std::cerr << "Failed to create sound file \"" << Filename << "\" (unknown format)" << std::endl; + return false; + } + + // Fill the sound infos with parameters + SF_INFO FileInfos; + FileInfos.channels = ChannelsCount; + FileInfos.samplerate = SampleRate; + FileInfos.format = Format | (Format == SF_FORMAT_OGG ? SF_FORMAT_VORBIS : SF_FORMAT_PCM_16); + + // Open the sound file for writing + myFile = sf_open(Filename.c_str(), SFM_WRITE, &FileInfos); + if (!myFile) + { + std::cerr << "Failed to create sound file \"" << Filename << "\" (" << sf_strerror(myFile) << ")" << std::endl; + return false; + } + + // Set the sound parameters + myChannelsCount = ChannelsCount; + mySampleRate = SampleRate; + myNbSamples = 0; + myFilename = ""; + myData = NULL; + mySize = 0; + + return true; } //////////////////////////////////////////////////////////// /// Read samples from the loaded sound //////////////////////////////////////////////////////////// -std::size_t SoundFile::Read(Int16*, std::size_t) +std::size_t SoundFile::Read(Int16* Data, std::size_t NbSamples) { - std::cerr << "Failed to read from sound file (not supported)" << std::endl; - - return 0; + if (myFile && Data && NbSamples) + return static_cast(sf_read_short(myFile, Data, NbSamples)); + else + return 0; } //////////////////////////////////////////////////////////// /// Write samples to the file //////////////////////////////////////////////////////////// -void SoundFile::Write(const Int16*, std::size_t) +void SoundFile::Write(const Int16* Data, std::size_t NbSamples) { - std::cerr << "Failed to write to sound file (not supported)" << std::endl; + if (myFile && Data && NbSamples) + { + // Write small chunks instead of everything at once, + // to avoid a stack overflow in libsndfile (happens only with OGG format) + while (NbSamples > 0) + { + std::size_t Count = NbSamples > 10000 ? 10000 : NbSamples; + sf_write_short(myFile, Data, Count); + Data += Count; + NbSamples -= Count; + } + } +} + + +//////////////////////////////////////////////////////////// +/// Get the internal format of an audio file according to +/// its filename extension +//////////////////////////////////////////////////////////// +int SoundFile::GetFormatFromFilename(const std::string& Filename) +{ + // Extract the extension + std::string Ext = "wav"; + std::string::size_type Pos = Filename.find_last_of("."); + if (Pos != std::string::npos) + Ext = Filename.substr(Pos + 1); + + // Match every supported extension with its format constant + if (Ext == "wav" || Ext == "WAV" ) return SF_FORMAT_WAV; + if (Ext == "aif" || Ext == "AIF" ) return SF_FORMAT_AIFF; + if (Ext == "aiff" || Ext == "AIFF") return SF_FORMAT_AIFF; + if (Ext == "au" || Ext == "AU" ) return SF_FORMAT_AU; + if (Ext == "raw" || Ext == "RAW" ) return SF_FORMAT_RAW; + if (Ext == "paf" || Ext == "PAF" ) return SF_FORMAT_PAF; + if (Ext == "svx" || Ext == "SVX" ) return SF_FORMAT_SVX; + if (Ext == "nist" || Ext == "NIST") return SF_FORMAT_NIST; + if (Ext == "voc" || Ext == "VOC" ) return SF_FORMAT_VOC; + if (Ext == "sf" || Ext == "SF" ) return SF_FORMAT_IRCAM; + if (Ext == "w64" || Ext == "W64" ) return SF_FORMAT_W64; + if (Ext == "mat4" || Ext == "MAT4") return SF_FORMAT_MAT4; + if (Ext == "mat5" || Ext == "MAT5") return SF_FORMAT_MAT5; + if (Ext == "pvf" || Ext == "PVF" ) return SF_FORMAT_PVF; + if (Ext == "xi" || Ext == "XI" ) return SF_FORMAT_XI; + if (Ext == "htk" || Ext == "HTK" ) return SF_FORMAT_HTK; + if (Ext == "sds" || Ext == "SDS" ) return SF_FORMAT_SDS; + if (Ext == "avr" || Ext == "AVR" ) return SF_FORMAT_AVR; + if (Ext == "sd2" || Ext == "SD2" ) return SF_FORMAT_SD2; + if (Ext == "flac" || Ext == "FLAC") return SF_FORMAT_FLAC; + if (Ext == "caf" || Ext == "CAF" ) return SF_FORMAT_CAF; + if (Ext == "wve" || Ext == "WVE" ) return SF_FORMAT_WVE; + if (Ext == "ogg" || Ext == "OGG") return SF_FORMAT_OGG; + if (Ext == "mpc2k" || Ext == "MPC2K") return SF_FORMAT_MPC2K; + if (Ext == "rf64" || Ext == "RF64") return SF_FORMAT_RF64; + + return -1; +} + + +//////////////////////////////////////////////////////////// +/// Functions for implementing custom read and write to memory files +//////////////////////////////////////////////////////////// +sf_count_t SoundFile::MemoryGetLength(void* UserData) +{ + MemoryInfos* Memory = static_cast(UserData); + + return Memory->TotalSize; +} +sf_count_t SoundFile::MemoryRead(void* Ptr, sf_count_t Count, void* UserData) +{ + MemoryInfos* Memory = static_cast(UserData); + + sf_count_t Position = Memory->DataPtr - Memory->DataStart; + if (Position + Count >= Memory->TotalSize) + Count = Memory->TotalSize - Position; + + memcpy(Ptr, Memory->DataPtr, static_cast(Count)); + + Memory->DataPtr += Count; + + return Count; +} +sf_count_t SoundFile::MemorySeek(sf_count_t Offset, int Whence, void* UserData) +{ + MemoryInfos* Memory = static_cast(UserData); + + sf_count_t Position = 0; + switch (Whence) + { + case SEEK_SET : + Position = Offset; + break; + case SEEK_CUR : + Position = Memory->DataPtr - Memory->DataStart + Offset; + break; + case SEEK_END : + Position = Memory->TotalSize - Offset; + break; + default : + Position = 0; + break; + } + + if (Position >= Memory->TotalSize) + Position = Memory->TotalSize - 1; + else if (Position < 0) + Position = 0; + + Memory->DataPtr = Memory->DataStart + Position; + + return Position; +} +sf_count_t SoundFile::MemoryTell(void* UserData) +{ + MemoryInfos* Memory = static_cast(UserData); + + return Memory->DataPtr - Memory->DataStart; +} +sf_count_t SoundFile::MemoryWrite(const void*, sf_count_t, void*) +{ + return 0; } } // namespace priv diff --git a/src/SFML/Audio/SoundFile.hpp b/src/SFML/Audio/SoundFile.hpp index fb36bdd32..81a2fd954 100644 --- a/src/SFML/Audio/SoundFile.hpp +++ b/src/SFML/Audio/SoundFile.hpp @@ -29,6 +29,7 @@ // Headers //////////////////////////////////////////////////////////// #include +#include #include @@ -37,57 +38,24 @@ namespace sf namespace priv { //////////////////////////////////////////////////////////// -/// SoundFile is the abstract base class for loading -/// and saving different sound file formats +/// SoundFile is used to load and save various sampled +/// sound file formats //////////////////////////////////////////////////////////// class SoundFile : NonCopyable { public : //////////////////////////////////////////////////////////// - /// Create a new sound from a file, for reading - /// - /// \param Filename : Path of sound file - /// \param NbSamples : Number of samples in the file - /// \param ChannelsCount : Number of channels in the loaded sound - /// \param SampleRate : Sample rate of the loaded sound - /// - /// \return Pointer to the new sound file (NULL if failed) + /// Default constructor /// //////////////////////////////////////////////////////////// - static SoundFile* CreateRead(const std::string& Filename); + SoundFile(); //////////////////////////////////////////////////////////// - /// Create a new sound from a file in memory, for reading - /// - /// \param Data : Pointer to the file data in memory - /// \param SizeInBytes : Size of the data to load, in bytes - /// \param NbSamples : Number of samples in the file - /// \param ChannelsCount : Number of channels in the loaded sound - /// \param SampleRate : Sample rate of the loaded sound - /// - /// \return Pointer to the new sound file (NULL if failed) + /// Destructor /// //////////////////////////////////////////////////////////// - static SoundFile* CreateRead(const char* Data, std::size_t SizeInBytes); - - //////////////////////////////////////////////////////////// - /// Create a new sound from a file, for writing - /// - /// \param Filename : Path of sound file - /// \param ChannelsCount : Number of channels in the sound - /// \param SampleRate : Sample rate of the sound - /// - /// \return Pointer to the new sound file (NULL if failed) - /// - //////////////////////////////////////////////////////////// - static SoundFile* CreateWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate); - - //////////////////////////////////////////////////////////// - /// Virtual destructor - /// - //////////////////////////////////////////////////////////// - virtual ~SoundFile(); + ~SoundFile(); //////////////////////////////////////////////////////////// /// Get the total number of samples in the file @@ -121,62 +89,26 @@ public : //////////////////////////////////////////////////////////// bool Restart(); - //////////////////////////////////////////////////////////// - /// Read samples from the loaded sound - /// - /// \param Data : Pointer to the samples array to fill - /// \param NbSamples : Number of samples to read - /// - /// \return Number of samples read - /// - //////////////////////////////////////////////////////////// - virtual std::size_t Read(Int16* Data, std::size_t NbSamples); - - //////////////////////////////////////////////////////////// - /// Write samples to the file - /// - /// \param Data : Pointer to the samples array to write - /// \param NbSamples : Number of samples to write - /// - //////////////////////////////////////////////////////////// - virtual void Write(const Int16* Data, std::size_t NbSamples); - -protected : - - //////////////////////////////////////////////////////////// - /// Default constructor - /// - //////////////////////////////////////////////////////////// - SoundFile(); - -private : - //////////////////////////////////////////////////////////// /// Open the sound file for reading /// - /// \param Filename : Path of sound file to load - /// \param NbSamples : Number of samples in the file - /// \param ChannelsCount : Number of channels in the loaded sound - /// \param SampleRate : Sample rate of the loaded sound + /// \param Filename : Path of sound file to load /// /// \return True if the file was successfully opened /// //////////////////////////////////////////////////////////// - virtual bool OpenRead(const std::string& Filename, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate); + bool OpenRead(const std::string& Filename); //////////////////////////////////////////////////////////// /// Open the sound file in memory for reading /// - /// \param Data : Pointer to the file data in memory - /// \param SizeInBytes : Size of the data to load, in bytes - /// \param NbSamples : Number of samples in the file - /// \param ChannelsCount : Number of channels in the loaded sound - /// \param SampleRate : Sample rate of the loaded sound + /// \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 /// //////////////////////////////////////////////////////////// - virtual bool OpenRead(const char* Data, std::size_t SizeInBytes, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate); + bool OpenRead(const char* Data, std::size_t SizeInBytes); //////////////////////////////////////////////////////////// /// Open the sound file for writing @@ -188,11 +120,66 @@ private : /// \return True if the file was successfully opened /// //////////////////////////////////////////////////////////// - virtual bool OpenWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate); + bool OpenWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate); + + //////////////////////////////////////////////////////////// + /// Read samples from the loaded sound + /// + /// \param Data : Pointer to the samples array to fill + /// \param NbSamples : Number of samples to read + /// + /// \return Number of samples read + /// + //////////////////////////////////////////////////////////// + std::size_t Read(Int16* Data, std::size_t NbSamples); + + //////////////////////////////////////////////////////////// + /// Write samples to the file + /// + /// \param Data : Pointer to the samples array to write + /// \param NbSamples : Number of samples to write + /// + //////////////////////////////////////////////////////////// + void Write(const Int16* Data, std::size_t NbSamples); + +private : + + //////////////////////////////////////////////////////////// + /// 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); + + //////////////////////////////////////////////////////////// + /// Functions for implementing custom read and write to memory files + /// + //////////////////////////////////////////////////////////// + static sf_count_t MemoryGetLength(void* UserData); + static sf_count_t MemoryRead(void* Ptr, sf_count_t Count, void* UserData); + static sf_count_t MemorySeek(sf_count_t Offset, int Whence, void* UserData); + static sf_count_t MemoryTell(void* UserData); + static sf_count_t MemoryWrite(const void* Ptr, sf_count_t Count, void* UserData); + + //////////////////////////////////////////////////////////// + /// Structure holding data related to memory operations + //////////////////////////////////////////////////////////// + struct MemoryInfos + { + const char* DataStart; ///< Pointer to the begining of the data + const char* DataPtr; ///< Pointer to the current read / write position + sf_count_t TotalSize; ///< Total size of the data, in bytes + }; //////////////////////////////////////////////////////////// // Member data //////////////////////////////////////////////////////////// + SNDFILE* myFile; ///< File descriptor + MemoryInfos myMemory; ///< Memory read / write data std::size_t myNbSamples; ///< Total number of samples in the file unsigned int myChannelsCount; ///< Number of channels used by the sound unsigned int mySampleRate; ///< Number of samples per second diff --git a/src/SFML/Audio/SoundFileDefault.cpp b/src/SFML/Audio/SoundFileDefault.cpp deleted file mode 100644 index 4413f6bbd..000000000 --- a/src/SFML/Audio/SoundFileDefault.cpp +++ /dev/null @@ -1,352 +0,0 @@ -//////////////////////////////////////////////////////////// -// -// SFML - Simple and Fast Multimedia Library -// Copyright (C) 2007-2009 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 -#include -#include - - -namespace sf -{ -namespace priv -{ -//////////////////////////////////////////////////////////// -/// Default constructor -//////////////////////////////////////////////////////////// -SoundFileDefault::SoundFileDefault() : -myFile(NULL) -{ - -} - - -//////////////////////////////////////////////////////////// -/// Destructor -//////////////////////////////////////////////////////////// -SoundFileDefault::~SoundFileDefault() -{ - if (myFile) - sf_close(myFile); -} - - -//////////////////////////////////////////////////////////// -/// Check if a given file is supported by this loader -//////////////////////////////////////////////////////////// -bool SoundFileDefault::IsFileSupported(const std::string& Filename, bool Read) -{ - if (Read) - { - // Open the sound file - SF_INFO FileInfos; - SNDFILE* File = sf_open(Filename.c_str(), SFM_READ, &FileInfos); - - if (File) - { - sf_close(File); - return true; - } - else - { - return false; - } - } - else - { - // Check the extension - return GetFormatFromFilename(Filename) != -1; - } -} - - -//////////////////////////////////////////////////////////// -/// Check if a given file in memory is supported by this loader -//////////////////////////////////////////////////////////// -bool SoundFileDefault::IsFileSupported(const char* Data, std::size_t SizeInBytes) -{ - // Define the I/O custom functions for reading from memory - SF_VIRTUAL_IO VirtualIO; - VirtualIO.get_filelen = &SoundFileDefault::MemoryGetLength; - VirtualIO.read = &SoundFileDefault::MemoryRead; - VirtualIO.seek = &SoundFileDefault::MemorySeek; - VirtualIO.tell = &SoundFileDefault::MemoryTell; - VirtualIO.write = &SoundFileDefault::MemoryWrite; - - // Initialize the memory data - MemoryInfos Memory; - Memory.DataStart = Data; - Memory.DataPtr = Data; - Memory.TotalSize = SizeInBytes; - - // Open the sound file - SF_INFO FileInfos; - SNDFILE* File = sf_open_virtual(&VirtualIO, SFM_READ, &FileInfos, &Memory); - - if (File) - { - sf_close(File); - return true; - } - else - { - return false; - } -} - - -//////////////////////////////////////////////////////////// -/// Open the sound file for reading -//////////////////////////////////////////////////////////// -bool SoundFileDefault::OpenRead(const std::string& Filename, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate) -{ - // If the file is already opened, first close it - if (myFile) - sf_close(myFile); - - // Open the sound file - SF_INFO FileInfos; - myFile = sf_open(Filename.c_str(), SFM_READ, &FileInfos); - if (!myFile) - { - std::cerr << "Failed to read sound file \"" << Filename << "\"" << std::endl; - return false; - } - - // Set the sound parameters - ChannelsCount = FileInfos.channels; - SampleRate = FileInfos.samplerate; - NbSamples = static_cast(FileInfos.frames) * ChannelsCount; - - return true; -} - - -//////////////////////////////////////////////////////////// -/// /see sf::SoundFile::OpenRead -//////////////////////////////////////////////////////////// -bool SoundFileDefault::OpenRead(const char* Data, std::size_t SizeInBytes, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate) -{ - // If the file is already opened, first close it - if (myFile) - sf_close(myFile); - - // Define the I/O custom functions for reading from memory - SF_VIRTUAL_IO VirtualIO; - VirtualIO.get_filelen = &SoundFileDefault::MemoryGetLength; - VirtualIO.read = &SoundFileDefault::MemoryRead; - VirtualIO.seek = &SoundFileDefault::MemorySeek; - VirtualIO.tell = &SoundFileDefault::MemoryTell; - VirtualIO.write = &SoundFileDefault::MemoryWrite; - - // Initialize the memory data - myMemory.DataStart = Data; - myMemory.DataPtr = Data; - myMemory.TotalSize = SizeInBytes; - - // Open the sound file - SF_INFO FileInfos; - myFile = sf_open_virtual(&VirtualIO, SFM_READ, &FileInfos, &myMemory); - if (!myFile) - { - std::cerr << "Failed to read sound file from memory" << std::endl; - return false; - } - - // Set the sound parameters - ChannelsCount = FileInfos.channels; - SampleRate = FileInfos.samplerate; - NbSamples = static_cast(FileInfos.frames) * ChannelsCount; - - return true; -} - - -//////////////////////////////////////////////////////////// -/// Open the sound file for writing -//////////////////////////////////////////////////////////// -bool SoundFileDefault::OpenWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate) -{ - // If the file is already opened, first close it - if (myFile) - sf_close(myFile); - - // Find the right format according to the file extension - int Format = GetFormatFromFilename(Filename); - if (Format == -1) - { - // Error : unrecognized extension - std::cerr << "Failed to create sound file \"" << Filename << "\" : unknown format" << std::endl; - return false; - } - - // Fill the sound infos with parameters - SF_INFO FileInfos; - FileInfos.channels = ChannelsCount; - FileInfos.samplerate = SampleRate; - FileInfos.format = Format | SF_FORMAT_PCM_16; - - // Open the sound file for writing - myFile = sf_open(Filename.c_str(), SFM_WRITE, &FileInfos); - if (!myFile) - { - std::cerr << "Failed to create sound file \"" << Filename << "\"" << std::endl; - return false; - } - - return true; -} - - -//////////////////////////////////////////////////////////// -/// Read samples from the loaded sound -//////////////////////////////////////////////////////////// -std::size_t SoundFileDefault::Read(Int16* Data, std::size_t NbSamples) -{ - if (myFile && Data && NbSamples) - return static_cast(sf_read_short(myFile, Data, NbSamples)); - else - return 0; -} - - -//////////////////////////////////////////////////////////// -/// Write samples to the file -//////////////////////////////////////////////////////////// -void SoundFileDefault::Write(const Int16* Data, std::size_t NbSamples) -{ - if (myFile && Data && NbSamples) - sf_write_short(myFile, Data, NbSamples); -} - - -//////////////////////////////////////////////////////////// -/// Get the internal format of an audio file according to -/// its filename extension -//////////////////////////////////////////////////////////// -int SoundFileDefault::GetFormatFromFilename(const std::string& Filename) -{ - // Extract the extension - std::string Ext = "wav"; - std::string::size_type Pos = Filename.find_last_of("."); - if (Pos != std::string::npos) - Ext = Filename.substr(Pos + 1); - - // Match every supported extension with its format constant - if (Ext == "wav" || Ext == "WAV" ) return SF_FORMAT_WAV; - if (Ext == "aif" || Ext == "AIF" ) return SF_FORMAT_AIFF; - if (Ext == "aiff" || Ext == "AIFF") return SF_FORMAT_AIFF; - if (Ext == "au" || Ext == "AU" ) return SF_FORMAT_AU; - if (Ext == "raw" || Ext == "RAW" ) return SF_FORMAT_RAW; - if (Ext == "paf" || Ext == "PAF" ) return SF_FORMAT_PAF; - if (Ext == "svx" || Ext == "SVX" ) return SF_FORMAT_SVX; - if (Ext == "voc" || Ext == "VOC" ) return SF_FORMAT_VOC; - if (Ext == "sf" || Ext == "SF" ) return SF_FORMAT_IRCAM; - if (Ext == "w64" || Ext == "W64" ) return SF_FORMAT_W64; - if (Ext == "mat4" || Ext == "MAT4") return SF_FORMAT_MAT4; - if (Ext == "mat5" || Ext == "MAT5") return SF_FORMAT_MAT5; - if (Ext == "pvf" || Ext == "PVF" ) return SF_FORMAT_PVF; - if (Ext == "htk" || Ext == "HTK" ) return SF_FORMAT_HTK; - if (Ext == "caf" || Ext == "CAF" ) return SF_FORMAT_CAF; - if (Ext == "nist" || Ext == "NIST") return SF_FORMAT_NIST; // SUPPORTED ? - if (Ext == "sds" || Ext == "SDS" ) return SF_FORMAT_SDS; // SUPPORTED ? - if (Ext == "avr" || Ext == "AVR" ) return SF_FORMAT_AVR; // SUPPORTED ? - if (Ext == "sd2" || Ext == "SD2" ) return SF_FORMAT_SD2; // SUPPORTED ? - if (Ext == "flac" || Ext == "FLAC") return SF_FORMAT_FLAC; // SUPPORTED ? - - return -1; -} - - -//////////////////////////////////////////////////////////// -/// Functions for implementing custom read and write to memory files -/// -//////////////////////////////////////////////////////////// -sf_count_t SoundFileDefault::MemoryGetLength(void* UserData) -{ - MemoryInfos* Memory = static_cast(UserData); - - return Memory->TotalSize; -} -sf_count_t SoundFileDefault::MemoryRead(void* Ptr, sf_count_t Count, void* UserData) -{ - MemoryInfos* Memory = static_cast(UserData); - - sf_count_t Position = Memory->DataPtr - Memory->DataStart; - if (Position + Count >= Memory->TotalSize) - Count = Memory->TotalSize - Position; - - memcpy(Ptr, Memory->DataPtr, static_cast(Count)); - - Memory->DataPtr += Count; - - return Count; -} -sf_count_t SoundFileDefault::MemorySeek(sf_count_t Offset, int Whence, void* UserData) -{ - MemoryInfos* Memory = static_cast(UserData); - - sf_count_t Position = 0; - switch (Whence) - { - case SEEK_SET : - Position = Offset; - break; - case SEEK_CUR : - Position = Memory->DataPtr - Memory->DataStart + Offset; - break; - case SEEK_END : - Position = Memory->TotalSize - Offset; - break; - default : - Position = 0; - break; - } - - if (Position >= Memory->TotalSize) - Position = Memory->TotalSize - 1; - else if (Position < 0) - Position = 0; - - Memory->DataPtr = Memory->DataStart + Position; - - return Position; -} -sf_count_t SoundFileDefault::MemoryTell(void* UserData) -{ - MemoryInfos* Memory = static_cast(UserData); - - return Memory->DataPtr - Memory->DataStart; -} -sf_count_t SoundFileDefault::MemoryWrite(const void*, sf_count_t, void*) -{ - return 0; -} - - -} // namespace priv - -} // namespace sf diff --git a/src/SFML/Audio/SoundFileDefault.hpp b/src/SFML/Audio/SoundFileDefault.hpp deleted file mode 100644 index 8317d5492..000000000 --- a/src/SFML/Audio/SoundFileDefault.hpp +++ /dev/null @@ -1,156 +0,0 @@ -//////////////////////////////////////////////////////////// -// -// SFML - Simple and Fast Multimedia Library -// Copyright (C) 2007-2009 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_SOUNDFILEDEFAULT_HPP -#define SFML_SOUNDFILEDEFAULT_HPP - -//////////////////////////////////////////////////////////// -// Headers -//////////////////////////////////////////////////////////// -#include -#include - - -namespace sf -{ -namespace priv -{ -//////////////////////////////////////////////////////////// -/// Specialization of SoundFile that can handle a lot of -/// sound formats (see libsndfile homepage for a complete list) -//////////////////////////////////////////////////////////// -class SoundFileDefault : public SoundFile -{ -public : - - //////////////////////////////////////////////////////////// - /// Default constructor - /// - //////////////////////////////////////////////////////////// - SoundFileDefault(); - - //////////////////////////////////////////////////////////// - /// Destructor - /// - //////////////////////////////////////////////////////////// - ~SoundFileDefault(); - - //////////////////////////////////////////////////////////// - /// Check if a given file is supported by this loader - /// - /// \param Filename : Path of the file to check - /// \param Read : Is the file opened for reading or writing ? - /// - /// \param return True if the loader can handle this file - /// - //////////////////////////////////////////////////////////// - static bool IsFileSupported(const std::string& Filename, bool Read); - - //////////////////////////////////////////////////////////// - /// Check if a given file in memory is supported by this loader - /// - /// \param Data : Pointer to the file data in memory - /// \param SizeInBytes : Size of the data to load, in bytes - /// - /// \param return True if the loader can handle this file - /// - //////////////////////////////////////////////////////////// - static bool IsFileSupported(const char* Data, std::size_t SizeInBytes); - - //////////////////////////////////////////////////////////// - /// /see sf::SoundFile::Read - /// - //////////////////////////////////////////////////////////// - virtual std::size_t Read(Int16* Data, std::size_t NbSamples); - - //////////////////////////////////////////////////////////// - /// /see sf::SoundFile::Write - /// - //////////////////////////////////////////////////////////// - virtual void Write(const Int16* Data, std::size_t NbSamples); - -private : - - //////////////////////////////////////////////////////////// - /// /see sf::SoundFile::OpenRead - /// - //////////////////////////////////////////////////////////// - virtual bool OpenRead(const std::string& Filename, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate); - - //////////////////////////////////////////////////////////// - /// /see sf::SoundFile::OpenRead - /// - //////////////////////////////////////////////////////////// - virtual bool OpenRead(const char* Data, std::size_t SizeInBytes, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate); - - //////////////////////////////////////////////////////////// - /// /see sf::SoundFile::OpenWrite - /// - //////////////////////////////////////////////////////////// - virtual bool OpenWrite(const std::string& Filename, unsigned int ChannelsCount, unsigned int SampleRate); - - //////////////////////////////////////////////////////////// - /// 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); - - //////////////////////////////////////////////////////////// - /// Functions for implementing custom read and write to memory files - /// - //////////////////////////////////////////////////////////// - static sf_count_t MemoryGetLength(void* UserData); - static sf_count_t MemoryRead(void* Ptr, sf_count_t Count, void* UserData); - static sf_count_t MemorySeek(sf_count_t Offset, int Whence, void* UserData); - static sf_count_t MemoryTell(void* UserData); - static sf_count_t MemoryWrite(const void* Ptr, sf_count_t Count, void* UserData); - - //////////////////////////////////////////////////////////// - /// Structure holding data related to memory operations - //////////////////////////////////////////////////////////// - struct MemoryInfos - { - const char* DataStart; ///< Pointer to the begining of the data - const char* DataPtr; ///< Pointer to the current read / write position - sf_count_t TotalSize; ///< Total size of the data, in bytes - }; - - //////////////////////////////////////////////////////////// - // Member data - //////////////////////////////////////////////////////////// - SNDFILE* myFile; ///< File descriptor - MemoryInfos myMemory; ///< Memory read / write data -}; - -} // namespace priv - -} // namespace sf - - -#endif // SFML_SOUNDFILEDEFAULT_HPP diff --git a/src/SFML/Audio/SoundFileOgg.cpp b/src/SFML/Audio/SoundFileOgg.cpp deleted file mode 100644 index 8aa94f8ab..000000000 --- a/src/SFML/Audio/SoundFileOgg.cpp +++ /dev/null @@ -1,182 +0,0 @@ -//////////////////////////////////////////////////////////// -// -// SFML - Simple and Fast Multimedia Library -// Copyright (C) 2007-2009 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 -#include -#include - - -namespace sf -{ -namespace priv -{ -//////////////////////////////////////////////////////////// -/// Default constructor -//////////////////////////////////////////////////////////// -SoundFileOgg::SoundFileOgg() : -myStream (NULL), -myChannelsCount(0) -{ - -} - - -//////////////////////////////////////////////////////////// -/// Destructor -//////////////////////////////////////////////////////////// -SoundFileOgg::~SoundFileOgg() -{ - if (myStream) - stb_vorbis_close(myStream); -} - - -//////////////////////////////////////////////////////////// -/// Check if a given file is supported by this loader -//////////////////////////////////////////////////////////// -bool SoundFileOgg::IsFileSupported(const std::string& Filename, bool Read) -{ - if (Read) - { - // Open the vorbis stream - stb_vorbis* Stream = stb_vorbis_open_filename(const_cast(Filename.c_str()), NULL, NULL); - - if (Stream) - { - stb_vorbis_close(Stream); - return true; - } - else - { - return false; - } - } - else - { - // No support for writing ogg files yet... - return false; - } -} - - -//////////////////////////////////////////////////////////// -/// Check if a given file in memory is supported by this loader -//////////////////////////////////////////////////////////// -bool SoundFileOgg::IsFileSupported(const char* Data, std::size_t SizeInBytes) -{ - // Open the vorbis stream - unsigned char* Buffer = reinterpret_cast(const_cast(Data)); - int Length = static_cast(SizeInBytes); - stb_vorbis* Stream = stb_vorbis_open_memory(Buffer, Length, NULL, NULL); - - if (Stream) - { - stb_vorbis_close(Stream); - return true; - } - else - { - return false; - } -} - - -//////////////////////////////////////////////////////////// -/// Open the sound file for reading -//////////////////////////////////////////////////////////// -bool SoundFileOgg::OpenRead(const std::string& Filename, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate) -{ - // Close the file if already opened - if (myStream) - stb_vorbis_close(myStream); - - // Open the vorbis stream - myStream = stb_vorbis_open_filename(const_cast(Filename.c_str()), NULL, NULL); - if (myStream == NULL) - { - std::cerr << "Failed to read sound file \"" << Filename << "\" (cannot open the file)" << std::endl; - return false; - } - - // Get the music parameters - stb_vorbis_info Infos = stb_vorbis_get_info(myStream); - ChannelsCount = myChannelsCount = Infos.channels; - SampleRate = Infos.sample_rate; - NbSamples = static_cast(stb_vorbis_stream_length_in_samples(myStream) * ChannelsCount); - - return true; -} - - -//////////////////////////////////////////////////////////// -/// /see sf::SoundFile::OpenRead -//////////////////////////////////////////////////////////// -bool SoundFileOgg::OpenRead(const char* Data, std::size_t SizeInBytes, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate) -{ - // Close the file if already opened - if (myStream) - stb_vorbis_close(myStream); - - // Open the vorbis stream - unsigned char* Buffer = reinterpret_cast(const_cast(Data)); - int Length = static_cast(SizeInBytes); - myStream = stb_vorbis_open_memory(Buffer, Length, NULL, NULL); - if (myStream == NULL) - { - std::cerr << "Failed to read sound file from memory (cannot open the file)" << std::endl; - return false; - } - - // Get the music parameters - stb_vorbis_info Infos = stb_vorbis_get_info(myStream); - ChannelsCount = myChannelsCount = Infos.channels; - SampleRate = Infos.sample_rate; - NbSamples = static_cast(stb_vorbis_stream_length_in_samples(myStream) * ChannelsCount); - - return true; -} - - -//////////////////////////////////////////////////////////// -/// Read samples from the loaded sound -//////////////////////////////////////////////////////////// -std::size_t SoundFileOgg::Read(Int16* Data, std::size_t NbSamples) -{ - if (myStream && Data && NbSamples) - { - int Read = stb_vorbis_get_samples_short_interleaved(myStream, myChannelsCount, Data, static_cast(NbSamples)); - return static_cast(Read * myChannelsCount); - } - else - { - return 0; - } -} - -} // namespace priv - -} // namespace sf diff --git a/src/SFML/Audio/SoundFileOgg.hpp b/src/SFML/Audio/SoundFileOgg.hpp deleted file mode 100644 index 98f47997e..000000000 --- a/src/SFML/Audio/SoundFileOgg.hpp +++ /dev/null @@ -1,114 +0,0 @@ -//////////////////////////////////////////////////////////// -// -// SFML - Simple and Fast Multimedia Library -// Copyright (C) 2007-2009 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_SOUNDFILEOGG_HPP -#define SFML_SOUNDFILEOGG_HPP - -//////////////////////////////////////////////////////////// -// Headers -//////////////////////////////////////////////////////////// -#include - -struct stb_vorbis; - - -namespace sf -{ -namespace priv -{ -//////////////////////////////////////////////////////////// -/// Specialization of SoundFile that handles ogg-vorbis files (.ogg) -/// (does not support variable bitrate / channels and writing) -//////////////////////////////////////////////////////////// -class SoundFileOgg : public SoundFile -{ -public : - - //////////////////////////////////////////////////////////// - /// Default constructor - /// - //////////////////////////////////////////////////////////// - SoundFileOgg(); - - //////////////////////////////////////////////////////////// - /// Destructor - /// - //////////////////////////////////////////////////////////// - ~SoundFileOgg(); - - //////////////////////////////////////////////////////////// - /// Check if a given file is supported by this loader - /// - /// \param Filename : Path of the file to check - /// \param Read : Is the file opened for reading or writing ? - /// - /// \param return True if the loader can handle this file - /// - //////////////////////////////////////////////////////////// - static bool IsFileSupported(const std::string& Filename, bool Read); - - //////////////////////////////////////////////////////////// - /// Check if a given file in memory is supported by this loader - /// - /// \param Data : Pointer to the file data in memory - /// \param SizeInBytes : Size of the data to load, in bytes - /// - /// \param return True if the loader can handle this file - /// - //////////////////////////////////////////////////////////// - static bool IsFileSupported(const char* Data, std::size_t SizeInBytes); - - //////////////////////////////////////////////////////////// - /// /see sf::SoundFile::Read - /// - //////////////////////////////////////////////////////////// - virtual std::size_t Read(Int16* Data, std::size_t NbSamples); - -private : - - //////////////////////////////////////////////////////////// - /// /see sf::SoundFile::OpenRead - /// - //////////////////////////////////////////////////////////// - virtual bool OpenRead(const std::string& Filename, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate); - - //////////////////////////////////////////////////////////// - /// /see sf::SoundFile::OpenRead - /// - //////////////////////////////////////////////////////////// - virtual bool OpenRead(const char* Data, std::size_t SizeInBytes, std::size_t& NbSamples, unsigned int& ChannelsCount, unsigned int& SampleRate); - - //////////////////////////////////////////////////////////// - // Member data - //////////////////////////////////////////////////////////// - stb_vorbis* myStream; ///< Vorbis stream - unsigned int myChannelsCount; ///< Number of channels (1 = mono, 2 = stereo) -}; - -} // namespace priv - -} // namespace sf - - -#endif // SFML_SOUNDFILEOGG_HPP diff --git a/src/SFML/Audio/stb_vorbis/stb_vorbis.c b/src/SFML/Audio/stb_vorbis/stb_vorbis.c deleted file mode 100755 index e34405f81..000000000 --- a/src/SFML/Audio/stb_vorbis/stb_vorbis.c +++ /dev/null @@ -1,5039 +0,0 @@ -// Ogg Vorbis I audio decoder -- version 0.99994 -// -// Written in April 2007 by Sean Barrett, sponsored by RAD Game Tools. -// -// Placed in the public domain April 2007 by the author: no copyright is -// claimed, and you may use it for any purpose you like. -// -// No warranty for any purpose is expressed or implied by the author (nor -// by RAD Game Tools). Report bugs and send enhancements to the author. -// -// Get the latest version and other information at: -// http://nothings.org/stb_vorbis/ - - -// Todo: -// -// - seeking (note you can seek yourself using the pushdata API) -// -// Limitations: -// -// - floor 0 not supported (used in old ogg vorbis files) -// - lossless sample-truncation at beginning ignored -// - cannot concatenate multiple vorbis streams -// - sample positions are 32-bit, limiting seekable 192Khz -// files to around 6 hours (Ogg supports 64-bit) -// -// All of these limitations may be removed in future versions. - -#include "stb_vorbis.h" - -#ifndef STB_VORBIS_HEADER_ONLY - -// global configuration settings (e.g. set these in the project/makefile), -// or just set them in this file at the top (although ideally the first few -// should be visible when the header file is compiled too, although it's not -// crucial) - -// STB_VORBIS_NO_PUSHDATA_API -// does not compile the code for the various stb_vorbis_*_pushdata() -// functions -// #define STB_VORBIS_NO_PUSHDATA_API - -// STB_VORBIS_NO_PULLDATA_API -// does not compile the code for the non-pushdata APIs -// #define STB_VORBIS_NO_PULLDATA_API - -// STB_VORBIS_NO_STDIO -// does not compile the code for the APIs that use FILE *s internally -// or externally (implied by STB_VORBIS_NO_PULLDATA_API) -// #define STB_VORBIS_NO_STDIO - -// STB_VORBIS_NO_INTEGER_CONVERSION -// does not compile the code for converting audio sample data from -// float to integer (implied by STB_VORBIS_NO_PULLDATA_API) -// #define STB_VORBIS_NO_INTEGER_CONVERSION - -// STB_VORBIS_NO_FAST_SCALED_FLOAT -// does not use a fast float-to-int trick to accelerate float-to-int on -// most platforms which requires endianness be defined correctly. -#define STB_VORBIS_NO_FAST_SCALED_FLOAT - - -// STB_VORBIS_MAX_CHANNELS [number] -// globally define this to the maximum number of channels you need. -// The spec does not put a restriction on channels except that -// the count is stored in a byte, so 255 is the hard limit. -// Reducing this saves about 16 bytes per value, so using 16 saves -// (255-16)*16 or around 4KB. Plus anything other memory usage -// I forgot to account for. Can probably go as low as 8 (7.1 audio), -// 6 (5.1 audio), or 2 (stereo only). -#ifndef STB_VORBIS_MAX_CHANNELS -#define STB_VORBIS_MAX_CHANNELS 16 // enough for anyone? -#endif - -// STB_VORBIS_PUSHDATA_CRC_COUNT [number] -// after a flush_pushdata(), stb_vorbis begins scanning for the -// next valid page, without backtracking. when it finds something -// that looks like a page, it streams through it and verifies its -// CRC32. Should that validation fail, it keeps scanning. But it's -// possible that _while_ streaming through to check the CRC32 of -// one candidate page, it sees another candidate page. This #define -// determines how many "overlapping" candidate pages it can search -// at once. Note that "real" pages are typically ~4KB to ~8KB, whereas -// garbage pages could be as big as 64KB, but probably average ~16KB. -// So don't hose ourselves by scanning an apparent 64KB page and -// missing a ton of real ones in the interim; so minimum of 2 -#ifndef STB_VORBIS_PUSHDATA_CRC_COUNT -#define STB_VORBIS_PUSHDATA_CRC_COUNT 4 -#endif - -// STB_VORBIS_FAST_HUFFMAN_LENGTH [number] -// sets the log size of the huffman-acceleration table. Maximum -// supported value is 24. with larger numbers, more decodings are O(1), -// but the table size is larger so worse cache missing, so you'll have -// to probe (and try multiple ogg vorbis files) to find the sweet spot. -#ifndef STB_VORBIS_FAST_HUFFMAN_LENGTH -#define STB_VORBIS_FAST_HUFFMAN_LENGTH 10 -#endif - -// STB_VORBIS_FAST_BINARY_LENGTH [number] -// sets the log size of the binary-search acceleration table. this -// is used in similar fashion to the fast-huffman size to set initial -// parameters for the binary search - -// STB_VORBIS_FAST_HUFFMAN_INT -// The fast huffman tables are much more efficient if they can be -// stored as 16-bit results instead of 32-bit results. This restricts -// the codebooks to having only 65535 possible outcomes, though. -// (At least, accelerated by the huffman table.) -#ifndef STB_VORBIS_FAST_HUFFMAN_INT -#define STB_VORBIS_FAST_HUFFMAN_SHORT -#endif - -// STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH -// If the 'fast huffman' search doesn't succeed, then stb_vorbis falls -// back on binary searching for the correct one. This requires storing -// extra tables with the huffman codes in sorted order. Defining this -// symbol trades off space for speed by forcing a linear search in the -// non-fast case, except for "sparse" codebooks. -// #define STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH - -// STB_VORBIS_DIVIDES_IN_RESIDUE -// stb_vorbis precomputes the result of the scalar residue decoding -// that would otherwise require a divide per chunk. you can trade off -// space for time by defining this symbol. -// #define STB_VORBIS_DIVIDES_IN_RESIDUE - -// STB_VORBIS_DIVIDES_IN_CODEBOOK -// vorbis VQ codebooks can be encoded two ways: with every case explicitly -// stored, or with all elements being chosen from a small range of values, -// and all values possible in all elements. By default, stb_vorbis expands -// this latter kind out to look like the former kind for ease of decoding, -// because otherwise an integer divide-per-vector-element is required to -// unpack the index. If you define STB_VORBIS_DIVIDES_IN_CODEBOOK, you can -// trade off storage for speed. -//#define STB_VORBIS_DIVIDES_IN_CODEBOOK - -// STB_VORBIS_CODEBOOK_SHORTS -// The vorbis file format encodes VQ codebook floats as ax+b where a and -// b are floating point per-codebook constants, and x is a 16-bit int. -// Normally, stb_vorbis decodes them to floats rather than leaving them -// as 16-bit ints and computing ax+b while decoding. This is a speed/space -// tradeoff; you can save space by defining this flag. -#ifndef STB_VORBIS_CODEBOOK_SHORTS -#define STB_VORBIS_CODEBOOK_FLOATS -#endif - -// STB_VORBIS_DIVIDE_TABLE -// this replaces small integer divides in the floor decode loop with -// table lookups. made less than 1% difference, so disabled by default. - -// STB_VORBIS_NO_INLINE_DECODE -// disables the inlining of the scalar codebook fast-huffman decode. -// might save a little codespace; useful for debugging -// #define STB_VORBIS_NO_INLINE_DECODE - -// STB_VORBIS_NO_DEFER_FLOOR -// Normally we only decode the floor without synthesizing the actual -// full curve. We can instead synthesize the curve immediately. This -// requires more memory and is very likely slower, so I don't think -// you'd ever want to do it except for debugging. -// #define STB_VORBIS_NO_DEFER_FLOOR - - - - -////////////////////////////////////////////////////////////////////////////// - -#ifdef STB_VORBIS_NO_PULLDATA_API - #define STB_VORBIS_NO_INTEGER_CONVERSION - #define STB_VORBIS_NO_STDIO -#endif - -#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO) - #define STB_VORBIS_NO_STDIO 1 -#endif - -#ifndef STB_VORBIS_NO_INTEGER_CONVERSION -#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT - - // only need endianness for fast-float-to-int, which we don't - // use for pushdata - - #ifndef STB_VORBIS_BIG_ENDIAN - #define STB_VORBIS_ENDIAN 0 - #else - #define STB_VORBIS_ENDIAN 1 - #endif - -#endif -#endif - - -#ifndef STB_VORBIS_NO_STDIO -#include -#endif - -#ifndef STB_VORBIS_NO_CRT -#include -#include -#include -#include - -#if !defined(__APPLE__) && !defined(MACOSX) && !defined(macintosh) && !defined(Macintosh) &&!defined(__FreeBSD__) -#include -#endif - -#else -#define NULL 0 -#endif - -#ifndef _MSC_VER - #if __GNUC__ - #define __forceinline inline - #else - #define __forceinline - #endif -#endif - -#if STB_VORBIS_MAX_CHANNELS > 256 -#error "Value of STB_VORBIS_MAX_CHANNELS outside of allowed range" -#endif - -#if STB_VORBIS_FAST_HUFFMAN_LENGTH > 24 -#error "Value of STB_VORBIS_FAST_HUFFMAN_LENGTH outside of allowed range" -#endif - - -#define MAX_BLOCKSIZE_LOG 13 // from specification -#define MAX_BLOCKSIZE (1 << MAX_BLOCKSIZE_LOG) - - -typedef unsigned char uint8; -typedef signed char int8; -typedef unsigned short uint16; -typedef signed short int16; -typedef unsigned int uint32; -typedef signed int int32; - -#ifndef TRUE -#define TRUE 1 -#define FALSE 0 -#endif - -#ifdef STB_VORBIS_CODEBOOK_FLOATS -typedef float codetype; -#else -typedef uint16 codetype; -#endif - -// @NOTE -// -// Some arrays below are tagged "//varies", which means it's actually -// a variable-sized piece of data, but rather than malloc I assume it's -// small enough it's better to just allocate it all together with the -// main thing -// -// Most of the variables are specified with the smallest size I could pack -// them into. It might give better performance to make them all full-sized -// integers. It should be safe to freely rearrange the structures or change -// the sizes larger--nothing relies on silently truncating etc., nor the -// order of variables. - -#define FAST_HUFFMAN_TABLE_SIZE (1 << STB_VORBIS_FAST_HUFFMAN_LENGTH) -#define FAST_HUFFMAN_TABLE_MASK (FAST_HUFFMAN_TABLE_SIZE - 1) - -typedef struct -{ - int dimensions, entries; - uint8 *codeword_lengths; - float minimum_value; - float delta_value; - uint8 value_bits; - uint8 lookup_type; - uint8 sequence_p; - uint8 sparse; - uint32 lookup_values; - codetype *multiplicands; - uint32 *codewords; - #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT - int16 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; - #else - int32 fast_huffman[FAST_HUFFMAN_TABLE_SIZE]; - #endif - uint32 *sorted_codewords; - int *sorted_values; - int sorted_entries; -} Codebook; - -typedef struct -{ - uint8 order; - uint16 rate; - uint16 bark_map_size; - uint8 amplitude_bits; - uint8 amplitude_offset; - uint8 number_of_books; - uint8 book_list[16]; // varies -} Floor0; - -typedef struct -{ - uint8 partitions; - uint8 partition_class_list[32]; // varies - uint8 class_dimensions[16]; // varies - uint8 class_subclasses[16]; // varies - uint8 class_masterbooks[16]; // varies - int16 subclass_books[16][8]; // varies - uint16 Xlist[31*8+2]; // varies - uint8 sorted_order[31*8+2]; - uint8 neighbors[31*8+2][2]; - uint8 floor1_multiplier; - uint8 rangebits; - int values; -} Floor1; - -typedef union -{ - Floor0 floor0; - Floor1 floor1; -} Floor; - -typedef struct -{ - uint32 begin, end; - uint32 part_size; - uint8 classifications; - uint8 classbook; - uint8 **classdata; - int16 (*residue_books)[8]; -} Residue; - -typedef struct -{ - uint8 magnitude; - uint8 angle; - uint8 mux; -} MappingChannel; - -typedef struct -{ - uint16 coupling_steps; - MappingChannel *chan; - uint8 submaps; - uint8 submap_floor[15]; // varies - uint8 submap_residue[15]; // varies -} Mapping; - -typedef struct -{ - uint8 blockflag; - uint8 mapping; - uint16 windowtype; - uint16 transformtype; -} Mode; - -typedef struct -{ - uint32 goal_crc; // expected crc if match - int bytes_left; // bytes left in packet - uint32 crc_so_far; // running crc - int bytes_done; // bytes processed in _current_ chunk - uint32 sample_loc; // granule pos encoded in page -} CRCscan; - -typedef struct -{ - uint32 page_start, page_end; - uint32 after_previous_page_start; - uint32 first_decoded_sample; - uint32 last_decoded_sample; -} ProbedPage; - -struct stb_vorbis -{ - // user-accessible info - unsigned int sample_rate; - int channels; - - unsigned int setup_memory_required; - unsigned int temp_memory_required; - unsigned int setup_temp_memory_required; - - // input config -#ifndef STB_VORBIS_NO_STDIO - FILE *f; - uint32 f_start; - int close_on_free; -#endif - - uint8 *stream; - uint8 *stream_start; - uint8 *stream_end; - - uint32 stream_len; - - uint8 push_mode; - - uint32 first_audio_page_offset; - - ProbedPage p_first, p_last; - - // memory management - stb_vorbis_alloc alloc; - int setup_offset; - int temp_offset; - - // run-time results - int eof; - enum STBVorbisError error; - - // user-useful data - - // header info - int blocksize[2]; - int blocksize_0, blocksize_1; - int codebook_count; - Codebook *codebooks; - int floor_count; - uint16 floor_types[64]; // varies - Floor *floor_config; - int residue_count; - uint16 residue_types[64]; // varies - Residue *residue_config; - int mapping_count; - Mapping *mapping; - int mode_count; - Mode mode_config[64]; // varies - - uint32 total_samples; - - // decode buffer - float *channel_buffers[STB_VORBIS_MAX_CHANNELS]; - float *outputs [STB_VORBIS_MAX_CHANNELS]; - - float *previous_window[STB_VORBIS_MAX_CHANNELS]; - int previous_length; - - #ifndef STB_VORBIS_NO_DEFER_FLOOR - int16 *finalY[STB_VORBIS_MAX_CHANNELS]; - #else - float *floor_buffers[STB_VORBIS_MAX_CHANNELS]; - #endif - - uint32 current_loc; // sample location of next frame to decode - int current_loc_valid; - - // per-blocksize precomputed data - - // twiddle factors - float *A[2],*B[2],*C[2]; - float *window[2]; - uint16 *bit_reverse[2]; - - // current page/packet/segment streaming info - uint32 serial; // stream serial number for verification - int last_page; - int segment_count; - uint8 segments[255]; - uint8 page_flag; - uint8 bytes_in_seg; - uint8 first_decode; - int next_seg; - int last_seg; // flag that we're on the last segment - int last_seg_which; // what was the segment number of the last seg? - uint32 acc; - int valid_bits; - int packet_bytes; - int end_seg_with_known_loc; - uint32 known_loc_for_packet; - int discard_samples_deferred; - uint32 samples_output; - - // push mode scanning - int page_crc_tests; // only in push_mode: number of tests active; -1 if not searching -#ifndef STB_VORBIS_NO_PUSHDATA_API - CRCscan scan[STB_VORBIS_PUSHDATA_CRC_COUNT]; -#endif - - // sample-access - int channel_buffer_start; - int channel_buffer_end; -}; - -extern int my_prof(int slot); -//#define stb_prof my_prof - -#ifndef stb_prof -#define stb_prof(x) 0 -#endif - -#if defined(STB_VORBIS_NO_PUSHDATA_API) - #define IS_PUSH_MODE(f) FALSE -#elif defined(STB_VORBIS_NO_PULLDATA_API) - #define IS_PUSH_MODE(f) TRUE -#else - #define IS_PUSH_MODE(f) ((f)->push_mode) -#endif - -typedef struct stb_vorbis vorb; - -static int error(vorb *f, enum STBVorbisError e) -{ - f->error = e; - if (!f->eof && e != VORBIS_need_more_data) { - f->error=e; // breakpoint for debugging - } - return 0; -} - - -// these functions are used for allocating temporary memory -// while decoding. if you can afford the stack space, use -// alloca(); otherwise, provide a temp buffer and it will -// allocate out of those. - -#define array_size_required(count,size) (count*(sizeof(void *)+(size))) - -#define temp_alloc(f,size) (f->alloc.alloc_buffer ? setup_temp_malloc(f,size) : alloca(size)) -#ifdef dealloca -#define temp_free(f,p) (f->alloc.alloc_buffer ? 0 : dealloca(size)) -#else -#define temp_free(f,p) 0 -#endif -#define temp_alloc_save(f) ((f)->temp_offset) -#define temp_alloc_restore(f,p) ((f)->temp_offset = (p)) - -#define temp_block_array(f,count,size) make_block_array(temp_alloc(f,array_size_required(count,size)), count, size) - -// given a sufficiently large block of memory, make an array of pointers to subblocks of it -static void *make_block_array(void *mem, int count, int size) -{ - int i; - void ** p = (void **) mem; - char *q = (char *) (p + count); - for (i=0; i < count; ++i) { - p[i] = q; - q += size; - } - return p; -} - -static void *setup_malloc(vorb *f, int sz) -{ - sz = (sz+3) & ~3; - f->setup_memory_required += sz; - if (f->alloc.alloc_buffer) { - void *p = (char *) f->alloc.alloc_buffer + f->setup_offset; - if (f->setup_offset + sz > f->temp_offset) return NULL; - f->setup_offset += sz; - return p; - } - return sz ? malloc(sz) : NULL; -} - -static void setup_free(vorb *f, void *p) -{ - if (f->alloc.alloc_buffer) return; // do nothing; setup mem is not a stack - free(p); -} - -static void *setup_temp_malloc(vorb *f, int sz) -{ - sz = (sz+3) & ~3; - if (f->alloc.alloc_buffer) { - if (f->temp_offset - sz < f->setup_offset) return NULL; - f->temp_offset -= sz; - return (char *) f->alloc.alloc_buffer + f->temp_offset; - } - return malloc(sz); -} - -static void setup_temp_free(vorb *f, void *p, size_t sz) -{ - if (f->alloc.alloc_buffer) { - f->temp_offset += (sz+3)&~3; - return; - } - free(p); -} - -#define CRC32_POLY 0x04c11db7 // from spec - -static uint32 crc_table[256]; -static void crc32_init(void) -{ - int i,j; - uint32 s; - for(i=0; i < 256; i++) { - for (s=i<<24, j=0; j < 8; ++j) - s = (s << 1) ^ (s >= (1<<31) ? CRC32_POLY : 0); - crc_table[i] = s; - } -} - -static __forceinline uint32 crc32_update(uint32 crc, uint8 byte) -{ - return (crc << 8) ^ crc_table[byte ^ (crc >> 24)]; -} - - -// used in setup, and for huffman that doesn't go fast path -static unsigned int bit_reverse(unsigned int n) -{ - n = ((n & 0xAAAAAAAA) >> 1) | ((n & 0x55555555) << 1); - n = ((n & 0xCCCCCCCC) >> 2) | ((n & 0x33333333) << 2); - n = ((n & 0xF0F0F0F0) >> 4) | ((n & 0x0F0F0F0F) << 4); - n = ((n & 0xFF00FF00) >> 8) | ((n & 0x00FF00FF) << 8); - return (n >> 16) | (n << 16); -} - -static float square(float x) -{ - return x*x; -} - -// this is a weird definition of log2() for which log2(1) = 1, log2(2) = 2, log2(4) = 3 -// as required by the specification. fast(?) implementation from stb.h -// @OPTIMIZE: called multiple times per-packet with "constants"; move to setup -static int ilog(int32 n) -{ - static signed char log2_4[16] = { 0,1,2,2,3,3,3,3,4,4,4,4,4,4,4,4 }; - - // 2 compares if n < 16, 3 compares otherwise (4 if signed or n > 1<<29) - if (n < (1U << 14)) - if (n < (1U << 4)) return 0 + log2_4[n ]; - else if (n < (1U << 9)) return 5 + log2_4[n >> 5]; - else return 10 + log2_4[n >> 10]; - else if (n < (1U << 24)) - if (n < (1U << 19)) return 15 + log2_4[n >> 15]; - else return 20 + log2_4[n >> 20]; - else if (n < (1U << 29)) return 25 + log2_4[n >> 25]; - else if (n < (1U << 31)) return 30 + log2_4[n >> 30]; - else return 0; // signed n returns 0 -} - -#ifndef M_PI - #define M_PI 3.14159265358979323846264f // from CRC -#endif - -// code length assigned to a value with no huffman encoding -#define NO_CODE 255 - -/////////////////////// LEAF SETUP FUNCTIONS ////////////////////////// -// -// these functions are only called at setup, and only a few times -// per file - -static float float32_unpack(uint32 x) -{ - // from the specification - uint32 mantissa = x & 0x1fffff; - uint32 sign = x & 0x80000000; - uint32 exp = (x & 0x7fe00000) >> 21; - double res = sign ? -(double)mantissa : (double)mantissa; - return (float) ldexp((float)res, exp-788); -} - - -// zlib & jpeg huffman tables assume that the output symbols -// can either be arbitrarily arranged, or have monotonically -// increasing frequencies--they rely on the lengths being sorted; -// this makes for a very simple generation algorithm. -// vorbis allows a huffman table with non-sorted lengths. This -// requires a more sophisticated construction, since symbols in -// order do not map to huffman codes "in order". -static void add_entry(Codebook *c, uint32 huff_code, int symbol, int count, int len, uint32 *values) -{ - if (!c->sparse) { - c->codewords [symbol] = huff_code; - } else { - c->codewords [count] = huff_code; - c->codeword_lengths[count] = len; - values [count] = symbol; - } -} - -static int compute_codewords(Codebook *c, uint8 *len, int n, uint32 *values) -{ - int i,k,m=0; - uint32 available[32]; - - memset(available, 0, sizeof(available)); - // find the first entry - for (k=0; k < n; ++k) if (len[k] < NO_CODE) break; - if (k == n) { assert(c->sorted_entries == 0); return TRUE; } - // add to the list - add_entry(c, 0, k, m++, len[k], values); - // add all available leaves - for (i=1; i <= len[k]; ++i) - available[i] = 1 << (32-i); - // note that the above code treats the first case specially, - // but it's really the same as the following code, so they - // could probably be combined (except the initial code is 0, - // and I use 0 in available[] to mean 'empty') - for (i=k+1; i < n; ++i) { - uint32 res; - int z = len[i], y; - if (z == NO_CODE) continue; - // find lowest available leaf (should always be earliest, - // which is what the specification calls for) - // note that this property, and the fact we can never have - // more than one free leaf at a given level, isn't totally - // trivial to prove, but it seems true and the assert never - // fires, so! - while (z > 0 && !available[z]) --z; - if (z == 0) { assert(0); return FALSE; } - res = available[z]; - available[z] = 0; - add_entry(c, bit_reverse(res), i, m++, len[i], values); - // propogate availability up the tree - if (z != len[i]) { - for (y=len[i]; y > z; --y) { - assert(available[y] == 0); - available[y] = res + (1 << (32-y)); - } - } - } - return TRUE; -} - -// accelerated huffman table allows fast O(1) match of all symbols -// of length <= STB_VORBIS_FAST_HUFFMAN_LENGTH -static void compute_accelerated_huffman(Codebook *c) -{ - int i, len; - for (i=0; i < FAST_HUFFMAN_TABLE_SIZE; ++i) - c->fast_huffman[i] = -1; - - len = c->sparse ? c->sorted_entries : c->entries; - #ifdef STB_VORBIS_FAST_HUFFMAN_SHORT - if (len > 32767) len = 32767; // largest possible value we can encode! - #endif - for (i=0; i < len; ++i) { - if (c->codeword_lengths[i] <= STB_VORBIS_FAST_HUFFMAN_LENGTH) { - uint32 z = c->sparse ? bit_reverse(c->sorted_codewords[i]) : c->codewords[i]; - // set table entries for all bit combinations in the higher bits - while (z < FAST_HUFFMAN_TABLE_SIZE) { - c->fast_huffman[z] = i; - z += 1 << c->codeword_lengths[i]; - } - } - } -} - -static int uint32_compare(const void *p, const void *q) -{ - uint32 x = * (uint32 *) p; - uint32 y = * (uint32 *) q; - return x < y ? -1 : x > y; -} - -static int include_in_sort(Codebook *c, uint8 len) -{ - if (c->sparse) { assert(len != NO_CODE); return TRUE; } - if (len == NO_CODE) return FALSE; - if (len > STB_VORBIS_FAST_HUFFMAN_LENGTH) return TRUE; - return FALSE; -} - -// if the fast table above doesn't work, we want to binary -// search them... need to reverse the bits -static void compute_sorted_huffman(Codebook *c, uint8 *lengths, uint32 *values) -{ - int i, len; - // build a list of all the entries - // OPTIMIZATION: don't include the short ones, since they'll be caught by FAST_HUFFMAN. - // this is kind of a frivolous optimization--I don't see any performance improvement, - // but it's like 4 extra lines of code, so. - if (!c->sparse) { - int k = 0; - for (i=0; i < c->entries; ++i) - if (include_in_sort(c, lengths[i])) - c->sorted_codewords[k++] = bit_reverse(c->codewords[i]); - assert(k == c->sorted_entries); - } else { - for (i=0; i < c->sorted_entries; ++i) - c->sorted_codewords[i] = bit_reverse(c->codewords[i]); - } - - qsort(c->sorted_codewords, c->sorted_entries, sizeof(c->sorted_codewords[0]), uint32_compare); - c->sorted_codewords[c->sorted_entries] = 0xffffffff; - - len = c->sparse ? c->sorted_entries : c->entries; - // now we need to indicate how they correspond; we could either - // #1: sort a different data structure that says who they correspond to - // #2: for each sorted entry, search the original list to find who corresponds - // #3: for each original entry, find the sorted entry - // #1 requires extra storage, #2 is slow, #3 can use binary search! - for (i=0; i < len; ++i) { - int huff_len = c->sparse ? lengths[values[i]] : lengths[i]; - if (include_in_sort(c,huff_len)) { - uint32 code = bit_reverse(c->codewords[i]); - int x=0, n=c->sorted_entries; - while (n > 1) { - // invariant: sc[x] <= code < sc[x+n] - int m = x + (n >> 1); - if (c->sorted_codewords[m] <= code) { - x = m; - n -= (n>>1); - } else { - n >>= 1; - } - } - assert(c->sorted_codewords[x] == code); - if (c->sparse) { - c->sorted_values[x] = values[i]; - c->codeword_lengths[x] = huff_len; - } else { - c->sorted_values[x] = i; - } - } - } -} - -// only run while parsing the header (3 times) -static int vorbis_validate(uint8 *data) -{ - static uint8 vorbis[6] = { 'v', 'o', 'r', 'b', 'i', 's' }; - return memcmp(data, vorbis, 6) == 0; -} - -// called from setup only, once per code book -// (formula implied by specification) -static int lookup1_values(int entries, int dim) -{ - int r = (int) floor(exp((float) log((float) entries) / dim)); - if ((int) floor(pow((float) r+1, dim)) <= entries) // (int) cast for MinGW warning; - ++r; // floor() to avoid _ftol() when non-CRT - assert(pow((float) r+1, dim) > entries); - assert((int) floor(pow((float) r, dim)) <= entries); // (int),floor() as above - return r; -} - -// called twice per file -static void compute_twiddle_factors(int n, float *A, float *B, float *C) -{ - int n4 = n >> 2, n8 = n >> 3; - int k,k2; - - for (k=k2=0; k < n4; ++k,k2+=2) { - A[k2 ] = (float) cos(4*k*M_PI/n); - A[k2+1] = (float) -sin(4*k*M_PI/n); - B[k2 ] = (float) cos((k2+1)*M_PI/n/2) * 0.5f; - B[k2+1] = (float) sin((k2+1)*M_PI/n/2) * 0.5f; - } - for (k=k2=0; k < n8; ++k,k2+=2) { - C[k2 ] = (float) cos(2*(k2+1)*M_PI/n); - C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n); - } -} - -static void compute_window(int n, float *window) -{ - int n2 = n >> 1, i; - for (i=0; i < n2; ++i) - window[i] = (float) sin(0.5 * M_PI * square((float) sin((i - 0 + 0.5) / n2 * 0.5 * M_PI))); -} - -static void compute_bitreverse(int n, uint16 *rev) -{ - int ld = ilog(n) - 1; // ilog is off-by-one from normal definitions - int i, n8 = n >> 3; - for (i=0; i < n8; ++i) - rev[i] = (bit_reverse(i) >> (32-ld+3)) << 2; -} - -static int init_blocksize(vorb *f, int b, int n) -{ - int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3; - f->A[b] = (float *) setup_malloc(f, sizeof(float) * n2); - f->B[b] = (float *) setup_malloc(f, sizeof(float) * n2); - f->C[b] = (float *) setup_malloc(f, sizeof(float) * n4); - if (!f->A[b] || !f->B[b] || !f->C[b]) return error(f, VORBIS_outofmem); - compute_twiddle_factors(n, f->A[b], f->B[b], f->C[b]); - f->window[b] = (float *) setup_malloc(f, sizeof(float) * n2); - if (!f->window[b]) return error(f, VORBIS_outofmem); - compute_window(n, f->window[b]); - f->bit_reverse[b] = (uint16 *) setup_malloc(f, sizeof(uint16) * n8); - if (!f->bit_reverse[b]) return error(f, VORBIS_outofmem); - compute_bitreverse(n, f->bit_reverse[b]); - return TRUE; -} - -static void neighbors(uint16 *x, int n, int *plow, int *phigh) -{ - int low = -1; - int high = 65536; - int i; - for (i=0; i < n; ++i) { - if (x[i] > low && x[i] < x[n]) { *plow = i; low = x[i]; } - if (x[i] < high && x[i] > x[n]) { *phigh = i; high = x[i]; } - } -} - -// this has been repurposed so y is now the original index instead of y -typedef struct -{ - uint16 x,y; -} Point; - -int point_compare(const void *p, const void *q) -{ - Point *a = (Point *) p; - Point *b = (Point *) q; - return a->x < b->x ? -1 : a->x > b->x; -} - -// -/////////////////////// END LEAF SETUP FUNCTIONS ////////////////////////// - - -#if defined(STB_VORBIS_NO_STDIO) - #define USE_MEMORY(z) TRUE -#else - #define USE_MEMORY(z) ((z)->stream) -#endif - -static uint8 get8(vorb *z) -{ - if (USE_MEMORY(z)) { - if (z->stream >= z->stream_end) { z->eof = TRUE; return 0; } - return *z->stream++; - } - - #ifndef STB_VORBIS_NO_STDIO - { - int c = fgetc(z->f); - if (c == EOF) { z->eof = TRUE; return 0; } - return c; - } - #endif -} - -static uint32 get32(vorb *f) -{ - uint32 x; - x = get8(f); - x += get8(f) << 8; - x += get8(f) << 16; - x += get8(f) << 24; - return x; -} - -static int getn(vorb *z, uint8 *data, int n) -{ - if (USE_MEMORY(z)) { - if (z->stream+n > z->stream_end) { z->eof = 1; return 0; } - memcpy(data, z->stream, n); - z->stream += n; - return 1; - } - - #ifndef STB_VORBIS_NO_STDIO - if (fread(data, n, 1, z->f) == 1) - return 1; - else { - z->eof = 1; - return 0; - } - #endif -} - -static void skip(vorb *z, int n) -{ - if (USE_MEMORY(z)) { - z->stream += n; - if (z->stream >= z->stream_end) z->eof = 1; - return; - } - #ifndef STB_VORBIS_NO_STDIO - { - long x = ftell(z->f); - fseek(z->f, x+n, SEEK_SET); - } - #endif -} - -static int set_file_offset(stb_vorbis *f, unsigned int loc) -{ - #ifndef STB_VORBIS_NO_PUSHDATA_API - if (f->push_mode) return 0; - #endif - f->eof = 0; - if (USE_MEMORY(f)) { - if (f->stream_start + loc >= f->stream_end || f->stream_start + loc < f->stream_start) { - f->stream = f->stream_end; - f->eof = 1; - return 0; - } else { - f->stream = f->stream_start + loc; - return 1; - } - } - #ifndef STB_VORBIS_NO_STDIO - if (loc + f->f_start < loc || loc >= 0x80000000) { - loc = 0x7fffffff; - f->eof = 1; - } else { - loc += f->f_start; - } - if (!fseek(f->f, loc, SEEK_SET)) - return 1; - f->eof = 1; - fseek(f->f, f->f_start, SEEK_END); - return 0; - #endif -} - - -static uint8 ogg_page_header[4] = { 0x4f, 0x67, 0x67, 0x53 }; - -static int capture_pattern(vorb *f) -{ - if (0x4f != get8(f)) return FALSE; - if (0x67 != get8(f)) return FALSE; - if (0x67 != get8(f)) return FALSE; - if (0x53 != get8(f)) return FALSE; - return TRUE; -} - -#define PAGEFLAG_continued_packet 1 -#define PAGEFLAG_first_page 2 -#define PAGEFLAG_last_page 4 - -static int start_page_no_capturepattern(vorb *f) -{ - uint32 loc0,loc1,n,i; - // stream structure version - if (0 != get8(f)) return error(f, VORBIS_invalid_stream_structure_version); - // header flag - f->page_flag = get8(f); - // absolute granule position - loc0 = get32(f); - loc1 = get32(f); - // @TODO: validate loc0,loc1 as valid positions? - // stream serial number -- vorbis doesn't interleave, so discard - get32(f); - //if (f->serial != get32(f)) return error(f, VORBIS_incorrect_stream_serial_number); - // page sequence number - n = get32(f); - f->last_page = n; - // CRC32 - get32(f); - // page_segments - f->segment_count = get8(f); - if (!getn(f, f->segments, f->segment_count)) - return error(f, VORBIS_unexpected_eof); - // assume we _don't_ know any the sample position of any segments - f->end_seg_with_known_loc = -2; - if (loc0 != ~0 || loc1 != ~0) { - // determine which packet is the last one that will complete - for (i=f->segment_count-1; i >= 0; --i) - if (f->segments[i] < 255) - break; - // 'i' is now the index of the _last_ segment of a packet that ends - if (i >= 0) { - f->end_seg_with_known_loc = i; - f->known_loc_for_packet = loc0; - } - } - if (f->first_decode) { - int i,len; - ProbedPage p; - len = 0; - for (i=0; i < f->segment_count; ++i) - len += f->segments[i]; - len += 27 + f->segment_count; - p.page_start = f->first_audio_page_offset; - p.page_end = p.page_start + len; - p.after_previous_page_start = p.page_start; - p.first_decoded_sample = 0; - p.last_decoded_sample = loc0; - f->p_first = p; - } - f->next_seg = 0; - return TRUE; -} - -static int start_page(vorb *f) -{ - if (!capture_pattern(f)) return error(f, VORBIS_missing_capture_pattern); - return start_page_no_capturepattern(f); -} - -static int start_packet(vorb *f) -{ - while (f->next_seg == -1) { - if (!start_page(f)) return FALSE; - if (f->page_flag & PAGEFLAG_continued_packet) - return error(f, VORBIS_continued_packet_flag_invalid); - } - f->last_seg = FALSE; - f->valid_bits = 0; - f->packet_bytes = 0; - f->bytes_in_seg = 0; - // f->next_seg is now valid - return TRUE; -} - -static int maybe_start_packet(vorb *f) -{ - if (f->next_seg == -1) { - int x = get8(f); - if (f->eof) return FALSE; // EOF at page boundary is not an error! - if (0x4f != x ) return error(f, VORBIS_missing_capture_pattern); - if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern); - if (0x67 != get8(f)) return error(f, VORBIS_missing_capture_pattern); - if (0x53 != get8(f)) return error(f, VORBIS_missing_capture_pattern); - if (!start_page_no_capturepattern(f)) return FALSE; - if (f->page_flag & PAGEFLAG_continued_packet) { - // set up enough state that we can read this packet if we want, - // e.g. during recovery - f->last_seg = FALSE; - f->bytes_in_seg = 0; - return error(f, VORBIS_continued_packet_flag_invalid); - } - } - return start_packet(f); -} - -static int next_segment(vorb *f) -{ - int len; - if (f->last_seg) return 0; - if (f->next_seg == -1) { - f->last_seg_which = f->segment_count-1; // in case start_page fails - if (!start_page(f)) { f->last_seg = 1; return 0; } - if (!(f->page_flag & PAGEFLAG_continued_packet)) return error(f, VORBIS_continued_packet_flag_invalid); - } - len = f->segments[f->next_seg++]; - if (len < 255) { - f->last_seg = TRUE; - f->last_seg_which = f->next_seg-1; - } - if (f->next_seg >= f->segment_count) - f->next_seg = -1; - assert(f->bytes_in_seg == 0); - f->bytes_in_seg = len; - return len; -} - -#define EOP (-1) -#define INVALID_BITS (-1) - -static int get8_packet_raw(vorb *f) -{ - if (!f->bytes_in_seg) - if (f->last_seg) return EOP; - else if (!next_segment(f)) return EOP; - assert(f->bytes_in_seg > 0); - --f->bytes_in_seg; - ++f->packet_bytes; - return get8(f); -} - -static int get8_packet(vorb *f) -{ - int x = get8_packet_raw(f); - f->valid_bits = 0; - return x; -} - -static void flush_packet(vorb *f) -{ - while (get8_packet_raw(f) != EOP); -} - -// @OPTIMIZE: this is the secondary bit decoder, so it's probably not as important -// as the huffman decoder? -static uint32 get_bits(vorb *f, int n) -{ - uint32 z; - - if (f->valid_bits < 0) return 0; - if (f->valid_bits < n) { - if (n > 24) { - // the accumulator technique below would not work correctly in this case - z = get_bits(f, 24); - z += get_bits(f, n-24) << 24; - return z; - } - if (f->valid_bits == 0) f->acc = 0; - while (f->valid_bits < n) { - int z = get8_packet_raw(f); - if (z == EOP) { - f->valid_bits = INVALID_BITS; - return 0; - } - f->acc += z << f->valid_bits; - f->valid_bits += 8; - } - } - if (f->valid_bits < 0) return 0; - z = f->acc & ((1 << n)-1); - f->acc >>= n; - f->valid_bits -= n; - return z; -} - -static int32 get_bits_signed(vorb *f, int n) -{ - uint32 z = get_bits(f, n); - if (z & (1 << (n-1))) - z += ~((1 << n) - 1); - return (int32) z; -} - -// @OPTIMIZE: primary accumulator for huffman -// expand the buffer to as many bits as possible without reading off end of packet -// it might be nice to allow f->valid_bits and f->acc to be stored in registers, -// e.g. cache them locally and decode locally -static __forceinline void prep_huffman(vorb *f) -{ - if (f->valid_bits <= 24) { - if (f->valid_bits == 0) f->acc = 0; - do { - int z; - if (f->last_seg && !f->bytes_in_seg) return; - z = get8_packet_raw(f); - if (z == EOP) return; - f->acc += z << f->valid_bits; - f->valid_bits += 8; - } while (f->valid_bits <= 24); - } -} - -enum -{ - VORBIS_packet_id = 1, - VORBIS_packet_comment = 3, - VORBIS_packet_setup = 5, -}; - -static int codebook_decode_scalar_raw(vorb *f, Codebook *c) -{ - int i; - prep_huffman(f); - - assert(c->sorted_codewords || c->codewords); - // cases to use binary search: sorted_codewords && !c->codewords - // sorted_codewords && c->entries > 8 - if (c->entries > 8 ? c->sorted_codewords!=NULL : !c->codewords) { - // binary search - uint32 code = bit_reverse(f->acc); - int x=0, n=c->sorted_entries, len; - - while (n > 1) { - // invariant: sc[x] <= code < sc[x+n] - int m = x + (n >> 1); - if (c->sorted_codewords[m] <= code) { - x = m; - n -= (n>>1); - } else { - n >>= 1; - } - } - // x is now the sorted index - if (!c->sparse) x = c->sorted_values[x]; - // x is now sorted index if sparse, or symbol otherwise - len = c->codeword_lengths[x]; - if (f->valid_bits >= len) { - f->acc >>= len; - f->valid_bits -= len; - return x; - } - - f->valid_bits = 0; - return -1; - } - - // if small, linear search - assert(!c->sparse); - for (i=0; i < c->entries; ++i) { - if (c->codeword_lengths[i] == NO_CODE) continue; - if (c->codewords[i] == (f->acc & ((1 << c->codeword_lengths[i])-1))) { - if (f->valid_bits >= c->codeword_lengths[i]) { - f->acc >>= c->codeword_lengths[i]; - f->valid_bits -= c->codeword_lengths[i]; - return i; - } - f->valid_bits = 0; - return -1; - } - } - - error(f, VORBIS_invalid_stream); - f->valid_bits = 0; - return -1; -} - -static int codebook_decode_scalar(vorb *f, Codebook *c) -{ - int i; - if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) - prep_huffman(f); - // fast huffman table lookup - i = f->acc & FAST_HUFFMAN_TABLE_MASK; - i = c->fast_huffman[i]; - if (i >= 0) { - f->acc >>= c->codeword_lengths[i]; - f->valid_bits -= c->codeword_lengths[i]; - if (f->valid_bits < 0) { f->valid_bits = 0; return -1; } - return i; - } - return codebook_decode_scalar_raw(f,c); -} - -#ifndef STB_VORBIS_NO_INLINE_DECODE - -#define DECODE_RAW(var, f,c) \ - if (f->valid_bits < STB_VORBIS_FAST_HUFFMAN_LENGTH) \ - prep_huffman(f); \ - var = f->acc & FAST_HUFFMAN_TABLE_MASK; \ - var = c->fast_huffman[var]; \ - if (var >= 0) { \ - int n = c->codeword_lengths[var]; \ - f->acc >>= n; \ - f->valid_bits -= n; \ - if (f->valid_bits < 0) { f->valid_bits = 0; var = -1; } \ - } else { \ - var = codebook_decode_scalar_raw(f,c); \ - } - -#else - -#define DECODE_RAW(var,f,c) var = codebook_decode_scalar(f,c); - -#endif - -#define DECODE(var,f,c) \ - DECODE_RAW(var,f,c) \ - if (c->sparse) var = c->sorted_values[var]; - -#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK - #define DECODE_VQ(var,f,c) DECODE_RAW(var,f,c) -#else - #define DECODE_VQ(var,f,c) DECODE(var,f,c) -#endif - - - - - - -// CODEBOOK_ELEMENT_FAST is an optimization for the CODEBOOK_FLOATS case -// where we avoid one addition -#ifndef STB_VORBIS_CODEBOOK_FLOATS - #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off] * c->delta_value + c->minimum_value) - #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off] * c->delta_value) - #define CODEBOOK_ELEMENT_BASE(c) (c->minimum_value) -#else - #define CODEBOOK_ELEMENT(c,off) (c->multiplicands[off]) - #define CODEBOOK_ELEMENT_FAST(c,off) (c->multiplicands[off]) - #define CODEBOOK_ELEMENT_BASE(c) (0) -#endif - -static int codebook_decode_start(vorb *f, Codebook *c, int len) -{ - int z = -1; - - // type 0 is only legal in a scalar context - if (c->lookup_type == 0) - error(f, VORBIS_invalid_stream); - else { - DECODE_VQ(z,f,c); - if (c->sparse) assert(z < c->sorted_entries); - if (z < 0) { // check for EOP - if (!f->bytes_in_seg) - if (f->last_seg) - return z; - error(f, VORBIS_invalid_stream); - } - } - return z; -} - -static int codebook_decode(vorb *f, Codebook *c, float *output, int len) -{ - int i,z = codebook_decode_start(f,c,len); - if (z < 0) return FALSE; - if (len > c->dimensions) len = c->dimensions; - -#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK - if (c->lookup_type == 1) { - float last = CODEBOOK_ELEMENT_BASE(c); - int div = 1; - for (i=0; i < len; ++i) { - int off = (z / div) % c->lookup_values; - float val = CODEBOOK_ELEMENT_FAST(c,off) + last; - output[i] += val; - if (c->sequence_p) last = val + c->minimum_value; - div *= c->lookup_values; - } - return TRUE; - } -#endif - - z *= c->dimensions; - if (c->sequence_p) { - float last = CODEBOOK_ELEMENT_BASE(c); - for (i=0; i < len; ++i) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - output[i] += val; - last = val + c->minimum_value; - } - } else { - float last = CODEBOOK_ELEMENT_BASE(c); - for (i=0; i < len; ++i) { - output[i] += CODEBOOK_ELEMENT_FAST(c,z+i) + last; - } - } - - return TRUE; -} - -static int codebook_decode_step(vorb *f, Codebook *c, float *output, int len, int step) -{ - int i,z = codebook_decode_start(f,c,len); - float last = CODEBOOK_ELEMENT_BASE(c); - if (z < 0) return FALSE; - if (len > c->dimensions) len = c->dimensions; - -#ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK - if (c->lookup_type == 1) { - int div = 1; - for (i=0; i < len; ++i) { - int off = (z / div) % c->lookup_values; - float val = CODEBOOK_ELEMENT_FAST(c,off) + last; - output[i*step] += val; - if (c->sequence_p) last = val; - div *= c->lookup_values; - } - return TRUE; - } -#endif - - z *= c->dimensions; - for (i=0; i < len; ++i) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - output[i*step] += val; - if (c->sequence_p) last = val; - } - - return TRUE; -} - -static int codebook_decode_deinterleave_repeat(vorb *f, Codebook *c, float **outputs, int ch, int *c_inter_p, int *p_inter_p, int len, int total_decode) -{ - int c_inter = *c_inter_p; - int p_inter = *p_inter_p; - int i,z, effective = c->dimensions; - - // type 0 is only legal in a scalar context - if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream); - - while (total_decode > 0) { - float last = CODEBOOK_ELEMENT_BASE(c); - DECODE_VQ(z,f,c); - #ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK - assert(!c->sparse || z < c->sorted_entries); - #endif - if (z < 0) { - if (!f->bytes_in_seg) - if (f->last_seg) return FALSE; - return error(f, VORBIS_invalid_stream); - } - - // if this will take us off the end of the buffers, stop short! - // we check by computing the length of the virtual interleaved - // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter), - // and the length we'll be using (effective) - if (c_inter + p_inter*ch + effective > len * ch) { - effective = len*ch - (p_inter*ch - c_inter); - } - - #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK - if (c->lookup_type == 1) { - int div = 1; - for (i=0; i < effective; ++i) { - int off = (z / div) % c->lookup_values; - float val = CODEBOOK_ELEMENT_FAST(c,off) + last; - outputs[c_inter][p_inter] += val; - if (++c_inter == ch) { c_inter = 0; ++p_inter; } - if (c->sequence_p) last = val; - div *= c->lookup_values; - } - } else - #endif - { - z *= c->dimensions; - if (c->sequence_p) { - for (i=0; i < effective; ++i) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - if (++c_inter == ch) { c_inter = 0; ++p_inter; } - last = val; - } - } else { - for (i=0; i < effective; ++i) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - if (++c_inter == ch) { c_inter = 0; ++p_inter; } - } - } - } - - total_decode -= effective; - } - *c_inter_p = c_inter; - *p_inter_p = p_inter; - return TRUE; -} - -#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK -static int codebook_decode_deinterleave_repeat_2(vorb *f, Codebook *c, float **outputs, int *c_inter_p, int *p_inter_p, int len, int total_decode) -{ - int c_inter = *c_inter_p; - int p_inter = *p_inter_p; - int i,z, effective = c->dimensions; - - // type 0 is only legal in a scalar context - if (c->lookup_type == 0) return error(f, VORBIS_invalid_stream); - - while (total_decode > 0) { - float last = CODEBOOK_ELEMENT_BASE(c); - DECODE_VQ(z,f,c); - - if (z < 0) { - if (!f->bytes_in_seg) - if (f->last_seg) return FALSE; - return error(f, VORBIS_invalid_stream); - } - - // if this will take us off the end of the buffers, stop short! - // we check by computing the length of the virtual interleaved - // buffer (len*ch), our current offset within it (p_inter*ch)+(c_inter), - // and the length we'll be using (effective) - if (c_inter + p_inter*2 + effective > len * 2) { - effective = len*2 - (p_inter*2 - c_inter); - } - - { - z *= c->dimensions; - stb_prof(11); - if (c->sequence_p) { - // haven't optimized this case because I don't have any examples - for (i=0; i < effective; ++i) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - if (++c_inter == 2) { c_inter = 0; ++p_inter; } - last = val; - } - } else { - i=0; - if (c_inter == 1) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - c_inter = 0; ++p_inter; - ++i; - } - { - float *z0 = outputs[0]; - float *z1 = outputs[1]; - for (; i+1 < effective;) { - z0[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i) + last; - z1[p_inter] += CODEBOOK_ELEMENT_FAST(c,z+i+1) + last; - ++p_inter; - i += 2; - } - } - if (i < effective) { - float val = CODEBOOK_ELEMENT_FAST(c,z+i) + last; - outputs[c_inter][p_inter] += val; - if (++c_inter == 2) { c_inter = 0; ++p_inter; } - } - } - } - - total_decode -= effective; - } - *c_inter_p = c_inter; - *p_inter_p = p_inter; - return TRUE; -} -#endif - -static int predict_point(int x, int x0, int x1, int y0, int y1) -{ - int dy = y1 - y0; - int adx = x1 - x0; - // @OPTIMIZE: force int division to round in the right direction... is this necessary on x86? - int err = abs(dy) * (x - x0); - int off = err / adx; - return dy < 0 ? y0 - off : y0 + off; -} - -// the following table is block-copied from the specification -static float inverse_db_table[256] = -{ - 1.0649863e-07f, 1.1341951e-07f, 1.2079015e-07f, 1.2863978e-07f, - 1.3699951e-07f, 1.4590251e-07f, 1.5538408e-07f, 1.6548181e-07f, - 1.7623575e-07f, 1.8768855e-07f, 1.9988561e-07f, 2.1287530e-07f, - 2.2670913e-07f, 2.4144197e-07f, 2.5713223e-07f, 2.7384213e-07f, - 2.9163793e-07f, 3.1059021e-07f, 3.3077411e-07f, 3.5226968e-07f, - 3.7516214e-07f, 3.9954229e-07f, 4.2550680e-07f, 4.5315863e-07f, - 4.8260743e-07f, 5.1396998e-07f, 5.4737065e-07f, 5.8294187e-07f, - 6.2082472e-07f, 6.6116941e-07f, 7.0413592e-07f, 7.4989464e-07f, - 7.9862701e-07f, 8.5052630e-07f, 9.0579828e-07f, 9.6466216e-07f, - 1.0273513e-06f, 1.0941144e-06f, 1.1652161e-06f, 1.2409384e-06f, - 1.3215816e-06f, 1.4074654e-06f, 1.4989305e-06f, 1.5963394e-06f, - 1.7000785e-06f, 1.8105592e-06f, 1.9282195e-06f, 2.0535261e-06f, - 2.1869758e-06f, 2.3290978e-06f, 2.4804557e-06f, 2.6416497e-06f, - 2.8133190e-06f, 2.9961443e-06f, 3.1908506e-06f, 3.3982101e-06f, - 3.6190449e-06f, 3.8542308e-06f, 4.1047004e-06f, 4.3714470e-06f, - 4.6555282e-06f, 4.9580707e-06f, 5.2802740e-06f, 5.6234160e-06f, - 5.9888572e-06f, 6.3780469e-06f, 6.7925283e-06f, 7.2339451e-06f, - 7.7040476e-06f, 8.2047000e-06f, 8.7378876e-06f, 9.3057248e-06f, - 9.9104632e-06f, 1.0554501e-05f, 1.1240392e-05f, 1.1970856e-05f, - 1.2748789e-05f, 1.3577278e-05f, 1.4459606e-05f, 1.5399272e-05f, - 1.6400004e-05f, 1.7465768e-05f, 1.8600792e-05f, 1.9809576e-05f, - 2.1096914e-05f, 2.2467911e-05f, 2.3928002e-05f, 2.5482978e-05f, - 2.7139006e-05f, 2.8902651e-05f, 3.0780908e-05f, 3.2781225e-05f, - 3.4911534e-05f, 3.7180282e-05f, 3.9596466e-05f, 4.2169667e-05f, - 4.4910090e-05f, 4.7828601e-05f, 5.0936773e-05f, 5.4246931e-05f, - 5.7772202e-05f, 6.1526565e-05f, 6.5524908e-05f, 6.9783085e-05f, - 7.4317983e-05f, 7.9147585e-05f, 8.4291040e-05f, 8.9768747e-05f, - 9.5602426e-05f, 0.00010181521f, 0.00010843174f, 0.00011547824f, - 0.00012298267f, 0.00013097477f, 0.00013948625f, 0.00014855085f, - 0.00015820453f, 0.00016848555f, 0.00017943469f, 0.00019109536f, - 0.00020351382f, 0.00021673929f, 0.00023082423f, 0.00024582449f, - 0.00026179955f, 0.00027881276f, 0.00029693158f, 0.00031622787f, - 0.00033677814f, 0.00035866388f, 0.00038197188f, 0.00040679456f, - 0.00043323036f, 0.00046138411f, 0.00049136745f, 0.00052329927f, - 0.00055730621f, 0.00059352311f, 0.00063209358f, 0.00067317058f, - 0.00071691700f, 0.00076350630f, 0.00081312324f, 0.00086596457f, - 0.00092223983f, 0.00098217216f, 0.0010459992f, 0.0011139742f, - 0.0011863665f, 0.0012634633f, 0.0013455702f, 0.0014330129f, - 0.0015261382f, 0.0016253153f, 0.0017309374f, 0.0018434235f, - 0.0019632195f, 0.0020908006f, 0.0022266726f, 0.0023713743f, - 0.0025254795f, 0.0026895994f, 0.0028643847f, 0.0030505286f, - 0.0032487691f, 0.0034598925f, 0.0036847358f, 0.0039241906f, - 0.0041792066f, 0.0044507950f, 0.0047400328f, 0.0050480668f, - 0.0053761186f, 0.0057254891f, 0.0060975636f, 0.0064938176f, - 0.0069158225f, 0.0073652516f, 0.0078438871f, 0.0083536271f, - 0.0088964928f, 0.009474637f, 0.010090352f, 0.010746080f, - 0.011444421f, 0.012188144f, 0.012980198f, 0.013823725f, - 0.014722068f, 0.015678791f, 0.016697687f, 0.017782797f, - 0.018938423f, 0.020169149f, 0.021479854f, 0.022875735f, - 0.024362330f, 0.025945531f, 0.027631618f, 0.029427276f, - 0.031339626f, 0.033376252f, 0.035545228f, 0.037855157f, - 0.040315199f, 0.042935108f, 0.045725273f, 0.048696758f, - 0.051861348f, 0.055231591f, 0.058820850f, 0.062643361f, - 0.066714279f, 0.071049749f, 0.075666962f, 0.080584227f, - 0.085821044f, 0.091398179f, 0.097337747f, 0.10366330f, - 0.11039993f, 0.11757434f, 0.12521498f, 0.13335215f, - 0.14201813f, 0.15124727f, 0.16107617f, 0.17154380f, - 0.18269168f, 0.19456402f, 0.20720788f, 0.22067342f, - 0.23501402f, 0.25028656f, 0.26655159f, 0.28387361f, - 0.30232132f, 0.32196786f, 0.34289114f, 0.36517414f, - 0.38890521f, 0.41417847f, 0.44109412f, 0.46975890f, - 0.50028648f, 0.53279791f, 0.56742212f, 0.60429640f, - 0.64356699f, 0.68538959f, 0.72993007f, 0.77736504f, - 0.82788260f, 0.88168307f, 0.9389798f, 1.0f -}; - - -// @OPTIMIZE: if you want to replace this bresenham line-drawing routine, -// note that you must produce bit-identical output to decode correctly; -// this specific sequence of operations is specified in the spec (it's -// drawing integer-quantized frequency-space lines that the encoder -// expects to be exactly the same) -// ... also, isn't the whole point of Bresenham's algorithm to NOT -// have to divide in the setup? sigh. -#ifndef STB_VORBIS_NO_DEFER_FLOOR -#define LINE_OP(a,b) a *= b -#else -#define LINE_OP(a,b) a = b -#endif - -#ifdef STB_VORBIS_DIVIDE_TABLE -#define DIVTAB_NUMER 32 -#define DIVTAB_DENOM 64 -int8 integer_divide_table[DIVTAB_NUMER][DIVTAB_DENOM]; // 2KB -#endif - -static __forceinline void draw_line(float *output, int x0, int y0, int x1, int y1, int n) -{ - int dy = y1 - y0; - int adx = x1 - x0; - int ady = abs(dy); - int base; - int x=x0,y=y0; - int err = 0; - int sy; - -#ifdef STB_VORBIS_DIVIDE_TABLE - if (adx < DIVTAB_DENOM && ady < DIVTAB_NUMER) { - if (dy < 0) { - base = -integer_divide_table[ady][adx]; - sy = base-1; - } else { - base = integer_divide_table[ady][adx]; - sy = base+1; - } - } else { - base = dy / adx; - if (dy < 0) - sy = base - 1; - else - sy = base+1; - } -#else - base = dy / adx; - if (dy < 0) - sy = base - 1; - else - sy = base+1; -#endif - ady -= abs(base) * adx; - if (x1 > n) x1 = n; - LINE_OP(output[x], inverse_db_table[y]); - for (++x; x < x1; ++x) { - err += ady; - if (err >= adx) { - err -= adx; - y += sy; - } else - y += base; - LINE_OP(output[x], inverse_db_table[y]); - } -} - -static int residue_decode(vorb *f, Codebook *book, float *target, int offset, int n, int rtype) -{ - int k; - if (rtype == 0) { - int step = n / book->dimensions; - for (k=0; k < step; ++k) - if (!codebook_decode_step(f, book, target+offset+k, n-offset-k, step)) - return FALSE; - } else { - for (k=0; k < n; ) { - if (!codebook_decode(f, book, target+offset, n-k)) - return FALSE; - k += book->dimensions; - offset += book->dimensions; - } - } - return TRUE; -} - -static void decode_residue(vorb *f, float *residue_buffers[], int ch, int n, int rn, uint8 *do_not_decode) -{ - int i,j,pass; - Residue *r = f->residue_config + rn; - int rtype = f->residue_types[rn]; - int c = r->classbook; - int classwords = f->codebooks[c].dimensions; - int n_read = r->end - r->begin; - int part_read = n_read / r->part_size; - int temp_alloc_point = temp_alloc_save(f); - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - uint8 ***part_classdata = (uint8 ***) temp_block_array(f,f->channels, part_read * sizeof(**part_classdata)); - #else - int **classifications = (int **) temp_block_array(f,f->channels, part_read * sizeof(**classifications)); - #endif - - stb_prof(2); - for (i=0; i < ch; ++i) - if (!do_not_decode[i]) - memset(residue_buffers[i], 0, sizeof(float) * n); - - if (rtype == 2 && ch != 1) { - int len = ch * n; - for (j=0; j < ch; ++j) - if (!do_not_decode[j]) - break; - if (j == ch) - goto done; - - stb_prof(3); - for (pass=0; pass < 8; ++pass) { - int pcount = 0, class_set = 0; - if (ch == 2) { - stb_prof(13); - while (pcount < part_read) { - int z = r->begin + pcount*r->part_size; - int c_inter = (z & 1), p_inter = z>>1; - if (pass == 0) { - Codebook *c = f->codebooks+r->classbook; - int q; - DECODE(q,f,c); - if (q == EOP) goto done; - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - part_classdata[0][class_set] = r->classdata[q]; - #else - for (i=classwords-1; i >= 0; --i) { - classifications[0][i+pcount] = q % r->classifications; - q /= r->classifications; - } - #endif - } - stb_prof(5); - for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { - int z = r->begin + pcount*r->part_size; - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - int c = part_classdata[0][class_set][i]; - #else - int c = classifications[0][pcount]; - #endif - int b = r->residue_books[c][pass]; - if (b >= 0) { - Codebook *book = f->codebooks + b; - stb_prof(20); // accounts for X time - #ifdef STB_VORBIS_DIVIDES_IN_CODEBOOK - if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) - goto done; - #else - // saves 1% - if (!codebook_decode_deinterleave_repeat_2(f, book, residue_buffers, &c_inter, &p_inter, n, r->part_size)) - goto done; - #endif - stb_prof(7); - } else { - z += r->part_size; - c_inter = z & 1; - p_inter = z >> 1; - } - } - stb_prof(8); - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - ++class_set; - #endif - } - } else if (ch == 1) { - while (pcount < part_read) { - int z = r->begin + pcount*r->part_size; - int c_inter = 0, p_inter = z; - if (pass == 0) { - Codebook *c = f->codebooks+r->classbook; - int q; - DECODE(q,f,c); - if (q == EOP) goto done; - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - part_classdata[0][class_set] = r->classdata[q]; - #else - for (i=classwords-1; i >= 0; --i) { - classifications[0][i+pcount] = q % r->classifications; - q /= r->classifications; - } - #endif - } - for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { - int z = r->begin + pcount*r->part_size; - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - int c = part_classdata[0][class_set][i]; - #else - int c = classifications[0][pcount]; - #endif - int b = r->residue_books[c][pass]; - if (b >= 0) { - Codebook *book = f->codebooks + b; - stb_prof(22); - if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) - goto done; - stb_prof(3); - } else { - z += r->part_size; - c_inter = 0; - p_inter = z; - } - } - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - ++class_set; - #endif - } - } else { - while (pcount < part_read) { - int z = r->begin + pcount*r->part_size; - int c_inter = z % ch, p_inter = z/ch; - if (pass == 0) { - Codebook *c = f->codebooks+r->classbook; - int q; - DECODE(q,f,c); - if (q == EOP) goto done; - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - part_classdata[0][class_set] = r->classdata[q]; - #else - for (i=classwords-1; i >= 0; --i) { - classifications[0][i+pcount] = q % r->classifications; - q /= r->classifications; - } - #endif - } - for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { - int z = r->begin + pcount*r->part_size; - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - int c = part_classdata[0][class_set][i]; - #else - int c = classifications[0][pcount]; - #endif - int b = r->residue_books[c][pass]; - if (b >= 0) { - Codebook *book = f->codebooks + b; - stb_prof(22); - if (!codebook_decode_deinterleave_repeat(f, book, residue_buffers, ch, &c_inter, &p_inter, n, r->part_size)) - goto done; - stb_prof(3); - } else { - z += r->part_size; - c_inter = z % ch; - p_inter = z / ch; - } - } - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - ++class_set; - #endif - } - } - } - goto done; - } - stb_prof(9); - - for (pass=0; pass < 8; ++pass) { - int pcount = 0, class_set=0; - while (pcount < part_read) { - if (pass == 0) { - for (j=0; j < ch; ++j) { - if (!do_not_decode[j]) { - Codebook *c = f->codebooks+r->classbook; - int temp; - DECODE(temp,f,c); - if (temp == EOP) goto done; - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - part_classdata[j][class_set] = r->classdata[temp]; - #else - for (i=classwords-1; i >= 0; --i) { - classifications[j][i+pcount] = temp % r->classifications; - temp /= r->classifications; - } - #endif - } - } - } - for (i=0; i < classwords && pcount < part_read; ++i, ++pcount) { - for (j=0; j < ch; ++j) { - if (!do_not_decode[j]) { - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - int c = part_classdata[j][class_set][i]; - #else - int c = classifications[j][pcount]; - #endif - int b = r->residue_books[c][pass]; - if (b >= 0) { - float *target = residue_buffers[j]; - int offset = r->begin + pcount * r->part_size; - int n = r->part_size; - Codebook *book = f->codebooks + b; - if (!residue_decode(f, book, target, offset, n, rtype)) - goto done; - } - } - } - } - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - ++class_set; - #endif - } - } - done: - stb_prof(0); - temp_alloc_restore(f,temp_alloc_point); -} - - -#if 0 -// slow way for debugging -void inverse_mdct_slow(float *buffer, int n) -{ - int i,j; - int n2 = n >> 1; - float *x = (float *) malloc(sizeof(*x) * n2); - memcpy(x, buffer, sizeof(*x) * n2); - for (i=0; i < n; ++i) { - float acc = 0; - for (j=0; j < n2; ++j) - // formula from paper: - //acc += n/4.0f * x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1)); - // formula from wikipedia - //acc += 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5)); - // these are equivalent, except the formula from the paper inverts the multiplier! - // however, what actually works is NO MULTIPLIER!?! - //acc += 64 * 2.0f / n2 * x[j] * (float) cos(M_PI/n2 * (i + 0.5 + n2/2)*(j + 0.5)); - acc += x[j] * (float) cos(M_PI / 2 / n * (2 * i + 1 + n/2.0)*(2*j+1)); - buffer[i] = acc; - } - free(x); -} -#elif 0 -// same as above, but just barely able to run in real time on modern machines -void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype) -{ - float mcos[16384]; - int i,j; - int n2 = n >> 1, nmask = (n << 2) -1; - float *x = (float *) malloc(sizeof(*x) * n2); - memcpy(x, buffer, sizeof(*x) * n2); - for (i=0; i < 4*n; ++i) - mcos[i] = (float) cos(M_PI / 2 * i / n); - - for (i=0; i < n; ++i) { - float acc = 0; - for (j=0; j < n2; ++j) - acc += x[j] * mcos[(2 * i + 1 + n2)*(2*j+1) & nmask]; - buffer[i] = acc; - } - free(x); -} -#else -// transform to use a slow dct-iv; this is STILL basically trivial, -// but only requires half as many ops -void dct_iv_slow(float *buffer, int n) -{ - float mcos[16384]; - float x[2048]; - int i,j; - int n2 = n >> 1, nmask = (n << 3) - 1; - memcpy(x, buffer, sizeof(*x) * n); - for (i=0; i < 8*n; ++i) - mcos[i] = (float) cos(M_PI / 4 * i / n); - for (i=0; i < n; ++i) { - float acc = 0; - for (j=0; j < n; ++j) - acc += x[j] * mcos[((2 * i + 1)*(2*j+1)) & nmask]; - //acc += x[j] * cos(M_PI / n * (i + 0.5) * (j + 0.5)); - buffer[i] = acc; - } -} - -void inverse_mdct_slow(float *buffer, int n, vorb *f, int blocktype) -{ - int i, n4 = n >> 2, n2 = n >> 1, n3_4 = n - n4; - float temp[4096]; - - memcpy(temp, buffer, n2 * sizeof(float)); - dct_iv_slow(temp, n2); // returns -c'-d, a-b' - - for (i=0; i < n4 ; ++i) buffer[i] = temp[i+n4]; // a-b' - for ( ; i < n3_4; ++i) buffer[i] = -temp[n3_4 - i - 1]; // b-a', c+d' - for ( ; i < n ; ++i) buffer[i] = -temp[i - n3_4]; // c'+d -} -#endif - -#ifndef LIBVORBIS_MDCT -#define LIBVORBIS_MDCT 0 -#endif - -#if LIBVORBIS_MDCT -// directly call the vorbis MDCT using an interface documented -// by Jeff Roberts... useful for performance comparison -typedef struct -{ - int n; - int log2n; - - float *trig; - int *bitrev; - - float scale; -} mdct_lookup; - -extern void mdct_init(mdct_lookup *lookup, int n); -extern void mdct_clear(mdct_lookup *l); -extern void mdct_backward(mdct_lookup *init, float *in, float *out); - -mdct_lookup M1,M2; - -void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) -{ - mdct_lookup *M; - if (M1.n == n) M = &M1; - else if (M2.n == n) M = &M2; - else if (M1.n == 0) { mdct_init(&M1, n); M = &M1; } - else { - if (M2.n) __asm int 3; - mdct_init(&M2, n); - M = &M2; - } - - mdct_backward(M, buffer, buffer); -} -#endif - - -// the following were split out into separate functions while optimizing; -// they could be pushed back up but eh. __forceinline showed no change; -// they're probably already being inlined. -static void imdct_step3_iter0_loop(int n, float *e, int i_off, int k_off, float *A) -{ - float *ee0 = e + i_off; - float *ee2 = ee0 + k_off; - int i; - - assert((n & 3) == 0); - for (i=(n>>2); i > 0; --i) { - float k00_20, k01_21; - k00_20 = ee0[ 0] - ee2[ 0]; - k01_21 = ee0[-1] - ee2[-1]; - ee0[ 0] += ee2[ 0];//ee0[ 0] = ee0[ 0] + ee2[ 0]; - ee0[-1] += ee2[-1];//ee0[-1] = ee0[-1] + ee2[-1]; - ee2[ 0] = k00_20 * A[0] - k01_21 * A[1]; - ee2[-1] = k01_21 * A[0] + k00_20 * A[1]; - A += 8; - - k00_20 = ee0[-2] - ee2[-2]; - k01_21 = ee0[-3] - ee2[-3]; - ee0[-2] += ee2[-2];//ee0[-2] = ee0[-2] + ee2[-2]; - ee0[-3] += ee2[-3];//ee0[-3] = ee0[-3] + ee2[-3]; - ee2[-2] = k00_20 * A[0] - k01_21 * A[1]; - ee2[-3] = k01_21 * A[0] + k00_20 * A[1]; - A += 8; - - k00_20 = ee0[-4] - ee2[-4]; - k01_21 = ee0[-5] - ee2[-5]; - ee0[-4] += ee2[-4];//ee0[-4] = ee0[-4] + ee2[-4]; - ee0[-5] += ee2[-5];//ee0[-5] = ee0[-5] + ee2[-5]; - ee2[-4] = k00_20 * A[0] - k01_21 * A[1]; - ee2[-5] = k01_21 * A[0] + k00_20 * A[1]; - A += 8; - - k00_20 = ee0[-6] - ee2[-6]; - k01_21 = ee0[-7] - ee2[-7]; - ee0[-6] += ee2[-6];//ee0[-6] = ee0[-6] + ee2[-6]; - ee0[-7] += ee2[-7];//ee0[-7] = ee0[-7] + ee2[-7]; - ee2[-6] = k00_20 * A[0] - k01_21 * A[1]; - ee2[-7] = k01_21 * A[0] + k00_20 * A[1]; - A += 8; - ee0 -= 8; - ee2 -= 8; - } -} - -static void imdct_step3_inner_r_loop(int lim, float *e, int d0, int k_off, float *A, int k1) -{ - int i; - float k00_20, k01_21; - - float *e0 = e + d0; - float *e2 = e0 + k_off; - - for (i=lim >> 2; i > 0; --i) { - k00_20 = e0[-0] - e2[-0]; - k01_21 = e0[-1] - e2[-1]; - e0[-0] += e2[-0];//e0[-0] = e0[-0] + e2[-0]; - e0[-1] += e2[-1];//e0[-1] = e0[-1] + e2[-1]; - e2[-0] = (k00_20)*A[0] - (k01_21) * A[1]; - e2[-1] = (k01_21)*A[0] + (k00_20) * A[1]; - - A += k1; - - k00_20 = e0[-2] - e2[-2]; - k01_21 = e0[-3] - e2[-3]; - e0[-2] += e2[-2];//e0[-2] = e0[-2] + e2[-2]; - e0[-3] += e2[-3];//e0[-3] = e0[-3] + e2[-3]; - e2[-2] = (k00_20)*A[0] - (k01_21) * A[1]; - e2[-3] = (k01_21)*A[0] + (k00_20) * A[1]; - - A += k1; - - k00_20 = e0[-4] - e2[-4]; - k01_21 = e0[-5] - e2[-5]; - e0[-4] += e2[-4];//e0[-4] = e0[-4] + e2[-4]; - e0[-5] += e2[-5];//e0[-5] = e0[-5] + e2[-5]; - e2[-4] = (k00_20)*A[0] - (k01_21) * A[1]; - e2[-5] = (k01_21)*A[0] + (k00_20) * A[1]; - - A += k1; - - k00_20 = e0[-6] - e2[-6]; - k01_21 = e0[-7] - e2[-7]; - e0[-6] += e2[-6];//e0[-6] = e0[-6] + e2[-6]; - e0[-7] += e2[-7];//e0[-7] = e0[-7] + e2[-7]; - e2[-6] = (k00_20)*A[0] - (k01_21) * A[1]; - e2[-7] = (k01_21)*A[0] + (k00_20) * A[1]; - - e0 -= 8; - e2 -= 8; - - A += k1; - } -} - -static void imdct_step3_inner_s_loop(int n, float *e, int i_off, int k_off, float *A, int a_off, int k0) -{ - int i; - float A0 = A[0]; - float A1 = A[0+1]; - float A2 = A[0+a_off]; - float A3 = A[0+a_off+1]; - float A4 = A[0+a_off*2+0]; - float A5 = A[0+a_off*2+1]; - float A6 = A[0+a_off*3+0]; - float A7 = A[0+a_off*3+1]; - - float k00,k11; - - float *ee0 = e +i_off; - float *ee2 = ee0+k_off; - - for (i=n; i > 0; --i) { - k00 = ee0[ 0] - ee2[ 0]; - k11 = ee0[-1] - ee2[-1]; - ee0[ 0] = ee0[ 0] + ee2[ 0]; - ee0[-1] = ee0[-1] + ee2[-1]; - ee2[ 0] = (k00) * A0 - (k11) * A1; - ee2[-1] = (k11) * A0 + (k00) * A1; - - k00 = ee0[-2] - ee2[-2]; - k11 = ee0[-3] - ee2[-3]; - ee0[-2] = ee0[-2] + ee2[-2]; - ee0[-3] = ee0[-3] + ee2[-3]; - ee2[-2] = (k00) * A2 - (k11) * A3; - ee2[-3] = (k11) * A2 + (k00) * A3; - - k00 = ee0[-4] - ee2[-4]; - k11 = ee0[-5] - ee2[-5]; - ee0[-4] = ee0[-4] + ee2[-4]; - ee0[-5] = ee0[-5] + ee2[-5]; - ee2[-4] = (k00) * A4 - (k11) * A5; - ee2[-5] = (k11) * A4 + (k00) * A5; - - k00 = ee0[-6] - ee2[-6]; - k11 = ee0[-7] - ee2[-7]; - ee0[-6] = ee0[-6] + ee2[-6]; - ee0[-7] = ee0[-7] + ee2[-7]; - ee2[-6] = (k00) * A6 - (k11) * A7; - ee2[-7] = (k11) * A6 + (k00) * A7; - - ee0 -= k0; - ee2 -= k0; - } -} - -static __forceinline void iter_54(float *z) -{ - float k00,k11,k22,k33; - float y0,y1,y2,y3; - - k00 = z[ 0] - z[-4]; - y0 = z[ 0] + z[-4]; - y2 = z[-2] + z[-6]; - k22 = z[-2] - z[-6]; - - z[-0] = y0 + y2; // z0 + z4 + z2 + z6 - z[-2] = y0 - y2; // z0 + z4 - z2 - z6 - - // done with y0,y2 - - k33 = z[-3] - z[-7]; - - z[-4] = k00 + k33; // z0 - z4 + z3 - z7 - z[-6] = k00 - k33; // z0 - z4 - z3 + z7 - - // done with k33 - - k11 = z[-1] - z[-5]; - y1 = z[-1] + z[-5]; - y3 = z[-3] + z[-7]; - - z[-1] = y1 + y3; // z1 + z5 + z3 + z7 - z[-3] = y1 - y3; // z1 + z5 - z3 - z7 - z[-5] = k11 - k22; // z1 - z5 + z2 - z6 - z[-7] = k11 + k22; // z1 - z5 - z2 + z6 -} - -static void imdct_step3_inner_s_loop_ld654(int n, float *e, int i_off, float *A, int base_n) -{ - int k_off = -8; - int a_off = base_n >> 3; - float A2 = A[0+a_off]; - float *z = e + i_off; - float *base = z - 16 * n; - - while (z > base) { - float k00,k11; - - k00 = z[-0] - z[-8]; - k11 = z[-1] - z[-9]; - z[-0] = z[-0] + z[-8]; - z[-1] = z[-1] + z[-9]; - z[-8] = k00; - z[-9] = k11 ; - - k00 = z[ -2] - z[-10]; - k11 = z[ -3] - z[-11]; - z[ -2] = z[ -2] + z[-10]; - z[ -3] = z[ -3] + z[-11]; - z[-10] = (k00+k11) * A2; - z[-11] = (k11-k00) * A2; - - k00 = z[-12] - z[ -4]; // reverse to avoid a unary negation - k11 = z[ -5] - z[-13]; - z[ -4] = z[ -4] + z[-12]; - z[ -5] = z[ -5] + z[-13]; - z[-12] = k11; - z[-13] = k00; - - k00 = z[-14] - z[ -6]; // reverse to avoid a unary negation - k11 = z[ -7] - z[-15]; - z[ -6] = z[ -6] + z[-14]; - z[ -7] = z[ -7] + z[-15]; - z[-14] = (k00+k11) * A2; - z[-15] = (k00-k11) * A2; - - iter_54(z); - iter_54(z-8); - z -= 16; - } -} - -static void inverse_mdct(float *buffer, int n, vorb *f, int blocktype) -{ - int n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l; - int n3_4 = n - n4, ld; - // @OPTIMIZE: reduce register pressure by using fewer variables? - int save_point = temp_alloc_save(f); - float *buf2 = (float *) temp_alloc(f, n2 * sizeof(*buf2)); - float *u=NULL,*v=NULL; - // twiddle factors - float *A = f->A[blocktype]; - - // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio" - // See notes about bugs in that paper in less-optimal implementation 'inverse_mdct_old' after this function. - - // kernel from paper - - - // merged: - // copy and reflect spectral data - // step 0 - - // note that it turns out that the items added together during - // this step are, in fact, being added to themselves (as reflected - // by step 0). inexplicable inefficiency! this became obvious - // once I combined the passes. - - // so there's a missing 'times 2' here (for adding X to itself). - // this propogates through linearly to the end, where the numbers - // are 1/2 too small, and need to be compensated for. - - { - float *d,*e, *AA, *e_stop; - d = &buf2[n2-2]; - AA = A; - e = &buffer[0]; - e_stop = &buffer[n2]; - while (e != e_stop) { - d[1] = (e[0] * AA[0] - e[2]*AA[1]); - d[0] = (e[0] * AA[1] + e[2]*AA[0]); - d -= 2; - AA += 2; - e += 4; - } - - e = &buffer[n2-3]; - while (d >= buf2) { - d[1] = (-e[2] * AA[0] - -e[0]*AA[1]); - d[0] = (-e[2] * AA[1] + -e[0]*AA[0]); - d -= 2; - AA += 2; - e -= 4; - } - } - - // now we use symbolic names for these, so that we can - // possibly swap their meaning as we change which operations - // are in place - - u = buffer; - v = buf2; - - // step 2 (paper output is w, now u) - // this could be in place, but the data ends up in the wrong - // place... _somebody_'s got to swap it, so this is nominated - { - float *AA = &A[n2-8]; - float *d0,*d1, *e0, *e1; - - e0 = &v[n4]; - e1 = &v[0]; - - d0 = &u[n4]; - d1 = &u[0]; - - while (AA >= A) { - float v40_20, v41_21; - - v41_21 = e0[1] - e1[1]; - v40_20 = e0[0] - e1[0]; - d0[1] = e0[1] + e1[1]; - d0[0] = e0[0] + e1[0]; - d1[1] = v41_21*AA[4] - v40_20*AA[5]; - d1[0] = v40_20*AA[4] + v41_21*AA[5]; - - v41_21 = e0[3] - e1[3]; - v40_20 = e0[2] - e1[2]; - d0[3] = e0[3] + e1[3]; - d0[2] = e0[2] + e1[2]; - d1[3] = v41_21*AA[0] - v40_20*AA[1]; - d1[2] = v40_20*AA[0] + v41_21*AA[1]; - - AA -= 8; - - d0 += 4; - d1 += 4; - e0 += 4; - e1 += 4; - } - } - - // step 3 - ld = ilog(n) - 1; // ilog is off-by-one from normal definitions - - // optimized step 3: - - // the original step3 loop can be nested r inside s or s inside r; - // it's written originally as s inside r, but this is dumb when r - // iterates many times, and s few. So I have two copies of it and - // switch between them halfway. - - // this is iteration 0 of step 3 - imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*0, -(n >> 3), A); - imdct_step3_iter0_loop(n >> 4, u, n2-1-n4*1, -(n >> 3), A); - - // this is iteration 1 of step 3 - imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*0, -(n >> 4), A, 16); - imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*1, -(n >> 4), A, 16); - imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*2, -(n >> 4), A, 16); - imdct_step3_inner_r_loop(n >> 5, u, n2-1 - n8*3, -(n >> 4), A, 16); - - l=2; - for (; l < (ld-3)>>1; ++l) { - int k0 = n >> (l+2), k0_2 = k0>>1; - int lim = 1 << (l+1); - int i; - for (i=0; i < lim; ++i) - imdct_step3_inner_r_loop(n >> (l+4), u, n2-1 - k0*i, -k0_2, A, 1 << (l+3)); - } - - for (; l < ld-6; ++l) { - int k0 = n >> (l+2), k1 = 1 << (l+3), k0_2 = k0>>1; - int rlim = n >> (l+6), r; - int lim = 1 << (l+1); - int i_off; - float *A0 = A; - i_off = n2-1; - for (r=rlim; r > 0; --r) { - imdct_step3_inner_s_loop(lim, u, i_off, -k0_2, A0, k1, k0); - A0 += k1*4; - i_off -= 8; - } - } - - // iterations with count: - // ld-6,-5,-4 all interleaved together - // the big win comes from getting rid of needless flops - // due to the constants on pass 5 & 4 being all 1 and 0; - // combining them to be simultaneous to improve cache made little difference - imdct_step3_inner_s_loop_ld654(n >> 5, u, n2-1, A, n); - - // output is u - - // step 4, 5, and 6 - // cannot be in-place because of step 5 - { - uint16 *bitrev = f->bit_reverse[blocktype]; - // weirdly, I'd have thought reading sequentially and writing - // erratically would have been better than vice-versa, but in - // fact that's not what my testing showed. (That is, with - // j = bitreverse(i), do you read i and write j, or read j and write i.) - - float *d0 = &v[n4-4]; - float *d1 = &v[n2-4]; - while (d0 >= v) { - int k4; - - k4 = bitrev[0]; - d1[3] = u[k4+0]; - d1[2] = u[k4+1]; - d0[3] = u[k4+2]; - d0[2] = u[k4+3]; - - k4 = bitrev[1]; - d1[1] = u[k4+0]; - d1[0] = u[k4+1]; - d0[1] = u[k4+2]; - d0[0] = u[k4+3]; - - d0 -= 4; - d1 -= 4; - bitrev += 2; - } - } - // (paper output is u, now v) - - - // data must be in buf2 - assert(v == buf2); - - // step 7 (paper output is v, now v) - // this is now in place - { - float *C = f->C[blocktype]; - float *d, *e; - - d = v; - e = v + n2 - 4; - - while (d < e) { - float a02,a11,b0,b1,b2,b3; - - a02 = d[0] - e[2]; - a11 = d[1] + e[3]; - - b0 = C[1]*a02 + C[0]*a11; - b1 = C[1]*a11 - C[0]*a02; - - b2 = d[0] + e[ 2]; - b3 = d[1] - e[ 3]; - - d[0] = b2 + b0; - d[1] = b3 + b1; - e[2] = b2 - b0; - e[3] = b1 - b3; - - a02 = d[2] - e[0]; - a11 = d[3] + e[1]; - - b0 = C[3]*a02 + C[2]*a11; - b1 = C[3]*a11 - C[2]*a02; - - b2 = d[2] + e[ 0]; - b3 = d[3] - e[ 1]; - - d[2] = b2 + b0; - d[3] = b3 + b1; - e[0] = b2 - b0; - e[1] = b1 - b3; - - C += 4; - d += 4; - e -= 4; - } - } - - // data must be in buf2 - - - // step 8+decode (paper output is X, now buffer) - // this generates pairs of data a la 8 and pushes them directly through - // the decode kernel (pushing rather than pulling) to avoid having - // to make another pass later - - // this cannot POSSIBLY be in place, so we refer to the buffers directly - - { - float *d0,*d1,*d2,*d3; - - float *B = f->B[blocktype] + n2 - 8; - float *e = buf2 + n2 - 8; - d0 = &buffer[0]; - d1 = &buffer[n2-4]; - d2 = &buffer[n2]; - d3 = &buffer[n-4]; - while (e >= v) { - float p0,p1,p2,p3; - - p3 = e[6]*B[7] - e[7]*B[6]; - p2 = -e[6]*B[6] - e[7]*B[7]; - - d0[0] = p3; - d1[3] = - p3; - d2[0] = p2; - d3[3] = p2; - - p1 = e[4]*B[5] - e[5]*B[4]; - p0 = -e[4]*B[4] - e[5]*B[5]; - - d0[1] = p1; - d1[2] = - p1; - d2[1] = p0; - d3[2] = p0; - - p3 = e[2]*B[3] - e[3]*B[2]; - p2 = -e[2]*B[2] - e[3]*B[3]; - - d0[2] = p3; - d1[1] = - p3; - d2[2] = p2; - d3[1] = p2; - - p1 = e[0]*B[1] - e[1]*B[0]; - p0 = -e[0]*B[0] - e[1]*B[1]; - - d0[3] = p1; - d1[0] = - p1; - d2[3] = p0; - d3[0] = p0; - - B -= 8; - e -= 8; - d0 += 4; - d2 += 4; - d1 -= 4; - d3 -= 4; - } - } - - temp_alloc_restore(f,save_point); -} - -#if 0 -// this is the original version of the above code, if you want to optimize it from scratch -void inverse_mdct_naive(float *buffer, int n) -{ - float s; - float A[1 << 12], B[1 << 12], C[1 << 11]; - int i,k,k2,k4, n2 = n >> 1, n4 = n >> 2, n8 = n >> 3, l; - int n3_4 = n - n4, ld; - // how can they claim this only uses N words?! - // oh, because they're only used sparsely, whoops - float u[1 << 13], X[1 << 13], v[1 << 13], w[1 << 13]; - // set up twiddle factors - - for (k=k2=0; k < n4; ++k,k2+=2) { - A[k2 ] = (float) cos(4*k*M_PI/n); - A[k2+1] = (float) -sin(4*k*M_PI/n); - B[k2 ] = (float) cos((k2+1)*M_PI/n/2); - B[k2+1] = (float) sin((k2+1)*M_PI/n/2); - } - for (k=k2=0; k < n8; ++k,k2+=2) { - C[k2 ] = (float) cos(2*(k2+1)*M_PI/n); - C[k2+1] = (float) -sin(2*(k2+1)*M_PI/n); - } - - // IMDCT algorithm from "The use of multirate filter banks for coding of high quality digital audio" - // Note there are bugs in that pseudocode, presumably due to them attempting - // to rename the arrays nicely rather than representing the way their actual - // implementation bounces buffers back and forth. As a result, even in the - // "some formulars corrected" version, a direct implementation fails. These - // are noted below as "paper bug". - - // copy and reflect spectral data - for (k=0; k < n2; ++k) u[k] = buffer[k]; - for ( ; k < n ; ++k) u[k] = -buffer[n - k - 1]; - // kernel from paper - // step 1 - for (k=k2=k4=0; k < n4; k+=1, k2+=2, k4+=4) { - v[n-k4-1] = (u[k4] - u[n-k4-1]) * A[k2] - (u[k4+2] - u[n-k4-3])*A[k2+1]; - v[n-k4-3] = (u[k4] - u[n-k4-1]) * A[k2+1] + (u[k4+2] - u[n-k4-3])*A[k2]; - } - // step 2 - for (k=k4=0; k < n8; k+=1, k4+=4) { - w[n2+3+k4] = v[n2+3+k4] + v[k4+3]; - w[n2+1+k4] = v[n2+1+k4] + v[k4+1]; - w[k4+3] = (v[n2+3+k4] - v[k4+3])*A[n2-4-k4] - (v[n2+1+k4]-v[k4+1])*A[n2-3-k4]; - w[k4+1] = (v[n2+1+k4] - v[k4+1])*A[n2-4-k4] + (v[n2+3+k4]-v[k4+3])*A[n2-3-k4]; - } - // step 3 - ld = ilog(n) - 1; // ilog is off-by-one from normal definitions - for (l=0; l < ld-3; ++l) { - int k0 = n >> (l+2), k1 = 1 << (l+3); - int rlim = n >> (l+4), r4, r; - int s2lim = 1 << (l+2), s2; - for (r=r4=0; r < rlim; r4+=4,++r) { - for (s2=0; s2 < s2lim; s2+=2) { - u[n-1-k0*s2-r4] = w[n-1-k0*s2-r4] + w[n-1-k0*(s2+1)-r4]; - u[n-3-k0*s2-r4] = w[n-3-k0*s2-r4] + w[n-3-k0*(s2+1)-r4]; - u[n-1-k0*(s2+1)-r4] = (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1] - - (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1+1]; - u[n-3-k0*(s2+1)-r4] = (w[n-3-k0*s2-r4] - w[n-3-k0*(s2+1)-r4]) * A[r*k1] - + (w[n-1-k0*s2-r4] - w[n-1-k0*(s2+1)-r4]) * A[r*k1+1]; - } - } - if (l+1 < ld-3) { - // paper bug: ping-ponging of u&w here is omitted - memcpy(w, u, sizeof(u)); - } - } - - // step 4 - for (i=0; i < n8; ++i) { - int j = bit_reverse(i) >> (32-ld+3); - assert(j < n8); - if (i == j) { - // paper bug: original code probably swapped in place; if copying, - // need to directly copy in this case - int i8 = i << 3; - v[i8+1] = u[i8+1]; - v[i8+3] = u[i8+3]; - v[i8+5] = u[i8+5]; - v[i8+7] = u[i8+7]; - } else if (i < j) { - int i8 = i << 3, j8 = j << 3; - v[j8+1] = u[i8+1], v[i8+1] = u[j8 + 1]; - v[j8+3] = u[i8+3], v[i8+3] = u[j8 + 3]; - v[j8+5] = u[i8+5], v[i8+5] = u[j8 + 5]; - v[j8+7] = u[i8+7], v[i8+7] = u[j8 + 7]; - } - } - // step 5 - for (k=0; k < n2; ++k) { - w[k] = v[k*2+1]; - } - // step 6 - for (k=k2=k4=0; k < n8; ++k, k2 += 2, k4 += 4) { - u[n-1-k2] = w[k4]; - u[n-2-k2] = w[k4+1]; - u[n3_4 - 1 - k2] = w[k4+2]; - u[n3_4 - 2 - k2] = w[k4+3]; - } - // step 7 - for (k=k2=0; k < n8; ++k, k2 += 2) { - v[n2 + k2 ] = ( u[n2 + k2] + u[n-2-k2] + C[k2+1]*(u[n2+k2]-u[n-2-k2]) + C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2; - v[n-2 - k2] = ( u[n2 + k2] + u[n-2-k2] - C[k2+1]*(u[n2+k2]-u[n-2-k2]) - C[k2]*(u[n2+k2+1]+u[n-2-k2+1]))/2; - v[n2+1+ k2] = ( u[n2+1+k2] - u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2; - v[n-1 - k2] = (-u[n2+1+k2] + u[n-1-k2] + C[k2+1]*(u[n2+1+k2]+u[n-1-k2]) - C[k2]*(u[n2+k2]-u[n-2-k2]))/2; - } - // step 8 - for (k=k2=0; k < n4; ++k,k2 += 2) { - X[k] = v[k2+n2]*B[k2 ] + v[k2+1+n2]*B[k2+1]; - X[n2-1-k] = v[k2+n2]*B[k2+1] - v[k2+1+n2]*B[k2 ]; - } - - // decode kernel to output - // determined the following value experimentally - // (by first figuring out what made inverse_mdct_slow work); then matching that here - // (probably vorbis encoder premultiplies by n or n/2, to save it on the decoder?) - s = 0.5; // theoretically would be n4 - - // [[[ note! the s value of 0.5 is compensated for by the B[] in the current code, - // so it needs to use the "old" B values to behave correctly, or else - // set s to 1.0 ]]] - for (i=0; i < n4 ; ++i) buffer[i] = s * X[i+n4]; - for ( ; i < n3_4; ++i) buffer[i] = -s * X[n3_4 - i - 1]; - for ( ; i < n ; ++i) buffer[i] = -s * X[i - n3_4]; -} -#endif - -static float *get_window(vorb *f, int len) -{ - len <<= 1; - if (len == f->blocksize_0) return f->window[0]; - if (len == f->blocksize_1) return f->window[1]; - assert(0); - return NULL; -} - -#ifndef STB_VORBIS_NO_DEFER_FLOOR -typedef int16 YTYPE; -#else -typedef int YTYPE; -#endif -static int do_floor(vorb *f, Mapping *map, int i, int n, float *target, YTYPE *finalY, uint8 *step2_flag) -{ - int n2 = n >> 1; - int s = map->chan[i].mux, floor; - floor = map->submap_floor[s]; - if (f->floor_types[floor] == 0) { - return error(f, VORBIS_invalid_stream); - } else { - Floor1 *g = &f->floor_config[floor].floor1; - int j,q; - int lx = 0, ly = finalY[0] * g->floor1_multiplier; - for (q=1; q < g->values; ++q) { - j = g->sorted_order[q]; - #ifndef STB_VORBIS_NO_DEFER_FLOOR - if (finalY[j] >= 0) - #else - if (step2_flag[j]) - #endif - { - int hy = finalY[j] * g->floor1_multiplier; - int hx = g->Xlist[j]; - draw_line(target, lx,ly, hx,hy, n2); - lx = hx, ly = hy; - } - } - if (lx < n2) - // optimization of: draw_line(target, lx,ly, n,ly, n2); - for (j=lx; j < n2; ++j) - LINE_OP(target[j], inverse_db_table[ly]); - } - return TRUE; -} - -static int vorbis_decode_initial(vorb *f, int *p_left_start, int *p_left_end, int *p_right_start, int *p_right_end, int *mode) -{ - Mode *m; - int i, n, prev, next, window_center; - f->channel_buffer_start = f->channel_buffer_end = 0; - - retry: - if (f->eof) return FALSE; - if (!maybe_start_packet(f)) - return FALSE; - // check packet type - if (get_bits(f,1) != 0) { - if (IS_PUSH_MODE(f)) - return error(f,VORBIS_bad_packet_type); - while (EOP != get8_packet(f)); - goto retry; - } - - if (f->alloc.alloc_buffer) - assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); - - i = get_bits(f, ilog(f->mode_count-1)); - if (i == EOP) return FALSE; - if (i >= f->mode_count) return FALSE; - *mode = i; - m = f->mode_config + i; - if (m->blockflag) { - n = f->blocksize_1; - prev = get_bits(f,1); - next = get_bits(f,1); - } else { - prev = next = 0; - n = f->blocksize_0; - } - -// WINDOWING - - window_center = n >> 1; - if (m->blockflag && !prev) { - *p_left_start = (n - f->blocksize_0) >> 2; - *p_left_end = (n + f->blocksize_0) >> 2; - } else { - *p_left_start = 0; - *p_left_end = window_center; - } - if (m->blockflag && !next) { - *p_right_start = (n*3 - f->blocksize_0) >> 2; - *p_right_end = (n*3 + f->blocksize_0) >> 2; - } else { - *p_right_start = window_center; - *p_right_end = n; - } - return TRUE; -} - -static int vorbis_decode_packet_rest(vorb *f, int *len, Mode *m, int left_start, int left_end, int right_start, int right_end, int *p_left) -{ - Mapping *map; - int i,j,k,n,n2; - int zero_channel[256]; - int really_zero_channel[256]; - int window_center; - -// WINDOWING - - n = f->blocksize[m->blockflag]; - window_center = n >> 1; - - map = &f->mapping[m->mapping]; - -// FLOORS - n2 = n >> 1; - - stb_prof(1); - for (i=0; i < f->channels; ++i) { - int s = map->chan[i].mux, floor; - zero_channel[i] = FALSE; - floor = map->submap_floor[s]; - if (f->floor_types[floor] == 0) { - return error(f, VORBIS_invalid_stream); - } else { - Floor1 *g = &f->floor_config[floor].floor1; - if (get_bits(f, 1)) { - short *finalY; - uint8 step2_flag[256]; - static int range_list[4] = { 256, 128, 86, 64 }; - int range = range_list[g->floor1_multiplier-1]; - int offset = 2; - finalY = f->finalY[i]; - finalY[0] = get_bits(f, ilog(range)-1); - finalY[1] = get_bits(f, ilog(range)-1); - for (j=0; j < g->partitions; ++j) { - int pclass = g->partition_class_list[j]; - int cdim = g->class_dimensions[pclass]; - int cbits = g->class_subclasses[pclass]; - int csub = (1 << cbits)-1; - int cval = 0; - if (cbits) { - Codebook *c = f->codebooks + g->class_masterbooks[pclass]; - DECODE(cval,f,c); - } - for (k=0; k < cdim; ++k) { - int book = g->subclass_books[pclass][cval & csub]; - cval = cval >> cbits; - if (book >= 0) { - int temp; - Codebook *c = f->codebooks + book; - DECODE(temp,f,c); - finalY[offset++] = temp; - } else - finalY[offset++] = 0; - } - } - if (f->valid_bits == INVALID_BITS) goto error; // behavior according to spec - step2_flag[0] = step2_flag[1] = 1; - for (j=2; j < g->values; ++j) { - int low, high, pred, highroom, lowroom, room, val; - low = g->neighbors[j][0]; - high = g->neighbors[j][1]; - //neighbors(g->Xlist, j, &low, &high); - pred = predict_point(g->Xlist[j], g->Xlist[low], g->Xlist[high], finalY[low], finalY[high]); - val = finalY[j]; - highroom = range - pred; - lowroom = pred; - if (highroom < lowroom) - room = highroom * 2; - else - room = lowroom * 2; - if (val) { - step2_flag[low] = step2_flag[high] = 1; - step2_flag[j] = 1; - if (val >= room) - if (highroom > lowroom) - finalY[j] = val - lowroom + pred; - else - finalY[j] = pred - val + highroom - 1; - else - if (val & 1) - finalY[j] = pred - ((val+1)>>1); - else - finalY[j] = pred + (val>>1); - } else { - step2_flag[j] = 0; - finalY[j] = pred; - } - } - -#ifdef STB_VORBIS_NO_DEFER_FLOOR - do_floor(f, map, i, n, f->floor_buffers[i], finalY, step2_flag); -#else - // defer final floor computation until _after_ residue - for (j=0; j < g->values; ++j) { - if (!step2_flag[j]) - finalY[j] = -1; - } -#endif - } else { - error: - zero_channel[i] = TRUE; - } - // So we just defer everything else to later - - // at this point we've decoded the floor into buffer - } - } - stb_prof(0); - // at this point we've decoded all floors - - if (f->alloc.alloc_buffer) - assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); - - // re-enable coupled channels if necessary - memcpy(really_zero_channel, zero_channel, sizeof(really_zero_channel[0]) * f->channels); - for (i=0; i < map->coupling_steps; ++i) - if (!zero_channel[map->chan[i].magnitude] || !zero_channel[map->chan[i].angle]) { - zero_channel[map->chan[i].magnitude] = zero_channel[map->chan[i].angle] = FALSE; - } - -// RESIDUE DECODE - for (i=0; i < map->submaps; ++i) { - float *residue_buffers[STB_VORBIS_MAX_CHANNELS]; - int r,t; - uint8 do_not_decode[256]; - int ch = 0; - for (j=0; j < f->channels; ++j) { - if (map->chan[j].mux == i) { - if (zero_channel[j]) { - do_not_decode[ch] = TRUE; - residue_buffers[ch] = NULL; - } else { - do_not_decode[ch] = FALSE; - residue_buffers[ch] = f->channel_buffers[j]; - } - ++ch; - } - } - r = map->submap_residue[i]; - t = f->residue_types[r]; - decode_residue(f, residue_buffers, ch, n2, r, do_not_decode); - } - - if (f->alloc.alloc_buffer) - assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); - -// INVERSE COUPLING - stb_prof(14); - for (i = map->coupling_steps-1; i >= 0; --i) { - int n2 = n >> 1; - float *m = f->channel_buffers[map->chan[i].magnitude]; - float *a = f->channel_buffers[map->chan[i].angle ]; - for (j=0; j < n2; ++j) { - float a2,m2; - if (m[j] > 0) - if (a[j] > 0) - m2 = m[j], a2 = m[j] - a[j]; - else - a2 = m[j], m2 = m[j] + a[j]; - else - if (a[j] > 0) - m2 = m[j], a2 = m[j] + a[j]; - else - a2 = m[j], m2 = m[j] - a[j]; - m[j] = m2; - a[j] = a2; - } - } - - // finish decoding the floors -#ifndef STB_VORBIS_NO_DEFER_FLOOR - stb_prof(15); - for (i=0; i < f->channels; ++i) { - if (really_zero_channel[i]) { - memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2); - } else { - do_floor(f, map, i, n, f->channel_buffers[i], f->finalY[i], NULL); - } - } -#else - for (i=0; i < f->channels; ++i) { - if (really_zero_channel[i]) { - memset(f->channel_buffers[i], 0, sizeof(*f->channel_buffers[i]) * n2); - } else { - for (j=0; j < n2; ++j) - f->channel_buffers[i][j] *= f->floor_buffers[i][j]; - } - } -#endif - -// INVERSE MDCT - stb_prof(16); - for (i=0; i < f->channels; ++i) - inverse_mdct(f->channel_buffers[i], n, f, m->blockflag); - stb_prof(0); - - // this shouldn't be necessary, unless we exited on an error - // and want to flush to get to the next packet - flush_packet(f); - - if (f->first_decode) { - // assume we start so first non-discarded sample is sample 0 - // this isn't to spec, but spec would require us to read ahead - // and decode the size of all current frames--could be done, - // but presumably it's not a commonly used feature - f->current_loc = -n2; // start of first frame is positioned for discard - // we might have to discard samples "from" the next frame too, - // if we're lapping a large block then a small at the start? - f->discard_samples_deferred = n - right_end; - f->current_loc_valid = TRUE; - f->first_decode = FALSE; - } else if (f->discard_samples_deferred) { - left_start += f->discard_samples_deferred; - *p_left = left_start; - f->discard_samples_deferred = 0; - } else if (f->previous_length == 0 && f->current_loc_valid) { - // we're recovering from a seek... that means we're going to discard - // the samples from this packet even though we know our position from - // the last page header, so we need to update the position based on - // the discarded samples here - // but wait, the code below is going to add this in itself even - // on a discard, so we don't need to do it here... - } - - // check if we have ogg information about the sample # for this packet - if (f->last_seg_which == f->end_seg_with_known_loc) { - // if we have a valid current loc, and this is final: - if (f->current_loc_valid && (f->page_flag & PAGEFLAG_last_page)) { - uint32 current_end = f->known_loc_for_packet - (n-right_end); - // then let's infer the size of the (probably) short final frame - if (current_end < f->current_loc + right_end) { - if (current_end < f->current_loc) { - // negative truncation, that's impossible! - *len = 0; - } else { - *len = current_end - f->current_loc; - } - *len += left_start; - f->current_loc += *len; - return TRUE; - } - } - // otherwise, just set our sample loc - // guess that the ogg granule pos refers to the _middle_ of the - // last frame? - // set f->current_loc to the position of left_start - f->current_loc = f->known_loc_for_packet - (n2-left_start); - f->current_loc_valid = TRUE; - } - if (f->current_loc_valid) - f->current_loc += (right_start - left_start); - - if (f->alloc.alloc_buffer) - assert(f->alloc.alloc_buffer_length_in_bytes == f->temp_offset); - *len = right_end; // ignore samples after the window goes to 0 - return TRUE; -} - -static int vorbis_decode_packet(vorb *f, int *len, int *p_left, int *p_right) -{ - int mode, left_end, right_end; - if (!vorbis_decode_initial(f, p_left, &left_end, p_right, &right_end, &mode)) return 0; - return vorbis_decode_packet_rest(f, len, f->mode_config + mode, *p_left, left_end, *p_right, right_end, p_left); -} - -static int vorbis_finish_frame(stb_vorbis *f, int len, int left, int right) -{ - int prev,i,j; - // we use right&left (the start of the right- and left-window sin()-regions) - // to determine how much to return, rather than inferring from the rules - // (same result, clearer code); 'left' indicates where our sin() window - // starts, therefore where the previous window's right edge starts, and - // therefore where to start mixing from the previous buffer. 'right' - // indicates where our sin() ending-window starts, therefore that's where - // we start saving, and where our returned-data ends. - - // mixin from previous window - if (f->previous_length) { - int i,j, n = f->previous_length; - float *w = get_window(f, n); - for (i=0; i < f->channels; ++i) { - for (j=0; j < n; ++j) - f->channel_buffers[i][left+j] = - f->channel_buffers[i][left+j]*w[ j] + - f->previous_window[i][ j]*w[n-1-j]; - } - } - - prev = f->previous_length; - - // last half of this data becomes previous window - f->previous_length = len - right; - - // @OPTIMIZE: could avoid this copy by double-buffering the - // output (flipping previous_window with channel_buffers), but - // then previous_window would have to be 2x as large, and - // channel_buffers couldn't be temp mem (although they're NOT - // currently temp mem, they could be (unless we want to level - // performance by spreading out the computation)) - for (i=0; i < f->channels; ++i) - for (j=0; right+j < len; ++j) - f->previous_window[i][j] = f->channel_buffers[i][right+j]; - - if (!prev) - // there was no previous packet, so this data isn't valid... - // this isn't entirely true, only the would-have-overlapped data - // isn't valid, but this seems to be what the spec requires - return 0; - - // truncate a short frame - if (len < right) right = len; - - f->samples_output += right-left; - - return right - left; -} - -static void vorbis_pump_first_frame(stb_vorbis *f) -{ - int len, right, left; - if (vorbis_decode_packet(f, &len, &left, &right)) - vorbis_finish_frame(f, len, left, right); -} - -#ifndef STB_VORBIS_NO_PUSHDATA_API -static int is_whole_packet_present(stb_vorbis *f, int end_page) -{ - // make sure that we have the packet available before continuing... - // this requires a full ogg parse, but we know we can fetch from f->stream - - // instead of coding this out explicitly, we could save the current read state, - // read the next packet with get8() until end-of-packet, check f->eof, then - // reset the state? but that would be slower, esp. since we'd have over 256 bytes - // of state to restore (primarily the page segment table) - - int s = f->next_seg, first = TRUE; - uint8 *p = f->stream; - - if (s != -1) { // if we're not starting the packet with a 'continue on next page' flag - for (; s < f->segment_count; ++s) { - p += f->segments[s]; - if (f->segments[s] < 255) // stop at first short segment - break; - } - // either this continues, or it ends it... - if (end_page) - if (s < f->segment_count-1) return error(f, VORBIS_invalid_stream); - if (s == f->segment_count) - s = -1; // set 'crosses page' flag - if (p > f->stream_end) return error(f, VORBIS_need_more_data); - first = FALSE; - } - for (; s == -1;) { - uint8 *q; - int n; - - // check that we have the page header ready - if (p + 26 >= f->stream_end) return error(f, VORBIS_need_more_data); - // validate the page - if (memcmp(p, ogg_page_header, 4)) return error(f, VORBIS_invalid_stream); - if (p[4] != 0) return error(f, VORBIS_invalid_stream); - if (first) { // the first segment must NOT have 'continued_packet', later ones MUST - if (f->previous_length) - if ((p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream); - // if no previous length, we're resynching, so we can come in on a continued-packet, - // which we'll just drop - } else { - if (!(p[5] & PAGEFLAG_continued_packet)) return error(f, VORBIS_invalid_stream); - } - n = p[26]; // segment counts - q = p+27; // q points to segment table - p = q + n; // advance past header - // make sure we've read the segment table - if (p > f->stream_end) return error(f, VORBIS_need_more_data); - for (s=0; s < n; ++s) { - p += q[s]; - if (q[s] < 255) - break; - } - if (end_page) - if (s < n-1) return error(f, VORBIS_invalid_stream); - if (s == f->segment_count) - s = -1; // set 'crosses page' flag - if (p > f->stream_end) return error(f, VORBIS_need_more_data); - first = FALSE; - } - return TRUE; -} -#endif // !STB_VORBIS_NO_PUSHDATA_API - -static int start_decoder(vorb *f) -{ - uint8 header[6], x,y; - int len,i,j,k, max_submaps = 0; - int longest_floorlist=0; - - // first page, first packet - - if (!start_page(f)) return FALSE; - // validate page flag - if (!(f->page_flag & PAGEFLAG_first_page)) return error(f, VORBIS_invalid_first_page); - if (f->page_flag & PAGEFLAG_last_page) return error(f, VORBIS_invalid_first_page); - if (f->page_flag & PAGEFLAG_continued_packet) return error(f, VORBIS_invalid_first_page); - // check for expected packet length - if (f->segment_count != 1) return error(f, VORBIS_invalid_first_page); - if (f->segments[0] != 30) return error(f, VORBIS_invalid_first_page); - // read packet - // check packet header - if (get8(f) != VORBIS_packet_id) return error(f, VORBIS_invalid_first_page); - if (!getn(f, header, 6)) return error(f, VORBIS_unexpected_eof); - if (!vorbis_validate(header)) return error(f, VORBIS_invalid_first_page); - // vorbis_version - if (get32(f) != 0) return error(f, VORBIS_invalid_first_page); - f->channels = get8(f); if (!f->channels) return error(f, VORBIS_invalid_first_page); - if (f->channels > STB_VORBIS_MAX_CHANNELS) return error(f, VORBIS_too_many_channels); - f->sample_rate = get32(f); if (!f->sample_rate) return error(f, VORBIS_invalid_first_page); - get32(f); // bitrate_maximum - get32(f); // bitrate_nominal - get32(f); // bitrate_minimum - x = get8(f); - { int log0,log1; - log0 = x & 15; - log1 = x >> 4; - f->blocksize_0 = 1 << log0; - f->blocksize_1 = 1 << log1; - if (log0 < 6 || log0 > 13) return error(f, VORBIS_invalid_setup); - if (log1 < 6 || log1 > 13) return error(f, VORBIS_invalid_setup); - if (log0 > log1) return error(f, VORBIS_invalid_setup); - } - - // framing_flag - x = get8(f); - if (!(x & 1)) return error(f, VORBIS_invalid_first_page); - - // second packet! - if (!start_page(f)) return FALSE; - - if (!start_packet(f)) return FALSE; - do { - len = next_segment(f); - skip(f, len); - f->bytes_in_seg = 0; - } while (len); - - // third packet! - if (!start_packet(f)) return FALSE; - - #ifndef STB_VORBIS_NO_PUSHDATA_API - if (IS_PUSH_MODE(f)) { - if (!is_whole_packet_present(f, TRUE)) { - // convert error in ogg header to write type - if (f->error == VORBIS_invalid_stream) - f->error = VORBIS_invalid_setup; - return FALSE; - } - } - #endif - - crc32_init(); // always init it, to avoid multithread race conditions - - if (get8_packet(f) != VORBIS_packet_setup) return error(f, VORBIS_invalid_setup); - for (i=0; i < 6; ++i) header[i] = get8_packet(f); - if (!vorbis_validate(header)) return error(f, VORBIS_invalid_setup); - - // codebooks - - f->codebook_count = get_bits(f,8) + 1; - f->codebooks = (Codebook *) setup_malloc(f, sizeof(*f->codebooks) * f->codebook_count); - if (f->codebooks == NULL) return error(f, VORBIS_outofmem); - memset(f->codebooks, 0, sizeof(*f->codebooks) * f->codebook_count); - for (i=0; i < f->codebook_count; ++i) { - uint32 *values; - int ordered, sorted_count; - int total=0; - uint8 *lengths; - Codebook *c = f->codebooks+i; - x = get_bits(f, 8); if (x != 0x42) return error(f, VORBIS_invalid_setup); - x = get_bits(f, 8); if (x != 0x43) return error(f, VORBIS_invalid_setup); - x = get_bits(f, 8); if (x != 0x56) return error(f, VORBIS_invalid_setup); - x = get_bits(f, 8); - c->dimensions = (get_bits(f, 8)<<8) + x; - x = get_bits(f, 8); - y = get_bits(f, 8); - c->entries = (get_bits(f, 8)<<16) + (y<<8) + x; - ordered = get_bits(f,1); - c->sparse = ordered ? 0 : get_bits(f,1); - - if (c->sparse) - lengths = (uint8 *) setup_temp_malloc(f, c->entries); - else - lengths = c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries); - - if (!lengths) return error(f, VORBIS_outofmem); - - if (ordered) { - int current_entry = 0; - int current_length = get_bits(f,5) + 1; - while (current_entry < c->entries) { - int limit = c->entries - current_entry; - int n = get_bits(f, ilog(limit)); - if (current_entry + n > (int) c->entries) { return error(f, VORBIS_invalid_setup); } - memset(lengths + current_entry, current_length, n); - current_entry += n; - ++current_length; - } - } else { - for (j=0; j < c->entries; ++j) { - int present = c->sparse ? get_bits(f,1) : 1; - if (present) { - lengths[j] = get_bits(f, 5) + 1; - ++total; - } else { - lengths[j] = NO_CODE; - } - } - } - - if (c->sparse && total >= c->entries >> 2) { - // convert sparse items to non-sparse! - if (c->entries > (int) f->setup_temp_memory_required) - f->setup_temp_memory_required = c->entries; - - c->codeword_lengths = (uint8 *) setup_malloc(f, c->entries); - memcpy(c->codeword_lengths, lengths, c->entries); - setup_temp_free(f, lengths, c->entries); // note this is only safe if there have been no intervening temp mallocs! - lengths = c->codeword_lengths; - c->sparse = 0; - } - - // compute the size of the sorted tables - if (c->sparse) { - sorted_count = total; - //assert(total != 0); - } else { - sorted_count = 0; - #ifndef STB_VORBIS_NO_HUFFMAN_BINARY_SEARCH - for (j=0; j < c->entries; ++j) - if (lengths[j] > STB_VORBIS_FAST_HUFFMAN_LENGTH && lengths[j] != NO_CODE) - ++sorted_count; - #endif - } - - c->sorted_entries = sorted_count; - values = NULL; - - if (!c->sparse) { - c->codewords = (uint32 *) setup_malloc(f, sizeof(c->codewords[0]) * c->entries); - if (!c->codewords) return error(f, VORBIS_outofmem); - } else { - unsigned int size; - if (c->sorted_entries) { - c->codeword_lengths = (uint8 *) setup_malloc(f, c->sorted_entries); - if (!c->codeword_lengths) return error(f, VORBIS_outofmem); - c->codewords = (uint32 *) setup_temp_malloc(f, sizeof(*c->codewords) * c->sorted_entries); - if (!c->codewords) return error(f, VORBIS_outofmem); - values = (uint32 *) setup_temp_malloc(f, sizeof(*values) * c->sorted_entries); - if (!values) return error(f, VORBIS_outofmem); - } - size = c->entries + (sizeof(*c->codewords) + sizeof(*values)) * c->sorted_entries; - if (size > f->setup_temp_memory_required) - f->setup_temp_memory_required = size; - } - - if (!compute_codewords(c, lengths, c->entries, values)) { - if (c->sparse) setup_temp_free(f, values, 0); - return error(f, VORBIS_invalid_setup); - } - - if (c->sorted_entries) { - // allocate an extra slot for sentinels - c->sorted_codewords = (uint32 *) setup_malloc(f, sizeof(*c->sorted_codewords) * (c->sorted_entries+1)); - // allocate an extra slot at the front so that c->sorted_values[-1] is defined - // so that we can catch that case without an extra if - c->sorted_values = ( int *) setup_malloc(f, sizeof(*c->sorted_values ) * (c->sorted_entries+1)); - if (c->sorted_values) { ++c->sorted_values; c->sorted_values[-1] = -1; } - compute_sorted_huffman(c, lengths, values); - } - - if (c->sparse) { - setup_temp_free(f, values, sizeof(*values)*c->sorted_entries); - setup_temp_free(f, c->codewords, sizeof(*c->codewords)*c->sorted_entries); - setup_temp_free(f, lengths, c->entries); - c->codewords = NULL; - } - - compute_accelerated_huffman(c); - - c->lookup_type = get_bits(f, 4); - if (c->lookup_type > 2) return error(f, VORBIS_invalid_setup); - if (c->lookup_type > 0) { - uint16 *mults; - c->minimum_value = float32_unpack(get_bits(f, 32)); - c->delta_value = float32_unpack(get_bits(f, 32)); - c->value_bits = get_bits(f, 4)+1; - c->sequence_p = get_bits(f,1); - if (c->lookup_type == 1) { - c->lookup_values = lookup1_values(c->entries, c->dimensions); - } else { - c->lookup_values = c->entries * c->dimensions; - } - mults = (uint16 *) setup_temp_malloc(f, sizeof(mults[0]) * c->lookup_values); - if (mults == NULL) return error(f, VORBIS_outofmem); - for (j=0; j < (int) c->lookup_values; ++j) { - int q = get_bits(f, c->value_bits); - if (q == EOP) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_invalid_setup); } - mults[j] = q; - } - -#ifndef STB_VORBIS_DIVIDES_IN_CODEBOOK - if (c->lookup_type == 1) { - int len, sparse = c->sparse; - // pre-expand the lookup1-style multiplicands, to avoid a divide in the inner loop - if (sparse) { - if (c->sorted_entries == 0) goto skip; - c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->sorted_entries * c->dimensions); - } else - c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->entries * c->dimensions); - if (c->multiplicands == NULL) { setup_temp_free(f,mults,sizeof(mults[0])*c->lookup_values); return error(f, VORBIS_outofmem); } - len = sparse ? c->sorted_entries : c->entries; - for (j=0; j < len; ++j) { - int z = sparse ? c->sorted_values[j] : j, div=1; - for (k=0; k < c->dimensions; ++k) { - int off = (z / div) % c->lookup_values; - c->multiplicands[j*c->dimensions + k] = - #ifndef STB_VORBIS_CODEBOOK_FLOATS - mults[off]; - #else - mults[off]*c->delta_value + c->minimum_value; - // in this case (and this case only) we could pre-expand c->sequence_p, - // and throw away the decode logic for it; have to ALSO do - // it in the case below, but it can only be done if - // STB_VORBIS_CODEBOOK_FLOATS - // !STB_VORBIS_DIVIDES_IN_CODEBOOK - #endif - div *= c->lookup_values; - } - } - setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); - c->lookup_type = 2; - } - else -#endif - { - c->multiplicands = (codetype *) setup_malloc(f, sizeof(c->multiplicands[0]) * c->lookup_values); - #ifndef STB_VORBIS_CODEBOOK_FLOATS - memcpy(c->multiplicands, mults, sizeof(c->multiplicands[0]) * c->lookup_values); - #else - for (j=0; j < (int) c->lookup_values; ++j) - c->multiplicands[j] = mults[j] * c->delta_value + c->minimum_value; - #endif - setup_temp_free(f, mults,sizeof(mults[0])*c->lookup_values); - } - skip:; - - #ifdef STB_VORBIS_CODEBOOK_FLOATS - if (c->lookup_type == 2 && c->sequence_p) { - for (j=1; j < (int) c->lookup_values; ++j) - c->multiplicands[j] = c->multiplicands[j-1]; - c->sequence_p = 0; - } - #endif - } - } - - // time domain transfers (notused) - - x = get_bits(f, 6) + 1; - for (i=0; i < x; ++i) { - uint32 z = get_bits(f, 16); - if (z != 0) return error(f, VORBIS_invalid_setup); - } - - // Floors - f->floor_count = get_bits(f, 6)+1; - f->floor_config = (Floor *) setup_malloc(f, f->floor_count * sizeof(*f->floor_config)); - for (i=0; i < f->floor_count; ++i) { - f->floor_types[i] = get_bits(f, 16); - if (f->floor_types[i] > 1) return error(f, VORBIS_invalid_setup); - if (f->floor_types[i] == 0) { - Floor0 *g = &f->floor_config[i].floor0; - g->order = get_bits(f,8); - g->rate = get_bits(f,16); - g->bark_map_size = get_bits(f,16); - g->amplitude_bits = get_bits(f,6); - g->amplitude_offset = get_bits(f,8); - g->number_of_books = get_bits(f,4) + 1; - for (j=0; j < g->number_of_books; ++j) - g->book_list[j] = get_bits(f,8); - return error(f, VORBIS_feature_not_supported); - } else { - Point p[31*8+2]; - Floor1 *g = &f->floor_config[i].floor1; - int max_class = -1; - g->partitions = get_bits(f, 5); - for (j=0; j < g->partitions; ++j) { - g->partition_class_list[j] = get_bits(f, 4); - if (g->partition_class_list[j] > max_class) - max_class = g->partition_class_list[j]; - } - for (j=0; j <= max_class; ++j) { - g->class_dimensions[j] = get_bits(f, 3)+1; - g->class_subclasses[j] = get_bits(f, 2); - if (g->class_subclasses[j]) { - g->class_masterbooks[j] = get_bits(f, 8); - if (g->class_masterbooks[j] >= f->codebook_count) return error(f, VORBIS_invalid_setup); - } - for (k=0; k < 1 << g->class_subclasses[j]; ++k) { - g->subclass_books[j][k] = get_bits(f,8)-1; - if (g->subclass_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup); - } - } - g->floor1_multiplier = get_bits(f,2)+1; - g->rangebits = get_bits(f,4); - g->Xlist[0] = 0; - g->Xlist[1] = 1 << g->rangebits; - g->values = 2; - for (j=0; j < g->partitions; ++j) { - int c = g->partition_class_list[j]; - for (k=0; k < g->class_dimensions[c]; ++k) { - g->Xlist[g->values] = get_bits(f, g->rangebits); - ++g->values; - } - } - // precompute the sorting - for (j=0; j < g->values; ++j) { - p[j].x = g->Xlist[j]; - p[j].y = j; - } - qsort(p, g->values, sizeof(p[0]), point_compare); - for (j=0; j < g->values; ++j) - g->sorted_order[j] = (uint8) p[j].y; - // precompute the neighbors - for (j=2; j < g->values; ++j) { - int low,hi; - neighbors(g->Xlist, j, &low,&hi); - g->neighbors[j][0] = low; - g->neighbors[j][1] = hi; - } - - if (g->values > longest_floorlist) - longest_floorlist = g->values; - } - } - - // Residue - f->residue_count = get_bits(f, 6)+1; - f->residue_config = (Residue *) setup_malloc(f, f->residue_count * sizeof(*f->residue_config)); - for (i=0; i < f->residue_count; ++i) { - uint8 residue_cascade[64]; - Residue *r = f->residue_config+i; - f->residue_types[i] = get_bits(f, 16); - if (f->residue_types[i] > 2) return error(f, VORBIS_invalid_setup); - r->begin = get_bits(f, 24); - r->end = get_bits(f, 24); - r->part_size = get_bits(f,24)+1; - r->classifications = get_bits(f,6)+1; - r->classbook = get_bits(f,8); - for (j=0; j < r->classifications; ++j) { - uint8 high_bits=0; - uint8 low_bits=get_bits(f,3); - if (get_bits(f,1)) - high_bits = get_bits(f,5); - residue_cascade[j] = high_bits*8 + low_bits; - } - r->residue_books = (short (*)[8]) setup_malloc(f, sizeof(r->residue_books[0]) * r->classifications); - for (j=0; j < r->classifications; ++j) { - for (k=0; k < 8; ++k) { - if (residue_cascade[j] & (1 << k)) { - r->residue_books[j][k] = get_bits(f, 8); - if (r->residue_books[j][k] >= f->codebook_count) return error(f, VORBIS_invalid_setup); - } else { - r->residue_books[j][k] = -1; - } - } - } - // precompute the classifications[] array to avoid inner-loop mod/divide - // call it 'classdata' since we already have r->classifications - r->classdata = (uint8 **) setup_malloc(f, sizeof(*r->classdata) * f->codebooks[r->classbook].entries); - if (!r->classdata) return error(f, VORBIS_outofmem); - memset(r->classdata, 0, sizeof(*r->classdata) * f->codebooks[r->classbook].entries); - for (j=0; j < f->codebooks[r->classbook].entries; ++j) { - int classwords = f->codebooks[r->classbook].dimensions; - int temp = j; - r->classdata[j] = (uint8 *) setup_malloc(f, sizeof(r->classdata[j][0]) * classwords); - for (k=classwords-1; k >= 0; --k) { - r->classdata[j][k] = temp % r->classifications; - temp /= r->classifications; - } - } - } - - f->mapping_count = get_bits(f,6)+1; - f->mapping = (Mapping *) setup_malloc(f, f->mapping_count * sizeof(*f->mapping)); - for (i=0; i < f->mapping_count; ++i) { - Mapping *m = f->mapping + i; - int mapping_type = get_bits(f,16); - if (mapping_type != 0) return error(f, VORBIS_invalid_setup); - m->chan = (MappingChannel *) setup_malloc(f, f->channels * sizeof(*m->chan)); - if (get_bits(f,1)) - m->submaps = get_bits(f,4); - else - m->submaps = 1; - if (m->submaps > max_submaps) - max_submaps = m->submaps; - if (get_bits(f,1)) { - m->coupling_steps = get_bits(f,8)+1; - for (k=0; k < m->coupling_steps; ++k) { - m->chan[k].magnitude = get_bits(f, ilog(f->channels)-1); - m->chan[k].angle = get_bits(f, ilog(f->channels)-1); - if (m->chan[k].magnitude >= f->channels) return error(f, VORBIS_invalid_setup); - if (m->chan[k].angle >= f->channels) return error(f, VORBIS_invalid_setup); - if (m->chan[k].magnitude == m->chan[k].angle) return error(f, VORBIS_invalid_setup); - } - } else - m->coupling_steps = 0; - - // reserved field - if (get_bits(f,2)) return error(f, VORBIS_invalid_setup); - if (m->submaps > 1) { - for (j=0; j < f->channels; ++j) { - m->chan[j].mux = get_bits(f, 4); - if (m->chan[j].mux >= m->submaps) return error(f, VORBIS_invalid_setup); - } - } else - // @SPECIFICATION: this case is missing from the spec - for (j=0; j < f->channels; ++j) - m->chan[j].mux = 0; - - for (j=0; j < m->submaps; ++j) { - get_bits(f,8); // discard - m->submap_floor[j] = get_bits(f,8); - m->submap_residue[j] = get_bits(f,8); - if (m->submap_floor[j] >= f->floor_count) return error(f, VORBIS_invalid_setup); - if (m->submap_residue[j] >= f->residue_count) return error(f, VORBIS_invalid_setup); - } - } - - // Modes - f->mode_count = get_bits(f, 6)+1; - for (i=0; i < f->mode_count; ++i) { - Mode *m = f->mode_config+i; - m->blockflag = get_bits(f,1); - m->windowtype = get_bits(f,16); - m->transformtype = get_bits(f,16); - m->mapping = get_bits(f,8); - if (m->windowtype != 0) return error(f, VORBIS_invalid_setup); - if (m->transformtype != 0) return error(f, VORBIS_invalid_setup); - if (m->mapping >= f->mapping_count) return error(f, VORBIS_invalid_setup); - } - - flush_packet(f); - - f->previous_length = 0; - - for (i=0; i < f->channels; ++i) { - f->channel_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1); - f->previous_window[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); - f->finalY[i] = (int16 *) setup_malloc(f, sizeof(int16) * longest_floorlist); - #ifdef STB_VORBIS_NO_DEFER_FLOOR - f->floor_buffers[i] = (float *) setup_malloc(f, sizeof(float) * f->blocksize_1/2); - #endif - } - - if (!init_blocksize(f, 0, f->blocksize_0)) return FALSE; - if (!init_blocksize(f, 1, f->blocksize_1)) return FALSE; - f->blocksize[0] = f->blocksize_0; - f->blocksize[1] = f->blocksize_1; - -#ifdef STB_VORBIS_DIVIDE_TABLE - if (integer_divide_table[1][1]==0) - for (i=0; i < DIVTAB_NUMER; ++i) - for (j=1; j < DIVTAB_DENOM; ++j) - integer_divide_table[i][j] = i / j; -#endif - - // compute how much temporary memory is needed - - // 1. - { - uint32 imdct_mem = (f->blocksize_1 * sizeof(float) >> 1); - uint32 classify_mem; - int i,max_part_read=0; - for (i=0; i < f->residue_count; ++i) { - Residue *r = f->residue_config + i; - int n_read = r->end - r->begin; - int part_read = n_read / r->part_size; - if (part_read > max_part_read) - max_part_read = part_read; - } - #ifndef STB_VORBIS_DIVIDES_IN_RESIDUE - classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(uint8 *)); - #else - classify_mem = f->channels * (sizeof(void*) + max_part_read * sizeof(int *)); - #endif - - f->temp_memory_required = classify_mem; - if (imdct_mem > f->temp_memory_required) - f->temp_memory_required = imdct_mem; - } - - f->first_decode = TRUE; - - if (f->alloc.alloc_buffer) { - assert(f->temp_offset == f->alloc.alloc_buffer_length_in_bytes); - // check if there's enough temp memory so we don't error later - if (f->setup_offset + sizeof(*f) + f->temp_memory_required > (unsigned) f->temp_offset) - return error(f, VORBIS_outofmem); - } - - f->first_audio_page_offset = stb_vorbis_get_file_offset(f); - - return TRUE; -} - -static void vorbis_deinit(stb_vorbis *p) -{ - int i,j; - for (i=0; i < p->residue_count; ++i) { - Residue *r = p->residue_config+i; - if (r->classdata) { - for (j=0; j < p->codebooks[r->classbook].entries; ++j) - setup_free(p, r->classdata[j]); - setup_free(p, r->classdata); - } - setup_free(p, r->residue_books); - } - - if (p->codebooks) { - for (i=0; i < p->codebook_count; ++i) { - Codebook *c = p->codebooks + i; - setup_free(p, c->codeword_lengths); - setup_free(p, c->multiplicands); - setup_free(p, c->codewords); - setup_free(p, c->sorted_codewords); - // c->sorted_values[-1] is the first entry in the array - setup_free(p, c->sorted_values ? c->sorted_values-1 : NULL); - } - setup_free(p, p->codebooks); - } - setup_free(p, p->floor_config); - setup_free(p, p->residue_config); - for (i=0; i < p->mapping_count; ++i) - setup_free(p, p->mapping[i].chan); - setup_free(p, p->mapping); - for (i=0; i < p->channels; ++i) { - setup_free(p, p->channel_buffers[i]); - setup_free(p, p->previous_window[i]); - #ifdef STB_VORBIS_NO_DEFER_FLOOR - setup_free(p, p->floor_buffers[i]); - #endif - setup_free(p, p->finalY[i]); - } - for (i=0; i < 2; ++i) { - setup_free(p, p->A[i]); - setup_free(p, p->B[i]); - setup_free(p, p->C[i]); - setup_free(p, p->window[i]); - setup_free(p, p->bit_reverse[i]); - } - #ifndef STB_VORBIS_NO_STDIO - if (p->close_on_free) fclose(p->f); - #endif -} - -void stb_vorbis_close(stb_vorbis *p) -{ - if (p == NULL) return; - vorbis_deinit(p); - setup_free(p,p); -} - -static void vorbis_init(stb_vorbis *p, stb_vorbis_alloc *z) -{ - memset(p, 0, sizeof(*p)); // NULL out all malloc'd pointers to start - if (z) { - p->alloc = *z; - p->alloc.alloc_buffer_length_in_bytes = (p->alloc.alloc_buffer_length_in_bytes+3) & ~3; - p->temp_offset = p->alloc.alloc_buffer_length_in_bytes; - } - p->eof = 0; - p->error = VORBIS__no_error; - p->stream = NULL; - p->codebooks = NULL; - p->page_crc_tests = -1; - #ifndef STB_VORBIS_NO_STDIO - p->close_on_free = FALSE; - p->f = NULL; - #endif -} - -int stb_vorbis_get_sample_offset(stb_vorbis *f) -{ - if (f->current_loc_valid) - return f->current_loc; - else - return -1; -} - -stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f) -{ - stb_vorbis_info d; - d.channels = f->channels; - d.sample_rate = f->sample_rate; - d.setup_memory_required = f->setup_memory_required; - d.setup_temp_memory_required = f->setup_temp_memory_required; - d.temp_memory_required = f->temp_memory_required; - d.max_frame_size = f->blocksize_1 >> 1; - return d; -} - -int stb_vorbis_get_error(stb_vorbis *f) -{ - int e = f->error; - f->error = VORBIS__no_error; - return e; -} - -static stb_vorbis * vorbis_alloc(stb_vorbis *f) -{ - stb_vorbis *p = (stb_vorbis *) setup_malloc(f, sizeof(*p)); - return p; -} - -#ifndef STB_VORBIS_NO_PUSHDATA_API - -void stb_vorbis_flush_pushdata(stb_vorbis *f) -{ - f->previous_length = 0; - f->page_crc_tests = 0; - f->discard_samples_deferred = 0; - f->current_loc_valid = FALSE; - f->first_decode = FALSE; - f->samples_output = 0; - f->channel_buffer_start = 0; - f->channel_buffer_end = 0; -} - -static int vorbis_search_for_page_pushdata(vorb *f, uint8 *data, int data_len) -{ - int i,n; - for (i=0; i < f->page_crc_tests; ++i) - f->scan[i].bytes_done = 0; - - // if we have room for more scans, search for them first, because - // they may cause us to stop early if their header is incomplete - if (f->page_crc_tests < STB_VORBIS_PUSHDATA_CRC_COUNT) { - if (data_len < 4) return 0; - data_len -= 3; // need to look for 4-byte sequence, so don't miss - // one that straddles a boundary - for (i=0; i < data_len; ++i) { - if (data[i] == 0x4f) { - if (0==memcmp(data+i, ogg_page_header, 4)) { - int j,len; - uint32 crc; - // make sure we have the whole page header - if (i+26 >= data_len || i+27+data[i+26] >= data_len) { - // only read up to this page start, so hopefully we'll - // have the whole page header start next time - data_len = i; - break; - } - // ok, we have it all; compute the length of the page - len = 27 + data[i+26]; - for (j=0; j < data[i+26]; ++j) - len += data[i+27+j]; - // scan everything up to the embedded crc (which we must 0) - crc = 0; - for (j=0; j < 22; ++j) - crc = crc32_update(crc, data[i+j]); - // now process 4 0-bytes - for ( ; j < 26; ++j) - crc = crc32_update(crc, 0); - // len is the total number of bytes we need to scan - n = f->page_crc_tests++; - f->scan[n].bytes_left = len-j; - f->scan[n].crc_so_far = crc; - f->scan[n].goal_crc = data[i+22] + (data[i+23] << 8) + (data[i+24]<<16) + (data[i+25]<<24); - // if the last frame on a page is continued to the next, then - // we can't recover the sample_loc immediately - if (data[i+27+data[i+26]-1] == 255) - f->scan[n].sample_loc = ~0; - else - f->scan[n].sample_loc = data[i+6] + (data[i+7] << 8) + (data[i+ 8]<<16) + (data[i+ 9]<<24); - f->scan[n].bytes_done = i+j; - if (f->page_crc_tests == STB_VORBIS_PUSHDATA_CRC_COUNT) - break; - // keep going if we still have room for more - } - } - } - } - - for (i=0; i < f->page_crc_tests;) { - uint32 crc; - int j; - int n = f->scan[i].bytes_done; - int m = f->scan[i].bytes_left; - if (m > data_len - n) m = data_len - n; - // m is the bytes to scan in the current chunk - crc = f->scan[i].crc_so_far; - for (j=0; j < m; ++j) - crc = crc32_update(crc, data[n+j]); - f->scan[i].bytes_left -= m; - f->scan[i].crc_so_far = crc; - if (f->scan[i].bytes_left == 0) { - // does it match? - if (f->scan[i].crc_so_far == f->scan[i].goal_crc) { - // Houston, we have page - data_len = n+m; // consumption amount is wherever that scan ended - f->page_crc_tests = -1; // drop out of page scan mode - f->previous_length = 0; // decode-but-don't-output one frame - f->next_seg = -1; // start a new page - f->current_loc = f->scan[i].sample_loc; // set the current sample location - // to the amount we'd have decoded had we decoded this page - f->current_loc_valid = f->current_loc != ~0; - return data_len; - } - // delete entry - f->scan[i] = f->scan[--f->page_crc_tests]; - } else { - ++i; - } - } - - return data_len; -} - -// return value: number of bytes we used -int stb_vorbis_decode_frame_pushdata( - stb_vorbis *f, // the file we're decoding - uint8 *data, int data_len, // the memory available for decoding - int *channels, // place to write number of float * buffers - float ***output, // place to write float ** array of float * buffers - int *samples // place to write number of output samples - ) -{ - int i; - int len,right,left; - - if (!IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); - - if (f->page_crc_tests >= 0) { - *samples = 0; - return vorbis_search_for_page_pushdata(f, data, data_len); - } - - f->stream = data; - f->stream_end = data + data_len; - f->error = VORBIS__no_error; - - // check that we have the entire packet in memory - if (!is_whole_packet_present(f, FALSE)) { - *samples = 0; - return 0; - } - - if (!vorbis_decode_packet(f, &len, &left, &right)) { - // save the actual error we encountered - enum STBVorbisError error = f->error; - if (error == VORBIS_bad_packet_type) { - // flush and resynch - f->error = VORBIS__no_error; - while (get8_packet(f) != EOP) - if (f->eof) break; - *samples = 0; - return f->stream - data; - } - if (error == VORBIS_continued_packet_flag_invalid) { - if (f->previous_length == 0) { - // we may be resynching, in which case it's ok to hit one - // of these; just discard the packet - f->error = VORBIS__no_error; - while (get8_packet(f) != EOP) - if (f->eof) break; - *samples = 0; - return f->stream - data; - } - } - // if we get an error while parsing, what to do? - // well, it DEFINITELY won't work to continue from where we are! - stb_vorbis_flush_pushdata(f); - // restore the error that actually made us bail - f->error = error; - *samples = 0; - return 1; - } - - // success! - len = vorbis_finish_frame(f, len, left, right); - for (i=0; i < f->channels; ++i) - f->outputs[i] = f->channel_buffers[i] + left; - - if (channels) *channels = f->channels; - *samples = len; - *output = f->outputs; - return f->stream - data; -} - -stb_vorbis *stb_vorbis_open_pushdata( - unsigned char *data, int data_len, // the memory available for decoding - int *data_used, // only defined if result is not NULL - int *error, stb_vorbis_alloc *alloc) -{ - stb_vorbis *f, p; - vorbis_init(&p, alloc); - p.stream = data; - p.stream_end = data + data_len; - p.push_mode = TRUE; - if (!start_decoder(&p)) { - if (p.eof) - *error = VORBIS_need_more_data; - else - *error = p.error; - return NULL; - } - f = vorbis_alloc(&p); - if (f) { - *f = p; - *data_used = f->stream - data; - *error = 0; - return f; - } else { - vorbis_deinit(&p); - return NULL; - } -} -#endif // STB_VORBIS_NO_PUSHDATA_API - -unsigned int stb_vorbis_get_file_offset(stb_vorbis *f) -{ - #ifndef STB_VORBIS_NO_PUSHDATA_API - if (f->push_mode) return 0; - #endif - if (USE_MEMORY(f)) return f->stream - f->stream_start; - #ifndef STB_VORBIS_NO_STDIO - return ftell(f->f) - f->f_start; - #endif -} - -#ifndef STB_VORBIS_NO_PULLDATA_API -// -// DATA-PULLING API -// - -static uint32 vorbis_find_page(stb_vorbis *f, uint32 *end, uint32 *last) -{ - for(;;) { - int n; - if (f->eof) return 0; - n = get8(f); - if (n == 0x4f) { // page header - unsigned int retry_loc = stb_vorbis_get_file_offset(f); - int i; - // check if we're off the end of a file_section stream - if (retry_loc - 25 > f->stream_len) - return 0; - // check the rest of the header - for (i=1; i < 4; ++i) - if (get8(f) != ogg_page_header[i]) - break; - if (f->eof) return 0; - if (i == 4) { - uint8 header[27]; - uint32 i, crc, goal, len; - for (i=0; i < 4; ++i) - header[i] = ogg_page_header[i]; - for (; i < 27; ++i) - header[i] = get8(f); - if (f->eof) return 0; - if (header[4] != 0) goto invalid; - goal = header[22] + (header[23] << 8) + (header[24]<<16) + (header[25]<<24); - for (i=22; i < 26; ++i) - header[i] = 0; - crc = 0; - for (i=0; i < 27; ++i) - crc = crc32_update(crc, header[i]); - len = 0; - for (i=0; i < header[26]; ++i) { - int s = get8(f); - crc = crc32_update(crc, s); - len += s; - } - if (len && f->eof) return 0; - for (i=0; i < len; ++i) - crc = crc32_update(crc, get8(f)); - // finished parsing probable page - if (crc == goal) { - // we could now check that it's either got the last - // page flag set, OR it's followed by the capture - // pattern, but I guess TECHNICALLY you could have - // a file with garbage between each ogg page and recover - // from it automatically? So even though that paranoia - // might decrease the chance of an invalid decode by - // another 2^32, not worth it since it would hose those - // invalid-but-useful files? - if (end) - *end = stb_vorbis_get_file_offset(f); - if (last) - if (header[5] & 0x04) - *last = 1; - else - *last = 0; - set_file_offset(f, retry_loc-1); - return 1; - } - } - invalid: - // not a valid page, so rewind and look for next one - set_file_offset(f, retry_loc); - } - } -} - -// seek is implemented with 'interpolation search'--this is like -// binary search, but we use the data values to estimate the likely -// location of the data item (plus a bit of a bias so when the -// estimation is wrong we don't waste overly much time) - -#define SAMPLE_unknown 0xffffffff - - -// ogg vorbis, in its insane infinite wisdom, only provides -// information about the sample at the END of the page. -// therefore we COULD have the data we need in the current -// page, and not know it. we could just use the end location -// as our only knowledge for bounds, seek back, and eventually -// the binary search finds it. or we can try to be smart and -// not waste time trying to locate more pages. we try to be -// smart, since this data is already in memory anyway, so -// doing needless I/O would be crazy! -static int vorbis_analyze_page(stb_vorbis *f, ProbedPage *z) -{ - uint8 header[27], lacing[255]; - uint8 packet_type[255]; - int num_packet, packet_start, previous =0; - int i,len; - uint32 samples; - - // record where the page starts - z->page_start = stb_vorbis_get_file_offset(f); - - // parse the header - getn(f, header, 27); - assert(header[0] == 'O' && header[1] == 'g' && header[2] == 'g' && header[3] == 'S'); - getn(f, lacing, header[26]); - - // determine the length of the payload - len = 0; - for (i=0; i < header[26]; ++i) - len += lacing[i]; - - // this implies where the page ends - z->page_end = z->page_start + 27 + header[26] + len; - - // read the last-decoded sample out of the data - z->last_decoded_sample = header[6] + (header[7] << 8) + (header[8] << 16) + (header[9] << 16); - - if (header[5] & 4) { - // if this is the last page, it's not possible to work - // backwards to figure out the first sample! whoops! fuck. - z->first_decoded_sample = SAMPLE_unknown; - set_file_offset(f, z->page_start); - return 1; - } - - // scan through the frames to determine the sample-count of each one... - // our goal is the sample # of the first fully-decoded sample on the - // page, which is the first decoded sample of the 2nd page - - num_packet=0; - - packet_start = ((header[5] & 1) == 0); - - for (i=0; i < header[26]; ++i) { - if (packet_start) { - uint8 n,b,m; - if (lacing[i] == 0) goto bail; // trying to read from zero-length packet - n = get8(f); - // if bottom bit is non-zero, we've got corruption - if (n & 1) goto bail; - n >>= 1; - b = ilog(f->mode_count-1); - m = n >> b; - n &= (1 << b)-1; - if (n >= f->mode_count) goto bail; - if (num_packet == 0 && f->mode_config[n].blockflag) - previous = (m & 1); - packet_type[num_packet++] = f->mode_config[n].blockflag; - skip(f, lacing[i]-1); - } else - skip(f, lacing[i]); - packet_start = (lacing[i] < 255); - } - - // now that we know the sizes of all the pages, we can start determining - // how much sample data there is. - - samples = 0; - - // for the last packet, we step by its whole length, because the definition - // is that we encoded the end sample loc of the 'last packet completed', - // where 'completed' refers to packets being split, and we are left to guess - // what 'end sample loc' means. we assume it means ignoring the fact that - // the last half of the data is useless without windowing against the next - // packet... (so it's not REALLY complete in that sense) - if (num_packet > 1) - samples += f->blocksize[packet_type[num_packet-1]]; - - for (i=num_packet-2; i >= 1; --i) { - // now, for this packet, how many samples do we have that - // do not overlap the following packet? - if (packet_type[i] == 1) - if (packet_type[i+1] == 1) - samples += f->blocksize_1 >> 1; - else - samples += ((f->blocksize_1 - f->blocksize_0) >> 2) + (f->blocksize_0 >> 1); - else - samples += f->blocksize_0 >> 1; - } - // now, at this point, we've rewound to the very beginning of the - // _second_ packet. if we entirely discard the first packet after - // a seek, this will be exactly the right sample number. HOWEVER! - // we can't as easily compute this number for the LAST page. The - // only way to get the sample offset of the LAST page is to use - // the end loc from the previous page. But what that returns us - // is _exactly_ the place where we get our first non-overlapped - // sample. (I think. Stupid spec for being ambiguous.) So for - // consistency it's better to do that here, too. However, that - // will then require us to NOT discard all of the first frame we - // decode, in some cases, which means an even weirder frame size - // and extra code. what a fucking pain. - - // we're going to discard the first packet if we - // start the seek here, so we don't care about it. (we could actually - // do better; if the first packet is long, and the previous packet - // is short, there's actually data in the first half of the first - // packet that doesn't need discarding... but not worth paying the - // effort of tracking that of that here and in the seeking logic) - // except crap, if we infer it from the _previous_ packet's end - // location, we DO need to use that definition... and we HAVE to - // infer the start loc of the LAST packet from the previous packet's - // end location. fuck you, ogg vorbis. - - z->first_decoded_sample = z->last_decoded_sample - samples; - - // restore file state to where we were - set_file_offset(f, z->page_start); - return 1; - - // restore file state to where we were - bail: - set_file_offset(f, z->page_start); - return 0; -} - -static int vorbis_seek_frame_from_page(stb_vorbis *f, uint32 page_start, uint32 first_sample, uint32 target_sample, int fine) -{ - int left_start, left_end, right_start, right_end, mode,i; - int frame=0; - uint32 frame_start; - int frames_to_skip, data_to_skip; - - // first_sample is the sample # of the first sample that doesn't - // overlap the previous page... note that this requires us to - // _partially_ discard the first packet! bleh. - set_file_offset(f, page_start); - - f->next_seg = -1; // force page resync - - frame_start = first_sample; - // frame start is where the previous packet's last decoded sample - // was, which corresponds to left_end... EXCEPT if the previous - // packet was long and this packet is short? Probably a bug here. - - - // now, we can start decoding frames... we'll only FAKE decode them, - // until we find the frame that contains our sample; then we'll rewind, - // and try again - for (;;) { - int start; - - if (!vorbis_decode_initial(f, &left_start, &left_end, &right_start, &right_end, &mode)) - return error(f, VORBIS_seek_failed); - - if (frame == 0) - start = left_end; - else - start = left_start; - - // the window starts at left_start; the last valid sample we generate - // before the next frame's window start is right_start-1 - if (target_sample < frame_start + right_start-start) - break; - - flush_packet(f); - if (f->eof) - return error(f, VORBIS_seek_failed); - - frame_start += right_start - start; - - ++frame; - } - - // ok, at this point, the sample we want is contained in frame #'frame' - - // to decode frame #'frame' normally, we have to decode the - // previous frame first... but if it's the FIRST frame of the page - // we can't. if it's the first frame, it means it falls in the part - // of the first frame that doesn't overlap either of the other frames. - // so, if we have to handle that case for the first frame, we might - // as well handle it for all of them, so: - if (target_sample > frame_start + (left_end - left_start)) { - // so what we want to do is go ahead and just immediately decode - // this frame, but then make it so the next get_frame_float() uses - // this already-decoded data? or do we want to go ahead and rewind, - // and leave a flag saying to skip the first N data? let's do that - frames_to_skip = frame; // if this is frame #1, skip 1 frame (#0) - data_to_skip = left_end - left_start; - } else { - // otherwise, we want to skip frames 0, 1, 2, ... frame-2 - // (which means frame-2+1 total frames) then decode frame-1, - // then leave frame pending - frames_to_skip = frame - 1; - assert(frames_to_skip >= 0); - data_to_skip = -1; - } - - set_file_offset(f, page_start); - f->next_seg = - 1; // force page resync - - for (i=0; i < frames_to_skip; ++i) { - maybe_start_packet(f); - flush_packet(f); - } - - if (data_to_skip >= 0) { - int i,j,n = f->blocksize_0 >> 1; - f->discard_samples_deferred = data_to_skip; - for (i=0; i < f->channels; ++i) - for (j=0; j < n; ++j) - f->previous_window[i][j] = 0; - f->previous_length = n; - frame_start += data_to_skip; - } else { - f->previous_length = 0; - vorbis_pump_first_frame(f); - } - - // at this point, the NEXT decoded frame will generate the desired sample - if (fine) { - // so if we're doing sample accurate streaming, we want to go ahead and decode it! - if (target_sample != frame_start) { - int n; - stb_vorbis_get_frame_float(f, &n, NULL); - assert(target_sample > frame_start); - assert(f->channel_buffer_start + (int) (target_sample-frame_start) < f->channel_buffer_end); - f->channel_buffer_start += (target_sample - frame_start); - } - } - - return 0; -} - -static int vorbis_seek_base(stb_vorbis *f, unsigned int sample_number, int fine) -{ - ProbedPage p[2],q; - if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); - - // do we know the location of the last page? - if (f->p_last.page_start == 0) { - uint32 z = stb_vorbis_stream_length_in_samples(f); - if (z == 0) return error(f, VORBIS_cant_find_last_page); - } - - p[0] = f->p_first; - p[1] = f->p_last; - - if (sample_number >= f->p_last.last_decoded_sample) - sample_number = f->p_last.last_decoded_sample-1; - - if (sample_number < f->p_first.last_decoded_sample) { - vorbis_seek_frame_from_page(f, p[0].page_start, 0, sample_number, fine); - return 0; - } else { - int attempts=0; - while (p[0].page_end < p[1].page_start) { - uint32 probe; - uint32 start_offset, end_offset; - uint32 start_sample, end_sample; - - // copy these into local variables so we can tweak them - // if any are unknown - start_offset = p[0].page_end; - end_offset = p[1].after_previous_page_start; // an address known to seek to page p[1] - start_sample = p[0].last_decoded_sample; - end_sample = p[1].last_decoded_sample; - - // currently there is no such tweaking logic needed/possible? - if (start_sample == SAMPLE_unknown || end_sample == SAMPLE_unknown) - return error(f, VORBIS_seek_failed); - - // now we want to lerp between these for the target samples... - - // step 1: we need to bias towards the page start... - if (start_offset + 4000 < end_offset) - end_offset -= 4000; - - // now compute an interpolated search loc - probe = start_offset + (int) floor((float) (end_offset - start_offset) / (end_sample - start_sample) * (sample_number - start_sample)); - - // next we need to bias towards binary search... - // code is a little wonky to allow for full 32-bit unsigned values - if (attempts >= 4) { - uint32 probe2 = start_offset + ((end_offset - start_offset) >> 1); - if (attempts >= 8) - probe = probe2; - else if (probe < probe2) - probe = probe + ((probe2 - probe) >> 1); - else - probe = probe2 + ((probe - probe2) >> 1); - } - ++attempts; - - set_file_offset(f, probe); - if (!vorbis_find_page(f, NULL, NULL)) return error(f, VORBIS_seek_failed); - if (!vorbis_analyze_page(f, &q)) return error(f, VORBIS_seek_failed); - q.after_previous_page_start = probe; - - // it's possible we've just found the last page again - if (q.page_start == p[1].page_start) { - p[1] = q; - continue; - } - - if (sample_number < q.last_decoded_sample) - p[1] = q; - else - p[0] = q; - } - - if (p[0].last_decoded_sample <= sample_number && sample_number < p[1].last_decoded_sample) { - vorbis_seek_frame_from_page(f, p[1].page_start, p[0].last_decoded_sample, sample_number, fine); - return 0; - } - return error(f, VORBIS_seek_failed); - } -} - -int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number) -{ - return vorbis_seek_base(f, sample_number, FALSE); -} - -int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number) -{ - return vorbis_seek_base(f, sample_number, TRUE); -} - -void stb_vorbis_seek_start(stb_vorbis *f) -{ - if (IS_PUSH_MODE(f)) { error(f, VORBIS_invalid_api_mixing); return; } - set_file_offset(f, f->first_audio_page_offset); - f->previous_length = 0; - f->first_decode = TRUE; - f->next_seg = -1; - vorbis_pump_first_frame(f); -} - -unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f) -{ - unsigned int restore_offset, previous_safe; - unsigned int end, last_page_loc; - - if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); - if (!f->total_samples) { - int last; - uint32 lo,hi; - char header[6]; - - // first, store the current decode position so we can restore it - restore_offset = stb_vorbis_get_file_offset(f); - - // now we want to seek back 64K from the end (the last page must - // be at most a little less than 64K, but let's allow a little slop) - if (f->stream_len >= 65536 && f->stream_len-65536 >= f->first_audio_page_offset) - previous_safe = f->stream_len - 65536; - else - previous_safe = f->first_audio_page_offset; - - set_file_offset(f, previous_safe); - // previous_safe is now our candidate 'earliest known place that seeking - // to will lead to the final page' - - if (!vorbis_find_page(f, &end, (int unsigned *)&last)) { - // if we can't find a page, we're hosed! - f->error = VORBIS_cant_find_last_page; - f->total_samples = 0xffffffff; - goto done; - } - - // check if there are more pages - last_page_loc = stb_vorbis_get_file_offset(f); - - // stop when the last_page flag is set, not when we reach eof; - // this allows us to stop short of a 'file_section' end without - // explicitly checking the length of the section - while (!last) { - set_file_offset(f, end); - if (!vorbis_find_page(f, &end, (int unsigned *)&last)) { - // the last page we found didn't have the 'last page' flag - // set. whoops! - break; - } - previous_safe = last_page_loc+1; - last_page_loc = stb_vorbis_get_file_offset(f); - } - - set_file_offset(f, last_page_loc); - - // parse the header - getn(f, (unsigned char *)header, 6); - // extract the absolute granule position - lo = get32(f); - hi = get32(f); - if (lo == 0xffffffff && hi == 0xffffffff) { - f->error = VORBIS_cant_find_last_page; - f->total_samples = SAMPLE_unknown; - goto done; - } - if (hi) - lo = 0xfffffffe; // saturate - f->total_samples = lo; - - f->p_last.page_start = last_page_loc; - f->p_last.page_end = end; - f->p_last.last_decoded_sample = lo; - f->p_last.first_decoded_sample = SAMPLE_unknown; - f->p_last.after_previous_page_start = previous_safe; - - done: - set_file_offset(f, restore_offset); - } - return f->total_samples == SAMPLE_unknown ? 0 : f->total_samples; -} - -float stb_vorbis_stream_length_in_seconds(stb_vorbis *f) -{ - return stb_vorbis_stream_length_in_samples(f) / (float) f->sample_rate; -} - - - -int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output) -{ - int len, right,left,i; - if (IS_PUSH_MODE(f)) return error(f, VORBIS_invalid_api_mixing); - - if (!vorbis_decode_packet(f, &len, &left, &right)) { - f->channel_buffer_start = f->channel_buffer_end = 0; - return 0; - } - - len = vorbis_finish_frame(f, len, left, right); - for (i=0; i < f->channels; ++i) - f->outputs[i] = f->channel_buffers[i] + left; - - f->channel_buffer_start = left; - f->channel_buffer_end = left+len; - - if (channels) *channels = f->channels; - if (output) *output = f->outputs; - return len; -} - -#ifndef STB_VORBIS_NO_STDIO - -stb_vorbis * stb_vorbis_open_file_section(FILE *file, int close_on_free, int *error, stb_vorbis_alloc *alloc, unsigned int length) -{ - stb_vorbis *f, p; - vorbis_init(&p, alloc); - p.f = file; - p.f_start = ftell(file); - p.stream_len = length; - p.close_on_free = close_on_free; - if (start_decoder(&p)) { - f = vorbis_alloc(&p); - if (f) { - *f = p; - vorbis_pump_first_frame(f); - return f; - } - } - if (error) *error = p.error; - vorbis_deinit(&p); - return NULL; -} - -stb_vorbis * stb_vorbis_open_file(FILE *file, int close_on_free, int *error, stb_vorbis_alloc *alloc) -{ - unsigned int len, start; - start = ftell(file); - fseek(file, 0, SEEK_END); - len = ftell(file) - start; - fseek(file, start, SEEK_SET); - return stb_vorbis_open_file_section(file, close_on_free, error, alloc, len); -} - -stb_vorbis * stb_vorbis_open_filename(char *filename, int *error, stb_vorbis_alloc *alloc) -{ - FILE *f = fopen(filename, "rb"); - if (f) - return stb_vorbis_open_file(f, TRUE, error, alloc); - if (error) *error = VORBIS_file_open_failure; - return NULL; -} -#endif // STB_VORBIS_NO_STDIO - -stb_vorbis * stb_vorbis_open_memory(unsigned char *data, int len, int *error, stb_vorbis_alloc *alloc) -{ - stb_vorbis *f, p; - if (data == NULL) return NULL; - vorbis_init(&p, alloc); - p.stream = data; - p.stream_end = data + len; - p.stream_start = p.stream; - p.stream_len = len; - p.push_mode = FALSE; - if (start_decoder(&p)) { - f = vorbis_alloc(&p); - if (f) { - *f = p; - vorbis_pump_first_frame(f); - return f; - } - } - if (error) *error = p.error; - vorbis_deinit(&p); - return NULL; -} - -#ifndef STB_VORBIS_NO_INTEGER_CONVERSION -#define PLAYBACK_MONO 1 -#define PLAYBACK_LEFT 2 -#define PLAYBACK_RIGHT 4 - -#define L (PLAYBACK_LEFT | PLAYBACK_MONO) -#define C (PLAYBACK_LEFT | PLAYBACK_RIGHT | PLAYBACK_MONO) -#define R (PLAYBACK_RIGHT | PLAYBACK_MONO) - -static int8 channel_position[7][6] = -{ - { 0 }, - { C }, - { L, R }, - { L, C, R }, - { L, R, L, R }, - { L, C, R, L, R }, - { L, C, R, L, R, C }, -}; - - -#ifndef STB_VORBIS_NO_FAST_SCALED_FLOAT - // add (1<<23) to convert to int, then divide by 2^SHIFT, then add 0.5/2^SHIFT to round - #define MAGIC(SHIFT) (1.5f * (1 << (23-SHIFT)) + 0.5f/(1 << SHIFT)) - #define ADDEND(SHIFT) (((150-SHIFT) << 23) + (1 << 22)) - #define FAST_SCALED_FLOAT_TO_INT(x,s) ((temp = (x) + MAGIC(s)), (*(int *)&temp) - ADDEND(s)) - #define check_endianness() - typedef char stb_vorbis_float_size_test[sizeof(float)==4 && sizeof(int) == 4]; - #define FASTDEF(x) x -#else - #define FAST_SCALED_FLOAT_TO_INT(x,s) ((int) ((x) * (1 << (s)))) - #define check_endianness() - #define FASTDEF(x) -#endif - -static void copy_samples(short *dest, float *src, int len) -{ - int i; - FASTDEF(float temp); - check_endianness(); - for (i=0; i < len; ++i) { - int v = FAST_SCALED_FLOAT_TO_INT(src[i],15); - if ((unsigned int) (v + 32768) > 65535) - v = v < 0 ? -32768 : 32767; - dest[i] = v; - } -} - -static void compute_samples(int mask, short *output, int num_c, float **data, int d_offset, int len) -{ - #define BUFFER_SIZE 32 - float buffer[BUFFER_SIZE]; - int i,j,o,n = BUFFER_SIZE; - FASTDEF(float temp); - check_endianness(); - for (o = 0; o < len; o += BUFFER_SIZE) { - memset(buffer, 0, sizeof(buffer)); - if (o + n > len) n = len - o; - for (j=0; j < num_c; ++j) { - if (channel_position[num_c][j] & mask) { - for (i=0; i < n; ++i) - buffer[i] += data[j][d_offset+o+i]; - } - } - for (i=0; i < n; ++i) { - int v = FAST_SCALED_FLOAT_TO_INT(buffer[i],15); - if ((unsigned int) (v + 32768) > 65535) - v = v < 0 ? -32768 : 32767; - output[o+i] = v; - } - } -} - -static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} }; -static void compute_stereo_samples(short *output, int num_c, float **data, int d_offset, int len) -{ - #define BUFFER_SIZE 32 - float buffer[BUFFER_SIZE]; - int i,j,o,n = BUFFER_SIZE >> 1; - FASTDEF(float temp); - // o is the offset in the source data - check_endianness(); - for (o = 0; o < len; o += BUFFER_SIZE >> 1) { - // o2 is the offset in the output data - int o2 = o << 1; - memset(buffer, 0, sizeof(buffer)); - if (o + n > len) n = len - o; - for (j=0; j < num_c; ++j) { - int m = channel_position[num_c][j] & (PLAYBACK_LEFT | PLAYBACK_RIGHT); - if (m == (PLAYBACK_LEFT | PLAYBACK_RIGHT)) { - for (i=0; i < n; ++i) { - buffer[i*2+0] += data[j][d_offset+o+i]; - buffer[i*2+1] += data[j][d_offset+o+i]; - } - } else if (m == PLAYBACK_LEFT) { - for (i=0; i < n; ++i) { - buffer[i*2+0] += data[j][d_offset+o+i]; - } - } else if (m == PLAYBACK_RIGHT) { - for (i=0; i < n; ++i) { - buffer[i*2+1] += data[j][d_offset+o+i]; - } - } - } - for (i=0; i < (n<<1); ++i) { - int v = FAST_SCALED_FLOAT_TO_INT(buffer[i],15); - if ((unsigned int) (v + 32768) > 65535) - v = v < 0 ? -32768 : 32767; - output[o2+i] = v; - } - } -} - -static void convert_samples_short(int buf_c, short **buffer, int b_offset, int data_c, float **data, int d_offset, int samples) -{ - int i; - if (buf_c != data_c && buf_c <= 2 && data_c <= 6) { - static int channel_selector[3][2] = { {0}, {PLAYBACK_MONO}, {PLAYBACK_LEFT, PLAYBACK_RIGHT} }; - for (i=0; i < buf_c; ++i) - compute_samples(channel_selector[buf_c][i], buffer[i]+b_offset, data_c, data, d_offset, samples); - } else { - int limit = buf_c < data_c ? buf_c : data_c; - for (i=0; i < limit; ++i) - copy_samples(buffer[i]+b_offset, data[i], samples); - for ( ; i < buf_c; ++i) - memset(buffer[i]+b_offset, 0, sizeof(short) * samples); - } -} - -int stb_vorbis_get_frame_short(stb_vorbis *f, int num_c, short **buffer, int num_samples) -{ - float **output; - int len = stb_vorbis_get_frame_float(f, NULL, &output); - if (len > num_samples) len = num_samples; - if (len) - convert_samples_short(num_c, buffer, 0, f->channels, output, 0, len); - return len; -} - -static void convert_channels_short_interleaved(int buf_c, short *buffer, int data_c, float **data, int d_offset, int len) -{ - int i; - check_endianness(); - if (buf_c != data_c && buf_c <= 2 && data_c <= 6) { - assert(buf_c == 2); - for (i=0; i < buf_c; ++i) - compute_stereo_samples(buffer, data_c, data, d_offset, len); - } else { - int limit = buf_c < data_c ? buf_c : data_c; - int j; - FASTDEF(float temp); - for (j=0; j < len; ++j) { - for (i=0; i < limit; ++i) { - int v = FAST_SCALED_FLOAT_TO_INT(data[i][d_offset+j],15); - if ((unsigned int) (v + 32768) > 65535) - v = v < 0 ? -32768 : 32767; - *buffer++ = v; - } - for ( ; i < buf_c; ++i) - *buffer++ = 0; - } - } -} - -int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts) -{ - float **output; - int len; - if (num_c == 1) return stb_vorbis_get_frame_short(f,num_c,&buffer, num_shorts); - len = stb_vorbis_get_frame_float(f, NULL, &output); - if (len) { - if (len*num_c > num_shorts) len = num_shorts / num_c; - convert_channels_short_interleaved(num_c, buffer, f->channels, output, 0, len); - } - return len; -} - -int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts) -{ - float **outputs; - int len = num_shorts / channels; - int n=0; - int z = f->channels; - if (z > channels) z = channels; - while (n < len) { - int k = f->channel_buffer_end - f->channel_buffer_start; - if (n+k >= len) k = len - n; - if (k) - convert_channels_short_interleaved(channels, buffer, f->channels, f->channel_buffers, f->channel_buffer_start, k); - buffer += k*channels; - n += k; - f->channel_buffer_start += k; - if (n == len) break; - if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; - } - return n; -} - -int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int len) -{ - float **outputs; - int n=0; - int z = f->channels; - if (z > channels) z = channels; - while (n < len) { - int k = f->channel_buffer_end - f->channel_buffer_start; - if (n+k >= len) k = len - n; - if (k) - convert_samples_short(channels, buffer, n, f->channels, f->channel_buffers, f->channel_buffer_start, k); - n += k; - f->channel_buffer_start += k; - if (n == len) break; - if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; - } - return n; -} - -#ifndef STB_VORBIS_NO_STDIO -int stb_vorbis_decode_filename(char *filename, int *channels, short **output) -{ - int data_len, offset, total, limit, error; - short *data; - stb_vorbis *v = stb_vorbis_open_filename(filename, &error, NULL); - if (v == NULL) return -1; - limit = v->channels * 4096; - *channels = v->channels; - offset = data_len = 0; - total = limit; - data = (short *) malloc(total * sizeof(*data)); - if (data == NULL) { - stb_vorbis_close(v); - return -2; - } - for (;;) { - int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset); - if (n == 0) break; - data_len += n; - offset += n * v->channels; - if (offset + limit > total) { - short *data2; - total *= 2; - data2 = (short *) realloc(data, total * sizeof(*data)); - if (data2 == NULL) { - free(data); - stb_vorbis_close(v); - return -2; - } - data = data2; - } - } - *output = data; - return data_len; -} -#endif // NO_STDIO - -int stb_vorbis_decode_memory(uint8 *mem, int len, int *channels, short **output) -{ - int data_len, offset, total, limit, error; - short *data; - stb_vorbis *v = stb_vorbis_open_memory(mem, len, &error, NULL); - if (v == NULL) return -1; - limit = v->channels * 4096; - *channels = v->channels; - offset = data_len = 0; - total = limit; - data = (short *) malloc(total * sizeof(*data)); - if (data == NULL) { - stb_vorbis_close(v); - return -2; - } - for (;;) { - int n = stb_vorbis_get_frame_short_interleaved(v, v->channels, data+offset, total-offset); - if (n == 0) break; - data_len += n; - offset += n * v->channels; - if (offset + limit > total) { - short *data2; - total *= 2; - data2 = (short *) realloc(data, total * sizeof(*data)); - if (data2 == NULL) { - free(data); - stb_vorbis_close(v); - return -2; - } - data = data2; - } - } - *output = data; - return data_len; -} -#endif - -int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats) -{ - float **outputs; - int len = num_floats / channels; - int n=0; - int z = f->channels; - if (z > channels) z = channels; - while (n < len) { - int i,j; - int k = f->channel_buffer_end - f->channel_buffer_start; - if (n+k >= len) k = len - n; - for (j=0; j < k; ++j) { - for (i=0; i < z; ++i) - *buffer++ = f->channel_buffers[i][f->channel_buffer_start+j]; - for ( ; i < channels; ++i) - *buffer++ = 0; - } - n += k; - f->channel_buffer_start += k; - if (n == len) break; - if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; - } - return n; -} - -int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples) -{ - float **outputs; - int n=0; - int z = f->channels; - if (z > channels) z = channels; - while (n < num_samples) { - int i; - int k = f->channel_buffer_end - f->channel_buffer_start; - if (n+k >= num_samples) k = num_samples - n; - if (k) { - for (i=0; i < z; ++i) - memcpy(buffer[i]+n, f->channel_buffers+f->channel_buffer_start, sizeof(float)*k); - for ( ; i < channels; ++i) - memset(buffer[i]+n, 0, sizeof(float) * k); - } - n += k; - f->channel_buffer_start += k; - if (n == num_samples) break; - if (!stb_vorbis_get_frame_float(f, NULL, &outputs)) break; - } - return n; -} -#endif // STB_VORBIS_NO_PULLDATA_API - -#endif // STB_VORBIS_HEADER_ONLY diff --git a/src/SFML/Audio/stb_vorbis/stb_vorbis.h b/src/SFML/Audio/stb_vorbis/stb_vorbis.h deleted file mode 100755 index e2355e6ce..000000000 --- a/src/SFML/Audio/stb_vorbis/stb_vorbis.h +++ /dev/null @@ -1,357 +0,0 @@ -// Ogg Vorbis I audio decoder -- version 0.99994 -// -// Written in April 2007 by Sean Barrett, sponsored by RAD Game Tools. -// -// Placed in the public domain April 2007 by the author: no copyright is -// claimed, and you may use it for any purpose you like. -// -// No warranty for any purpose is expressed or implied by the author (nor -// by RAD Game Tools). Report bugs and send enhancements to the author. -// -// Get the latest version and other information at: -// http://nothings.org/stb_vorbis/ - - -// Todo: -// -// - seeking (note you can seek yourself using the pushdata API) -// -// Limitations: -// -// - floor 0 not supported (used in old ogg vorbis files) -// - lossless sample-truncation at beginning ignored -// - cannot concatenate multiple vorbis streams -// - sample positions are 32-bit, limiting seekable 192Khz -// files to around 6 hours (Ogg supports 64-bit) -// -// All of these limitations may be removed in future versions. - - -////////////////////////////////////////////////////////////////////////////// -// -// HEADER BEGINS HERE -// - -#ifndef STB_VORBIS_INCLUDE_STB_VORBIS_H -#define STB_VORBIS_INCLUDE_STB_VORBIS_H - -#if defined(STB_VORBIS_NO_CRT) && !defined(STB_VORBIS_NO_STDIO) -#define STB_VORBIS_NO_STDIO 1 -#endif - -#ifndef STB_VORBIS_NO_STDIO -#include -#endif - -#ifdef __cplusplus -extern "C" { -#endif - -/////////// THREAD SAFETY - -// Individual stb_vorbis* handles are not thread-safe; you cannot decode from -// them from multiple threads at the same time. However, you can have multiple -// stb_vorbis* handles and decode from them independently in multiple thrads. - - -/////////// MEMORY ALLOCATION - -// normally stb_vorbis uses malloc() to allocate memory at startup, -// and alloca() to allocate temporary memory during a frame on the -// stack. (Memory consumption will depend on the amount of setup -// data in the file and how you set the compile flags for speed -// vs. size. In my test files the maximal-size usage is ~150KB.) -// -// You can modify the wrapper functions in the source (setup_malloc, -// setup_temp_malloc, temp_malloc) to change this behavior, or you -// can use a simpler allocation model: you pass in a buffer from -// which stb_vorbis will allocate _all_ its memory (including the -// temp memory). "open" may fail with a VORBIS_outofmem if you -// do not pass in enough data; there is no way to determine how -// much you do need except to succeed (at which point you can -// query get_info to find the exact amount required. yes I know -// this is lame). -// -// If you pass in a non-NULL buffer of the type below, allocation -// will occur from it as described above. Otherwise just pass NULL -// to use malloc()/alloca() - -typedef struct -{ - char *alloc_buffer; - int alloc_buffer_length_in_bytes; -} stb_vorbis_alloc; - - -/////////// FUNCTIONS USEABLE WITH ALL INPUT MODES - -typedef struct stb_vorbis stb_vorbis; - -typedef struct -{ - unsigned int sample_rate; - int channels; - - unsigned int setup_memory_required; - unsigned int setup_temp_memory_required; - unsigned int temp_memory_required; - - int max_frame_size; -} stb_vorbis_info; - -// get general information about the file -extern stb_vorbis_info stb_vorbis_get_info(stb_vorbis *f); - -// get the last error detected (clears it, too) -extern int stb_vorbis_get_error(stb_vorbis *f); - -// close an ogg vorbis file and free all memory in use -extern void stb_vorbis_close(stb_vorbis *f); - -// this function returns the offset (in samples) from the beginning of the -// file that will be returned by the next decode, if it is known, or -1 -// otherwise. after a flush_pushdata() call, this may take a while before -// it becomes valid again. -// NOT WORKING YET after a seek with PULLDATA API -extern int stb_vorbis_get_sample_offset(stb_vorbis *f); - -// returns the current seek point within the file, or offset from the beginning -// of the memory buffer. In pushdata mode it returns 0. -extern unsigned int stb_vorbis_get_file_offset(stb_vorbis *f); - -/////////// PUSHDATA API - -#ifndef STB_VORBIS_NO_PUSHDATA_API - -// this API allows you to get blocks of data from any source and hand -// them to stb_vorbis. you have to buffer them; stb_vorbis will tell -// you how much it used, and you have to give it the rest next time; -// and stb_vorbis may not have enough data to work with and you will -// need to give it the same data again PLUS more. Note that the Vorbis -// specification does not bound the size of an individual frame. - -extern stb_vorbis *stb_vorbis_open_pushdata( - unsigned char *datablock, int datablock_length_in_bytes, - int *datablock_memory_consumed_in_bytes, - int *error, - stb_vorbis_alloc *alloc_buffer); -// create a vorbis decoder by passing in the initial data block containing -// the ogg&vorbis headers (you don't need to do parse them, just provide -// the first N bytes of the file--you're told if it's not enough, see below) -// on success, returns an stb_vorbis *, does not set error, returns the amount of -// data parsed/consumed on this call in *datablock_memory_consumed_in_bytes; -// on failure, returns NULL on error and sets *error, does not change *datablock_memory_consumed -// if returns NULL and *error is VORBIS_need_more_data, then the input block was -// incomplete and you need to pass in a larger block from the start of the file - -extern int stb_vorbis_decode_frame_pushdata( - stb_vorbis *f, unsigned char *datablock, int datablock_length_in_bytes, - int *channels, // place to write number of float * buffers - float ***output, // place to write float ** array of float * buffers - int *samples // place to write number of output samples - ); -// decode a frame of audio sample data if possible from the passed-in data block -// -// return value: number of bytes we used from datablock -// possible cases: -// 0 bytes used, 0 samples output (need more data) -// N bytes used, 0 samples output (resynching the stream, keep going) -// N bytes used, M samples output (one frame of data) -// note that after opening a file, you will ALWAYS get one N-bytes,0-sample -// frame, because Vorbis always "discards" the first frame. -// -// Note that on resynch, stb_vorbis will rarely consume all of the buffer, -// instead only datablock_length_in_bytes-3 or less. This is because it wants -// to avoid missing parts of a page header if they cross a datablock boundary, -// without writing state-machiney code to record a partial detection. -// -// The number of channels returned are stored in *channels (which can be -// NULL--it is always the same as the number of channels reported by -// get_info). *output will contain an array of float* buffers, one per -// channel. In other words, (*output)[0][0] contains the first sample from -// the first channel, and (*output)[1][0] contains the first sample from -// the second channel. - -extern void stb_vorbis_flush_pushdata(stb_vorbis *f); -// inform stb_vorbis that your next datablock will not be contiguous with -// previous ones (e.g. you've seeked in the data); future attempts to decode -// frames will cause stb_vorbis to resynchronize (as noted above), and -// once it sees a valid Ogg page (typically 4-8KB, as large as 64KB), it -// will begin decoding the _next_ frame. -// -// if you want to seek using pushdata, you need to seek in your file, then -// call stb_vorbis_flush_pushdata(), then start calling decoding, then once -// decoding is returning you data, call stb_vorbis_get_sample_offset, and -// if you don't like the result, seek your file again and repeat. -#endif - - -////////// PULLING INPUT API - -#ifndef STB_VORBIS_NO_PULLDATA_API -// This API assumes stb_vorbis is allowed to pull data from a source-- -// either a block of memory containing the _entire_ vorbis stream, or a -// FILE * that you or it create, or possibly some other reading mechanism -// if you go modify the source to replace the FILE * case with some kind -// of callback to your code. (But if you don't support seeking, you may -// just want to go ahead and use pushdata.) - -#if !defined(STB_VORBIS_NO_STDIO) && !defined(STB_VORBIS_NO_INTEGER_CONVERSION) -extern int stb_vorbis_decode_filename(char *filename, int *channels, short **output); -#endif -extern int stb_vorbis_decode_memory(unsigned char *mem, int len, int *channels, short **output); -// decode an entire file and output the data interleaved into a malloc()ed -// buffer stored in *output. The return value is the number of samples -// decoded, or -1 if the file could not be opened or was not an ogg vorbis file. -// When you're done with it, just free() the pointer returned in *output. - -extern stb_vorbis * stb_vorbis_open_memory(unsigned char *data, int len, - int *error, stb_vorbis_alloc *alloc_buffer); -// create an ogg vorbis decoder from an ogg vorbis stream in memory (note -// this must be the entire stream!). on failure, returns NULL and sets *error - -#ifndef STB_VORBIS_NO_STDIO -extern stb_vorbis * stb_vorbis_open_filename(char *filename, - int *error, stb_vorbis_alloc *alloc_buffer); -// create an ogg vorbis decoder from a filename via fopen(). on failure, -// returns NULL and sets *error (possibly to VORBIS_file_open_failure). - -extern stb_vorbis * stb_vorbis_open_file(FILE *f, int close_handle_on_close, - int *error, stb_vorbis_alloc *alloc_buffer); -// create an ogg vorbis decoder from an open FILE *, looking for a stream at -// the _current_ seek point (ftell). on failure, returns NULL and sets *error. -// note that stb_vorbis must "own" this stream; if you seek it in between -// calls to stb_vorbis, it will become confused. Morever, if you attempt to -// perform stb_vorbis_seek_*() operations on this file, it will assume it -// owns the _entire_ rest of the file after the start point. Use the next -// function, stb_vorbis_open_file_section(), to limit it. - -extern stb_vorbis * stb_vorbis_open_file_section(FILE *f, int close_handle_on_close, - int *error, stb_vorbis_alloc *alloc_buffer, unsigned int len); -// create an ogg vorbis decoder from an open FILE *, looking for a stream at -// the _current_ seek point (ftell); the stream will be of length 'len' bytes. -// on failure, returns NULL and sets *error. note that stb_vorbis must "own" -// this stream; if you seek it in between calls to stb_vorbis, it will become -// confused. -#endif - -extern int stb_vorbis_seek_frame(stb_vorbis *f, unsigned int sample_number); -extern int stb_vorbis_seek(stb_vorbis *f, unsigned int sample_number); -// NOT WORKING YET -// these functions seek in the Vorbis file to (approximately) 'sample_number'. -// after calling seek_frame(), the next call to get_frame_*() will include -// the specified sample. after calling stb_vorbis_seek(), the next call to -// stb_vorbis_get_samples_* will start with the specified sample. If you -// do not need to seek to EXACTLY the target sample when using get_samples_*, -// you can also use seek_frame(). - -extern void stb_vorbis_seek_start(stb_vorbis *f); -// this function is equivalent to stb_vorbis_seek(f,0), but it -// actually works - -extern unsigned int stb_vorbis_stream_length_in_samples(stb_vorbis *f); -extern float stb_vorbis_stream_length_in_seconds(stb_vorbis *f); -// these functions return the total length of the vorbis stream - -extern int stb_vorbis_get_frame_float(stb_vorbis *f, int *channels, float ***output); -// decode the next frame and return the number of samples. the number of -// channels returned are stored in *channels (which can be NULL--it is always -// the same as the number of channels reported by get_info). *output will -// contain an array of float* buffers, one per channel. These outputs will -// be overwritten on the next call to stb_vorbis_get_frame_*. -// -// You generally should not intermix calls to stb_vorbis_get_frame_*() -// and stb_vorbis_get_samples_*(), since the latter calls the former. - -#ifndef STB_VORBIS_NO_INTEGER_CONVERSION -extern int stb_vorbis_get_frame_short_interleaved(stb_vorbis *f, int num_c, short *buffer, int num_shorts); -extern int stb_vorbis_get_frame_short (stb_vorbis *f, int num_c, short **buffer, int num_samples); -#endif -// decode the next frame and return the number of samples per channel. the -// data is coerced to the number of channels you request according to the -// channel coercion rules (see below). You must pass in the size of your -// buffer(s) so that stb_vorbis will not overwrite the end of the buffer. -// The maximum buffer size needed can be gotten from get_info(); however, -// the Vorbis I specification implies an absolute maximum of 4096 samples -// per channel. Note that for interleaved data, you pass in the number of -// shorts (the size of your array), but the return value is the number of -// samples per channel, not the total number of samples. - -// Channel coercion rules: -// Let M be the number of channels requested, and N the number of channels present, -// and Cn be the nth channel; let stereo L be the sum of all L and center channels, -// and stereo R be the sum of all R and center channels (channel assignment from the -// vorbis spec). -// M N output -// 1 k sum(Ck) for all k -// 2 * stereo L, stereo R -// k l k > l, the first l channels, then 0s -// k l k <= l, the first k channels -// Note that this is not _good_ surround etc. mixing at all! It's just so -// you get something useful. - -extern int stb_vorbis_get_samples_float_interleaved(stb_vorbis *f, int channels, float *buffer, int num_floats); -extern int stb_vorbis_get_samples_float(stb_vorbis *f, int channels, float **buffer, int num_samples); -// gets num_samples samples, not necessarily on a frame boundary--this requires -// buffering so you have to supply the buffers. DOES NOT APPLY THE COERCION RULES. -// Returns the number of samples stored per channel; it may be less than requested -// at the end of the file. If there are no more samples in the file, returns 0. - -#ifndef STB_VORBIS_NO_INTEGER_CONVERSION -extern int stb_vorbis_get_samples_short_interleaved(stb_vorbis *f, int channels, short *buffer, int num_shorts); -extern int stb_vorbis_get_samples_short(stb_vorbis *f, int channels, short **buffer, int num_samples); -#endif -// gets num_samples samples, not necessarily on a frame boundary--this requires -// buffering so you have to supply the buffers. Applies the coercion rules above -// to produce 'channels' channels. Returns the number of samples stored per channel; -// it may be less than requested at the end of the file. If there are no more -// samples in the file, returns 0. - -#endif - -//////// ERROR CODES - -enum STBVorbisError -{ - VORBIS__no_error, - - VORBIS_need_more_data=1, // not a real error - - VORBIS_invalid_api_mixing, // can't mix API modes - VORBIS_outofmem, // not enough memory - VORBIS_feature_not_supported, // uses floor 0 - VORBIS_too_many_channels, // STB_VORBIS_MAX_CHANNELS is too small - VORBIS_file_open_failure, // fopen() failed - VORBIS_seek_without_length, // can't seek in unknown-length file - - VORBIS_unexpected_eof=10, // file is truncated? - VORBIS_seek_invalid, // seek past EOF - - // decoding errors (corrupt/invalid stream) -- you probably - // don't care about the exact details of these - - // vorbis errors: - VORBIS_invalid_setup=20, - VORBIS_invalid_stream, - - // ogg errors: - VORBIS_missing_capture_pattern=30, - VORBIS_invalid_stream_structure_version, - VORBIS_continued_packet_flag_invalid, - VORBIS_incorrect_stream_serial_number, - VORBIS_invalid_first_page, - VORBIS_bad_packet_type, - VORBIS_cant_find_last_page, - VORBIS_seek_failed -}; - - -#ifdef __cplusplus -} -#endif - -#endif // STB_VORBIS_INCLUDE_STB_VORBIS_H -// -// HEADER ENDS HERE -// -//////////////////////////////////////////////////////////////////////////////