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update SDL_mixer to SDL3_mixer 3.2.0

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This commit is contained in:
Sven Balzer
2026-04-02 16:52:07 +02:00
parent 05b19704f8
commit 8a5caf5c0d
1202 changed files with 366156 additions and 152445 deletions
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libs/SDL3_mixer/VisualC/external/include/gme/gme.h
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/* Game music emulator library C interface (also usable from C++) */
/* Game_Music_Emu 0.6.4 */
#ifndef GME_H
#define GME_H
#ifdef __cplusplus
extern "C" {
#endif
#define GME_VERSION 0x000604 /* 1 byte major, 1 byte minor, 1 byte patch-level */
/* Error string returned by library functions, or NULL if no error (success) */
typedef const char* gme_err_t;
/* First parameter of most gme_ functions is a pointer to the Music_Emu */
typedef struct Music_Emu Music_Emu;
/* Setup compiler defines useful for exporting required public API symbols in gme.cpp */
#ifndef BLARGG_EXPORT
#if defined (_WIN32)
#if defined(BLARGG_BUILD_DLL)
#define BLARGG_EXPORT __declspec(dllexport)
#else
#define BLARGG_EXPORT /* Leave blank: friendly with both static and shared linking. */
#endif
#elif defined (LIBGME_VISIBILITY)
#define BLARGG_EXPORT __attribute__((visibility ("default")))
#else
#define BLARGG_EXPORT
#endif
#endif
/******** Basic operations ********/
/* Create emulator and load game music file/data into it. Sets *out to new emulator. */
BLARGG_EXPORT gme_err_t gme_open_file( const char path [], Music_Emu** out, int sample_rate );
/* Number of tracks available */
BLARGG_EXPORT int gme_track_count( Music_Emu const* );
/* Start a track, where 0 is the first track */
BLARGG_EXPORT gme_err_t gme_start_track( Music_Emu*, int index );
/* Generate 'count' 16-bit signed samples info 'out'. Output is in stereo. */
BLARGG_EXPORT gme_err_t gme_play( Music_Emu*, int count, short out [] );
/* Finish using emulator and free memory */
BLARGG_EXPORT void gme_delete( Music_Emu* );
/******** Track position/length ********/
/* Set time to start fading track out. Once fade ends track_ended() returns true.
Fade time can be changed while track is playing. */
BLARGG_EXPORT void gme_set_fade( Music_Emu*, int start_msec );
/** See gme_set_fade.
* @since 0.6.4
*/
BLARGG_EXPORT void gme_set_fade_msecs( Music_Emu*, int start_msec, int length_msecs );
/**
* If do_autoload_limit is nonzero, then automatically load track length
* metadata (if present) and terminate playback once the track length has been
* reached. Otherwise playback will continue for an arbitrary period of time
* until a prolonged period of silence is detected.
*
* Not all individual emulators support this setting.
*
* By default, playback limits are loaded and applied.
*
* @since 0.6.3
*/
BLARGG_EXPORT void gme_set_autoload_playback_limit( Music_Emu *, int do_autoload_limit );
/** See gme_set_autoload_playback_limit.
* (This was actually added in 0.6.3, but wasn't exported because of a typo.)
* @since 0.6.4
*/
BLARGG_EXPORT int gme_autoload_playback_limit( Music_Emu const* );
/* True if a track has reached its end */
BLARGG_EXPORT int gme_track_ended( Music_Emu const* );
/* Number of milliseconds (1000 = one second) played since beginning of track */
BLARGG_EXPORT int gme_tell( Music_Emu const* );
/* Number of samples generated since beginning of track */
BLARGG_EXPORT int gme_tell_samples( Music_Emu const* );
/* Seek to new time in track. Seeking backwards or far forward can take a while. */
BLARGG_EXPORT gme_err_t gme_seek( Music_Emu*, int msec );
/* Equivalent to restarting track then skipping n samples */
BLARGG_EXPORT gme_err_t gme_seek_samples( Music_Emu*, int n );
/******** Informational ********/
/* If you only need track information from a music file, pass gme_info_only for
sample_rate to open/load. */
enum { gme_info_only = -1 };
/* Most recent warning string, or NULL if none. Clears current warning after returning.
Warning is also cleared when loading a file and starting a track. */
BLARGG_EXPORT const char* gme_warning( Music_Emu* );
/* Load m3u playlist file (must be done after loading music) */
BLARGG_EXPORT gme_err_t gme_load_m3u( Music_Emu*, const char path [] );
/* Clear any loaded m3u playlist and any internal playlist that the music format
supports (NSFE for example). */
BLARGG_EXPORT void gme_clear_playlist( Music_Emu* );
/* Gets information for a particular track (length, name, author, etc.).
Must be freed after use. */
typedef struct gme_info_t gme_info_t;
BLARGG_EXPORT gme_err_t gme_track_info( Music_Emu const*, gme_info_t** out, int track );
/* Frees track information */
BLARGG_EXPORT void gme_free_info( gme_info_t* );
struct gme_info_t
{
/* times in milliseconds; -1 if unknown */
int length; /* total length, if file specifies it */
int intro_length; /* length of song up to looping section */
int loop_length; /* length of looping section */
/* Length if available, otherwise intro_length+loop_length*2 if available,
otherwise a default of 150000 (2.5 minutes). */
int play_length;
/* fade length in milliseconds; -1 if unknown */
int fade_length;
int i5,i6,i7,i8,i9,i10,i11,i12,i13,i14,i15; /* reserved */
/* empty string ("") if not available */
const char* system;
const char* game;
const char* song;
const char* author;
const char* copyright;
const char* comment;
const char* dumper;
const char *s7,*s8,*s9,*s10,*s11,*s12,*s13,*s14,*s15; /* reserved */
};
/******** Advanced playback ********/
/* Adjust stereo echo depth, where 0.0 = off and 1.0 = maximum. Has no effect for
GYM, SPC, and Sega Genesis VGM music */
BLARGG_EXPORT void gme_set_stereo_depth( Music_Emu*, double depth );
/* Disable automatic end-of-track detection and skipping of silence at beginning
if ignore is true */
BLARGG_EXPORT void gme_ignore_silence( Music_Emu*, int ignore );
/* Adjust song tempo, where 1.0 = normal, 0.5 = half speed, 2.0 = double speed.
Track length as returned by track_info() assumes a tempo of 1.0. */
BLARGG_EXPORT void gme_set_tempo( Music_Emu*, double tempo );
/* Number of voices used by currently loaded file */
BLARGG_EXPORT int gme_voice_count( Music_Emu const* );
/* Name of voice i, from 0 to gme_voice_count() - 1 */
BLARGG_EXPORT const char* gme_voice_name( Music_Emu const*, int i );
/* Mute/unmute voice i, where voice 0 is first voice */
BLARGG_EXPORT void gme_mute_voice( Music_Emu*, int index, int mute );
/* Set muting state of all voices at once using a bit mask, where -1 mutes all
voices, 0 unmutes them all, 0x01 mutes just the first voice, etc. */
BLARGG_EXPORT void gme_mute_voices( Music_Emu*, int muting_mask );
/* Disable/Enable echo effect for SPC files */
/* Available since 0.6.4 */
BLARGG_EXPORT void gme_disable_echo( Music_Emu*, int disable );
/* Frequency equalizer parameters (see gme.txt) */
/* Implementers: If modified, also adjust Music_Emu::make_equalizer as needed */
typedef struct gme_equalizer_t
{
double treble; /* -50.0 = muffled, 0 = flat, +5.0 = extra-crisp */
double bass; /* 1 = full bass, 90 = average, 16000 = almost no bass */
double d2,d3,d4,d5,d6,d7,d8,d9; /* reserved */
} gme_equalizer_t;
/* Get current frequency equalizater parameters */
BLARGG_EXPORT void gme_equalizer( Music_Emu const*, gme_equalizer_t* out );
/* Change frequency equalizer parameters */
BLARGG_EXPORT void gme_set_equalizer( Music_Emu*, gme_equalizer_t const* eq );
/* Enables/disables most accurate sound emulation options */
BLARGG_EXPORT void gme_enable_accuracy( Music_Emu*, int enabled );
/******** Game music types ********/
/* Music file type identifier. Can also hold NULL. */
typedef const struct gme_type_t_* gme_type_t;
/* Emulator type constants for each supported file type */
extern BLARGG_EXPORT const gme_type_t
gme_ay_type,
gme_gbs_type,
gme_gym_type,
gme_hes_type,
gme_kss_type,
gme_nsf_type,
gme_nsfe_type,
gme_sap_type,
gme_spc_type,
gme_vgm_type,
gme_vgz_type;
/* Type of this emulator */
BLARGG_EXPORT gme_type_t gme_type( Music_Emu const* );
/* Pointer to array of all music types, with NULL entry at end. Allows a player linked
to this library to support new music types without having to be updated. */
BLARGG_EXPORT gme_type_t const* gme_type_list();
/* Name of game system for this music file type */
BLARGG_EXPORT const char* gme_type_system( gme_type_t );
/* True if this music file type supports multiple tracks */
BLARGG_EXPORT int gme_type_multitrack( gme_type_t );
/* whether the pcm output retrieved by gme_play() will have all 8 voices rendered to their
* individual stereo channel or (if false) these voices get mixed into one single stereo channel
* @since 0.6.3 */
BLARGG_EXPORT int gme_multi_channel( Music_Emu const* );
/******** Advanced file loading ********/
/* Error returned if file type is not supported */
extern BLARGG_EXPORT const char* const gme_wrong_file_type;
/* Same as gme_open_file(), but uses file data already in memory. Makes copy of data.
* The resulting Music_Emu object will be set to single channel mode. */
BLARGG_EXPORT gme_err_t gme_open_data( void const* data, long size, Music_Emu** out, int sample_rate );
/* Determine likely game music type based on first four bytes of file. Returns
string containing proper file suffix (i.e. "NSF", "SPC", etc.) or "" if
file header is not recognized. */
BLARGG_EXPORT const char* gme_identify_header( void const* header );
/* Get corresponding music type for file path or extension passed in. */
BLARGG_EXPORT gme_type_t gme_identify_extension( const char path_or_extension [] );
/**
* Get typical file extension for a given music type. This is not a replacement
* for a file content identification library (but see gme_identify_header).
*
* @since 0.6.3
*/
BLARGG_EXPORT const char* gme_type_extension( gme_type_t music_type );
/* Determine file type based on file's extension or header (if extension isn't recognized).
Sets *type_out to type, or 0 if unrecognized or error. */
BLARGG_EXPORT gme_err_t gme_identify_file( const char path [], gme_type_t* type_out );
/* Create new emulator and set sample rate. Returns NULL if out of memory. If you only need
track information, pass gme_info_only for sample_rate. */
BLARGG_EXPORT Music_Emu* gme_new_emu( gme_type_t, int sample_rate );
/* Create new multichannel emulator and set sample rate. Returns NULL if out of memory.
* If you only need track information, pass gme_info_only for sample_rate.
* (see gme_multi_channel for more information on multichannel support)
* @since 0.6.3
*/
BLARGG_EXPORT Music_Emu* gme_new_emu_multi_channel( gme_type_t, int sample_rate );
/* Load music file into emulator */
BLARGG_EXPORT gme_err_t gme_load_file( Music_Emu*, const char path [] );
/* Load music file from memory into emulator. Makes a copy of data passed. */
BLARGG_EXPORT gme_err_t gme_load_data( Music_Emu*, void const* data, long size );
/* Load multiple single-track music files from memory into emulator.
* @since 0.6.4
*/
BLARGG_EXPORT gme_err_t gme_load_tracks( Music_Emu* me,
void const* data, long* sizes, int count );
/* Return the fixed track count of an emu file type
* @since 0.6.4
*/
BLARGG_EXPORT int gme_fixed_track_count( gme_type_t );
/* Load music file using custom data reader function that will be called to
read file data. Most emulators load the entire file in one read call. */
typedef gme_err_t (*gme_reader_t)( void* your_data, void* out, int count );
BLARGG_EXPORT gme_err_t gme_load_custom( Music_Emu*, gme_reader_t, long file_size, void* your_data );
/* Load m3u playlist file from memory (must be done after loading music) */
BLARGG_EXPORT gme_err_t gme_load_m3u_data( Music_Emu*, void const* data, long size );
/******** User data ********/
/* Set/get pointer to data you want to associate with this emulator.
You can use this for whatever you want. */
BLARGG_EXPORT void gme_set_user_data( Music_Emu*, void* new_user_data );
BLARGG_EXPORT void* gme_user_data( Music_Emu const* );
/* Register cleanup function to be called when deleting emulator, or NULL to
clear it. Passes user_data to cleanup function. */
typedef void (*gme_user_cleanup_t)( void* user_data );
BLARGG_EXPORT void gme_set_user_cleanup( Music_Emu*, gme_user_cleanup_t func );
#ifdef __cplusplus
}
#endif
#endif
libs/SDL3_mixer/VisualC/external/include/ogg/config_types.h
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#ifndef __CONFIG_TYPES_H__
#define __CONFIG_TYPES_H__
/* these are filled in by configure or cmake*/
#define INCLUDE_INTTYPES_H 1
#define INCLUDE_STDINT_H 1
#define INCLUDE_SYS_TYPES_H 1
#if INCLUDE_INTTYPES_H
# include <inttypes.h>
#endif
#if INCLUDE_STDINT_H
# include <stdint.h>
#endif
#if INCLUDE_SYS_TYPES_H
# include <sys/types.h>
#endif
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef uint64_t ogg_uint64_t;
#endif
libs/SDL3_mixer/VisualC/external/include/ogg/ogg.h
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/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: toplevel libogg include
********************************************************************/
#ifndef _OGG_H
#define _OGG_H
#ifdef __cplusplus
extern "C" {
#endif
#include <stddef.h>
#include <ogg/os_types.h>
typedef struct {
void *iov_base;
size_t iov_len;
} ogg_iovec_t;
typedef struct {
long endbyte;
int endbit;
unsigned char *buffer;
unsigned char *ptr;
long storage;
} oggpack_buffer;
/* ogg_page is used to encapsulate the data in one Ogg bitstream page *****/
typedef struct {
unsigned char *header;
long header_len;
unsigned char *body;
long body_len;
} ogg_page;
/* ogg_stream_state contains the current encode/decode state of a logical
Ogg bitstream **********************************************************/
typedef struct {
unsigned char *body_data; /* bytes from packet bodies */
long body_storage; /* storage elements allocated */
long body_fill; /* elements stored; fill mark */
long body_returned; /* elements of fill returned */
int *lacing_vals; /* The values that will go to the segment table */
ogg_int64_t *granule_vals; /* granulepos values for headers. Not compact
this way, but it is simple coupled to the
lacing fifo */
long lacing_storage;
long lacing_fill;
long lacing_packet;
long lacing_returned;
unsigned char header[282]; /* working space for header encode */
int header_fill;
int e_o_s; /* set when we have buffered the last packet in the
logical bitstream */
int b_o_s; /* set after we've written the initial page
of a logical bitstream */
long serialno;
long pageno;
ogg_int64_t packetno; /* sequence number for decode; the framing
knows where there's a hole in the data,
but we need coupling so that the codec
(which is in a separate abstraction
layer) also knows about the gap */
ogg_int64_t granulepos;
} ogg_stream_state;
/* ogg_packet is used to encapsulate the data and metadata belonging
to a single raw Ogg/Vorbis packet *************************************/
typedef struct {
unsigned char *packet;
long bytes;
long b_o_s;
long e_o_s;
ogg_int64_t granulepos;
ogg_int64_t packetno; /* sequence number for decode; the framing
knows where there's a hole in the data,
but we need coupling so that the codec
(which is in a separate abstraction
layer) also knows about the gap */
} ogg_packet;
typedef struct {
unsigned char *data;
int storage;
int fill;
int returned;
int unsynced;
int headerbytes;
int bodybytes;
} ogg_sync_state;
/* Ogg BITSTREAM PRIMITIVES: bitstream ************************/
extern void oggpack_writeinit(oggpack_buffer *b);
extern int oggpack_writecheck(oggpack_buffer *b);
extern void oggpack_writetrunc(oggpack_buffer *b,long bits);
extern void oggpack_writealign(oggpack_buffer *b);
extern void oggpack_writecopy(oggpack_buffer *b,void *source,long bits);
extern void oggpack_reset(oggpack_buffer *b);
extern void oggpack_writeclear(oggpack_buffer *b);
extern void oggpack_readinit(oggpack_buffer *b,unsigned char *buf,int bytes);
extern void oggpack_write(oggpack_buffer *b,unsigned long value,int bits);
extern long oggpack_look(oggpack_buffer *b,int bits);
extern long oggpack_look1(oggpack_buffer *b);
extern void oggpack_adv(oggpack_buffer *b,int bits);
extern void oggpack_adv1(oggpack_buffer *b);
extern long oggpack_read(oggpack_buffer *b,int bits);
extern long oggpack_read1(oggpack_buffer *b);
extern long oggpack_bytes(oggpack_buffer *b);
extern long oggpack_bits(oggpack_buffer *b);
extern unsigned char *oggpack_get_buffer(oggpack_buffer *b);
extern void oggpackB_writeinit(oggpack_buffer *b);
extern int oggpackB_writecheck(oggpack_buffer *b);
extern void oggpackB_writetrunc(oggpack_buffer *b,long bits);
extern void oggpackB_writealign(oggpack_buffer *b);
extern void oggpackB_writecopy(oggpack_buffer *b,void *source,long bits);
extern void oggpackB_reset(oggpack_buffer *b);
extern void oggpackB_writeclear(oggpack_buffer *b);
extern void oggpackB_readinit(oggpack_buffer *b,unsigned char *buf,int bytes);
extern void oggpackB_write(oggpack_buffer *b,unsigned long value,int bits);
extern long oggpackB_look(oggpack_buffer *b,int bits);
extern long oggpackB_look1(oggpack_buffer *b);
extern void oggpackB_adv(oggpack_buffer *b,int bits);
extern void oggpackB_adv1(oggpack_buffer *b);
extern long oggpackB_read(oggpack_buffer *b,int bits);
extern long oggpackB_read1(oggpack_buffer *b);
extern long oggpackB_bytes(oggpack_buffer *b);
extern long oggpackB_bits(oggpack_buffer *b);
extern unsigned char *oggpackB_get_buffer(oggpack_buffer *b);
/* Ogg BITSTREAM PRIMITIVES: encoding **************************/
extern int ogg_stream_packetin(ogg_stream_state *os, ogg_packet *op);
extern int ogg_stream_iovecin(ogg_stream_state *os, ogg_iovec_t *iov,
int count, long e_o_s, ogg_int64_t granulepos);
extern int ogg_stream_pageout(ogg_stream_state *os, ogg_page *og);
extern int ogg_stream_pageout_fill(ogg_stream_state *os, ogg_page *og, int nfill);
extern int ogg_stream_flush(ogg_stream_state *os, ogg_page *og);
extern int ogg_stream_flush_fill(ogg_stream_state *os, ogg_page *og, int nfill);
/* Ogg BITSTREAM PRIMITIVES: decoding **************************/
extern int ogg_sync_init(ogg_sync_state *oy);
extern int ogg_sync_clear(ogg_sync_state *oy);
extern int ogg_sync_reset(ogg_sync_state *oy);
extern int ogg_sync_destroy(ogg_sync_state *oy);
extern int ogg_sync_check(ogg_sync_state *oy);
extern char *ogg_sync_buffer(ogg_sync_state *oy, long size);
extern int ogg_sync_wrote(ogg_sync_state *oy, long bytes);
extern long ogg_sync_pageseek(ogg_sync_state *oy,ogg_page *og);
extern int ogg_sync_pageout(ogg_sync_state *oy, ogg_page *og);
extern int ogg_stream_pagein(ogg_stream_state *os, ogg_page *og);
extern int ogg_stream_packetout(ogg_stream_state *os,ogg_packet *op);
extern int ogg_stream_packetpeek(ogg_stream_state *os,ogg_packet *op);
/* Ogg BITSTREAM PRIMITIVES: general ***************************/
extern int ogg_stream_init(ogg_stream_state *os,int serialno);
extern int ogg_stream_clear(ogg_stream_state *os);
extern int ogg_stream_reset(ogg_stream_state *os);
extern int ogg_stream_reset_serialno(ogg_stream_state *os,int serialno);
extern int ogg_stream_destroy(ogg_stream_state *os);
extern int ogg_stream_check(ogg_stream_state *os);
extern int ogg_stream_eos(ogg_stream_state *os);
extern void ogg_page_checksum_set(ogg_page *og);
extern int ogg_page_version(const ogg_page *og);
extern int ogg_page_continued(const ogg_page *og);
extern int ogg_page_bos(const ogg_page *og);
extern int ogg_page_eos(const ogg_page *og);
extern ogg_int64_t ogg_page_granulepos(const ogg_page *og);
extern int ogg_page_serialno(const ogg_page *og);
extern long ogg_page_pageno(const ogg_page *og);
extern int ogg_page_packets(const ogg_page *og);
extern void ogg_packet_clear(ogg_packet *op);
#ifdef __cplusplus
}
#endif
#endif /* _OGG_H */
libs/SDL3_mixer/VisualC/external/include/ogg/os_types.h
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/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: Define a consistent set of types on each platform.
********************************************************************/
#ifndef _OS_TYPES_H
#define _OS_TYPES_H
/* make it easy on the folks that want to compile the libs with a
different malloc than stdlib */
#define _ogg_malloc malloc
#define _ogg_calloc calloc
#define _ogg_realloc realloc
#define _ogg_free free
#if defined(_WIN32)
# if defined(__CYGWIN__)
# include <stdint.h>
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef uint64_t ogg_uint64_t;
# elif defined(__MINGW32__)
# include <sys/types.h>
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
# elif defined(__MWERKS__)
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
# else
# if defined(_MSC_VER) && (_MSC_VER >= 1800) /* MSVC 2013 and newer */
# include <stdint.h>
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef uint64_t ogg_uint64_t;
# else
/* MSVC/Borland */
typedef __int64 ogg_int64_t;
typedef __int32 ogg_int32_t;
typedef unsigned __int32 ogg_uint32_t;
typedef unsigned __int64 ogg_uint64_t;
typedef __int16 ogg_int16_t;
typedef unsigned __int16 ogg_uint16_t;
# endif
# endif
#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
# include <sys/types.h>
# include <stdint.h>
typedef int16_t ogg_int16_t;
typedef u_int16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef u_int32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef u_int64_t ogg_uint64_t;
#elif defined(__HAIKU__)
/* Haiku */
# include <sys/types.h>
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
#elif defined(__BEOS__)
/* Be */
# include <inttypes.h>
typedef int16_t ogg_int16_t;
typedef uint16_t ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef uint32_t ogg_uint32_t;
typedef int64_t ogg_int64_t;
typedef uint64_t ogg_uint64_t;
#elif defined (__EMX__)
/* OS/2 GCC */
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
#elif defined (DJGPP)
/* DJGPP */
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long ogg_int64_t;
typedef unsigned long long ogg_uint64_t;
#elif defined(R5900)
/* PS2 EE */
typedef long ogg_int64_t;
typedef unsigned long ogg_uint64_t;
typedef int ogg_int32_t;
typedef unsigned ogg_uint32_t;
typedef short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
#elif defined(__SYMBIAN32__)
/* Symbian GCC */
typedef signed short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef signed int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long int ogg_int64_t;
typedef unsigned long long int ogg_uint64_t;
#elif defined(__TMS320C6X__)
/* TI C64x compiler */
typedef signed short ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef signed int ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef long long int ogg_int64_t;
typedef unsigned long long int ogg_uint64_t;
#else
# include <ogg/config_types.h>
#endif
#endif /* _OS_TYPES_H */
libs/SDL3_mixer/VisualC/external/include/opus/opus.h
Vendored
+981
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@@ -0,0 +1,981 @@
/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
Written by Jean-Marc Valin and Koen Vos */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/**
* @file opus.h
* @brief Opus reference implementation API
*/
#ifndef OPUS_H
#define OPUS_H
#include <opus/opus_types.h>
#include <opus/opus_defines.h>
#ifdef __cplusplus
extern "C" {
#endif
/**
* @mainpage Opus
*
* The Opus codec is designed for interactive speech and audio transmission over the Internet.
* It is designed by the IETF Codec Working Group and incorporates technology from
* Skype's SILK codec and Xiph.Org's CELT codec.
*
* The Opus codec is designed to handle a wide range of interactive audio applications,
* including Voice over IP, videoconferencing, in-game chat, and even remote live music
* performances. It can scale from low bit-rate narrowband speech to very high quality
* stereo music. Its main features are:
* @li Sampling rates from 8 to 48 kHz
* @li Bit-rates from 6 kb/s to 510 kb/s
* @li Support for both constant bit-rate (CBR) and variable bit-rate (VBR)
* @li Audio bandwidth from narrowband to full-band
* @li Support for speech and music
* @li Support for mono and stereo
* @li Support for multichannel (up to 255 channels)
* @li Frame sizes from 2.5 ms to 60 ms
* @li Good loss robustness and packet loss concealment (PLC)
* @li Floating point and fixed-point implementation
*
* Documentation sections:
* @li @ref opus_encoder
* @li @ref opus_decoder
* @li @ref opus_repacketizer
* @li @ref opus_multistream
* @li @ref opus_libinfo
* @li @ref opus_custom
*/
/** @defgroup opus_encoder Opus Encoder
* @{
*
* @brief This page describes the process and functions used to encode Opus.
*
* Since Opus is a stateful codec, the encoding process starts with creating an encoder
* state. This can be done with:
*
* @code
* int error;
* OpusEncoder *enc;
* enc = opus_encoder_create(Fs, channels, application, &error);
* @endcode
*
* From this point, @c enc can be used for encoding an audio stream. An encoder state
* @b must @b not be used for more than one stream at the same time. Similarly, the encoder
* state @b must @b not be re-initialized for each frame.
*
* While opus_encoder_create() allocates memory for the state, it's also possible
* to initialize pre-allocated memory:
*
* @code
* int size;
* int error;
* OpusEncoder *enc;
* size = opus_encoder_get_size(channels);
* enc = malloc(size);
* error = opus_encoder_init(enc, Fs, channels, application);
* @endcode
*
* where opus_encoder_get_size() returns the required size for the encoder state. Note that
* future versions of this code may change the size, so no assuptions should be made about it.
*
* The encoder state is always continuous in memory and only a shallow copy is sufficient
* to copy it (e.g. memcpy())
*
* It is possible to change some of the encoder's settings using the opus_encoder_ctl()
* interface. All these settings already default to the recommended value, so they should
* only be changed when necessary. The most common settings one may want to change are:
*
* @code
* opus_encoder_ctl(enc, OPUS_SET_BITRATE(bitrate));
* opus_encoder_ctl(enc, OPUS_SET_COMPLEXITY(complexity));
* opus_encoder_ctl(enc, OPUS_SET_SIGNAL(signal_type));
* @endcode
*
* where
*
* @arg bitrate is in bits per second (b/s)
* @arg complexity is a value from 1 to 10, where 1 is the lowest complexity and 10 is the highest
* @arg signal_type is either OPUS_AUTO (default), OPUS_SIGNAL_VOICE, or OPUS_SIGNAL_MUSIC
*
* See @ref opus_encoderctls and @ref opus_genericctls for a complete list of parameters that can be set or queried. Most parameters can be set or changed at any time during a stream.
*
* To encode a frame, opus_encode() or opus_encode_float() must be called with exactly one frame (2.5, 5, 10, 20, 40 or 60 ms) of audio data:
* @code
* len = opus_encode(enc, audio_frame, frame_size, packet, max_packet);
* @endcode
*
* where
* <ul>
* <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li>
* <li>frame_size is the duration of the frame in samples (per channel)</li>
* <li>packet is the byte array to which the compressed data is written</li>
* <li>max_packet is the maximum number of bytes that can be written in the packet (4000 bytes is recommended).
* Do not use max_packet to control VBR target bitrate, instead use the #OPUS_SET_BITRATE CTL.</li>
* </ul>
*
* opus_encode() and opus_encode_float() return the number of bytes actually written to the packet.
* The return value <b>can be negative</b>, which indicates that an error has occurred. If the return value
* is 2 bytes or less, then the packet does not need to be transmitted (DTX).
*
* Once the encoder state if no longer needed, it can be destroyed with
*
* @code
* opus_encoder_destroy(enc);
* @endcode
*
* If the encoder was created with opus_encoder_init() rather than opus_encoder_create(),
* then no action is required aside from potentially freeing the memory that was manually
* allocated for it (calling free(enc) for the example above)
*
*/
/** Opus encoder state.
* This contains the complete state of an Opus encoder.
* It is position independent and can be freely copied.
* @see opus_encoder_create,opus_encoder_init
*/
typedef struct OpusEncoder OpusEncoder;
/** Gets the size of an <code>OpusEncoder</code> structure.
* @param[in] channels <tt>int</tt>: Number of channels.
* This must be 1 or 2.
* @returns The size in bytes.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_encoder_get_size(int channels);
/**
*/
/** Allocates and initializes an encoder state.
* There are three coding modes:
*
* @ref OPUS_APPLICATION_VOIP gives best quality at a given bitrate for voice
* signals. It enhances the input signal by high-pass filtering and
* emphasizing formants and harmonics. Optionally it includes in-band
* forward error correction to protect against packet loss. Use this
* mode for typical VoIP applications. Because of the enhancement,
* even at high bitrates the output may sound different from the input.
*
* @ref OPUS_APPLICATION_AUDIO gives best quality at a given bitrate for most
* non-voice signals like music. Use this mode for music and mixed
* (music/voice) content, broadcast, and applications requiring less
* than 15 ms of coding delay.
*
* @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY configures low-delay mode that
* disables the speech-optimized mode in exchange for slightly reduced delay.
* This mode can only be set on an newly initialized or freshly reset encoder
* because it changes the codec delay.
*
* This is useful when the caller knows that the speech-optimized modes will not be needed (use with caution).
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
* @param [in] application <tt>int</tt>: Coding mode (one of @ref OPUS_APPLICATION_VOIP, @ref OPUS_APPLICATION_AUDIO, or @ref OPUS_APPLICATION_RESTRICTED_LOWDELAY)
* @param [out] error <tt>int*</tt>: @ref opus_errorcodes
* @note Regardless of the sampling rate and number channels selected, the Opus encoder
* can switch to a lower audio bandwidth or number of channels if the bitrate
* selected is too low. This also means that it is safe to always use 48 kHz stereo input
* and let the encoder optimize the encoding.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusEncoder *opus_encoder_create(
opus_int32 Fs,
int channels,
int application,
int *error
);
/** Initializes a previously allocated encoder state
* The memory pointed to by st must be at least the size returned by opus_encoder_get_size().
* This is intended for applications which use their own allocator instead of malloc.
* @see opus_encoder_create(),opus_encoder_get_size()
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate of input signal (Hz)
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) in input signal
* @param [in] application <tt>int</tt>: Coding mode (one of OPUS_APPLICATION_VOIP, OPUS_APPLICATION_AUDIO, or OPUS_APPLICATION_RESTRICTED_LOWDELAY)
* @retval #OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_encoder_init(
OpusEncoder *st,
opus_int32 Fs,
int channels,
int application
) OPUS_ARG_NONNULL(1);
/** Encodes an Opus frame.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16)
* @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
* input signal.
* This must be an Opus frame size for
* the encoder's sampling rate.
* For example, at 48 kHz the permitted
* values are 120, 240, 480, 960, 1920,
* and 2880.
* Passing in a duration of less than
* 10 ms (480 samples at 48 kHz) will
* prevent the encoder from using the LPC
* or hybrid modes.
* @param [out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode(
OpusEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes an Opus frame from floating point input.
* @param [in] st <tt>OpusEncoder*</tt>: Encoder state
* @param [in] pcm <tt>float*</tt>: Input in float format (interleaved if 2 channels), with a normal range of +/-1.0.
* Samples with a range beyond +/-1.0 are supported but will
* be clipped by decoders using the integer API and should
* only be used if it is known that the far end supports
* extended dynamic range.
* length is frame_size*channels*sizeof(float)
* @param [in] frame_size <tt>int</tt>: Number of samples per channel in the
* input signal.
* This must be an Opus frame size for
* the encoder's sampling rate.
* For example, at 48 kHz the permitted
* values are 120, 240, 480, 960, 1920,
* and 2880.
* Passing in a duration of less than
* 10 ms (480 samples at 48 kHz) will
* prevent the encoder from using the LPC
* or hybrid modes.
* @param [out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_encode_float(
OpusEncoder *st,
const float *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Frees an <code>OpusEncoder</code> allocated by opus_encoder_create().
* @param[in] st <tt>OpusEncoder*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_encoder_destroy(OpusEncoder *st);
/** Perform a CTL function on an Opus encoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @param st <tt>OpusEncoder*</tt>: Encoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls or
* @ref opus_encoderctls.
* @see opus_genericctls
* @see opus_encoderctls
*/
OPUS_EXPORT int opus_encoder_ctl(OpusEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
/** @defgroup opus_decoder Opus Decoder
* @{
*
* @brief This page describes the process and functions used to decode Opus.
*
* The decoding process also starts with creating a decoder
* state. This can be done with:
* @code
* int error;
* OpusDecoder *dec;
* dec = opus_decoder_create(Fs, channels, &error);
* @endcode
* where
* @li Fs is the sampling rate and must be 8000, 12000, 16000, 24000, or 48000
* @li channels is the number of channels (1 or 2)
* @li error will hold the error code in case of failure (or #OPUS_OK on success)
* @li the return value is a newly created decoder state to be used for decoding
*
* While opus_decoder_create() allocates memory for the state, it's also possible
* to initialize pre-allocated memory:
* @code
* int size;
* int error;
* OpusDecoder *dec;
* size = opus_decoder_get_size(channels);
* dec = malloc(size);
* error = opus_decoder_init(dec, Fs, channels);
* @endcode
* where opus_decoder_get_size() returns the required size for the decoder state. Note that
* future versions of this code may change the size, so no assuptions should be made about it.
*
* The decoder state is always continuous in memory and only a shallow copy is sufficient
* to copy it (e.g. memcpy())
*
* To decode a frame, opus_decode() or opus_decode_float() must be called with a packet of compressed audio data:
* @code
* frame_size = opus_decode(dec, packet, len, decoded, max_size, 0);
* @endcode
* where
*
* @li packet is the byte array containing the compressed data
* @li len is the exact number of bytes contained in the packet
* @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float())
* @li max_size is the max duration of the frame in samples (per channel) that can fit into the decoded_frame array
*
* opus_decode() and opus_decode_float() return the number of samples (per channel) decoded from the packet.
* If that value is negative, then an error has occurred. This can occur if the packet is corrupted or if the audio
* buffer is too small to hold the decoded audio.
*
* Opus is a stateful codec with overlapping blocks and as a result Opus
* packets are not coded independently of each other. Packets must be
* passed into the decoder serially and in the correct order for a correct
* decode. Lost packets can be replaced with loss concealment by calling
* the decoder with a null pointer and zero length for the missing packet.
*
* A single codec state may only be accessed from a single thread at
* a time and any required locking must be performed by the caller. Separate
* streams must be decoded with separate decoder states and can be decoded
* in parallel unless the library was compiled with NONTHREADSAFE_PSEUDOSTACK
* defined.
*
*/
/** Opus decoder state.
* This contains the complete state of an Opus decoder.
* It is position independent and can be freely copied.
* @see opus_decoder_create,opus_decoder_init
*/
typedef struct OpusDecoder OpusDecoder;
/** Gets the size of an <code>OpusDecoder</code> structure.
* @param [in] channels <tt>int</tt>: Number of channels.
* This must be 1 or 2.
* @returns The size in bytes.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_size(int channels);
/** Allocates and initializes a decoder state.
* @param [in] Fs <tt>opus_int32</tt>: Sample rate to decode at (Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
* @param [out] error <tt>int*</tt>: #OPUS_OK Success or @ref opus_errorcodes
*
* Internally Opus stores data at 48000 Hz, so that should be the default
* value for Fs. However, the decoder can efficiently decode to buffers
* at 8, 12, 16, and 24 kHz so if for some reason the caller cannot use
* data at the full sample rate, or knows the compressed data doesn't
* use the full frequency range, it can request decoding at a reduced
* rate. Likewise, the decoder is capable of filling in either mono or
* interleaved stereo pcm buffers, at the caller's request.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusDecoder *opus_decoder_create(
opus_int32 Fs,
int channels,
int *error
);
/** Initializes a previously allocated decoder state.
* The state must be at least the size returned by opus_decoder_get_size().
* This is intended for applications which use their own allocator instead of malloc. @see opus_decoder_create,opus_decoder_get_size
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state.
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate to decode to (Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param [in] channels <tt>int</tt>: Number of channels (1 or 2) to decode
* @retval #OPUS_OK Success or @ref opus_errorcodes
*/
OPUS_EXPORT int opus_decoder_init(
OpusDecoder *st,
opus_int32 Fs,
int channels
) OPUS_ARG_NONNULL(1);
/** Decode an Opus packet.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>opus_int32</tt>: Number of bytes in payload*
* @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(opus_int16)
* @param [in] frame_size Number of samples per channel of available space in \a pcm.
* If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
* not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
* then frame_size needs to be exactly the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
* FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
* @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
* decoded. If no such data is available, the frame is decoded as if it were lost.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode(
OpusDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int16 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode an Opus packet with floating point output.
* @param [in] st <tt>OpusDecoder*</tt>: Decoder state
* @param [in] data <tt>char*</tt>: Input payload. Use a NULL pointer to indicate packet loss
* @param [in] len <tt>opus_int32</tt>: Number of bytes in payload
* @param [out] pcm <tt>float*</tt>: Output signal (interleaved if 2 channels). length
* is frame_size*channels*sizeof(float)
* @param [in] frame_size Number of samples per channel of available space in \a pcm.
* If this is less than the maximum packet duration (120ms; 5760 for 48kHz), this function will
* not be capable of decoding some packets. In the case of PLC (data==NULL) or FEC (decode_fec=1),
* then frame_size needs to be exactly the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the next incoming packet. For the PLC and
* FEC cases, frame_size <b>must</b> be a multiple of 2.5 ms.
* @param [in] decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band forward error correction data be
* decoded. If no such data is available the frame is decoded as if it were lost.
* @returns Number of decoded samples or @ref opus_errorcodes
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decode_float(
OpusDecoder *st,
const unsigned char *data,
opus_int32 len,
float *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on an Opus decoder.
*
* Generally the request and subsequent arguments are generated
* by a convenience macro.
* @param st <tt>OpusDecoder*</tt>: Decoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls or
* @ref opus_decoderctls.
* @see opus_genericctls
* @see opus_decoderctls
*/
OPUS_EXPORT int opus_decoder_ctl(OpusDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/** Frees an <code>OpusDecoder</code> allocated by opus_decoder_create().
* @param[in] st <tt>OpusDecoder*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_decoder_destroy(OpusDecoder *st);
/** Parse an opus packet into one or more frames.
* Opus_decode will perform this operation internally so most applications do
* not need to use this function.
* This function does not copy the frames, the returned pointers are pointers into
* the input packet.
* @param [in] data <tt>char*</tt>: Opus packet to be parsed
* @param [in] len <tt>opus_int32</tt>: size of data
* @param [out] out_toc <tt>char*</tt>: TOC pointer
* @param [out] frames <tt>char*[48]</tt> encapsulated frames
* @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames
* @param [out] payload_offset <tt>int*</tt>: returns the position of the payload within the packet (in bytes)
* @returns number of frames
*/
OPUS_EXPORT int opus_packet_parse(
const unsigned char *data,
opus_int32 len,
unsigned char *out_toc,
const unsigned char *frames[48],
opus_int16 size[48],
int *payload_offset
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5);
/** Gets the bandwidth of an Opus packet.
* @param [in] data <tt>char*</tt>: Opus packet
* @retval OPUS_BANDWIDTH_NARROWBAND Narrowband (4kHz bandpass)
* @retval OPUS_BANDWIDTH_MEDIUMBAND Mediumband (6kHz bandpass)
* @retval OPUS_BANDWIDTH_WIDEBAND Wideband (8kHz bandpass)
* @retval OPUS_BANDWIDTH_SUPERWIDEBAND Superwideband (12kHz bandpass)
* @retval OPUS_BANDWIDTH_FULLBAND Fullband (20kHz bandpass)
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_bandwidth(const unsigned char *data) OPUS_ARG_NONNULL(1);
/** Gets the number of samples per frame from an Opus packet.
* @param [in] data <tt>char*</tt>: Opus packet.
* This must contain at least one byte of
* data.
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
* This must be a multiple of 400, or
* inaccurate results will be returned.
* @returns Number of samples per frame.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_samples_per_frame(const unsigned char *data, opus_int32 Fs) OPUS_ARG_NONNULL(1);
/** Gets the number of channels from an Opus packet.
* @param [in] data <tt>char*</tt>: Opus packet
* @returns Number of channels
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_channels(const unsigned char *data) OPUS_ARG_NONNULL(1);
/** Gets the number of frames in an Opus packet.
* @param [in] packet <tt>char*</tt>: Opus packet
* @param [in] len <tt>opus_int32</tt>: Length of packet
* @returns Number of frames
* @retval OPUS_BAD_ARG Insufficient data was passed to the function
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_frames(const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1);
/** Gets the number of samples of an Opus packet.
* @param [in] packet <tt>char*</tt>: Opus packet
* @param [in] len <tt>opus_int32</tt>: Length of packet
* @param [in] Fs <tt>opus_int32</tt>: Sampling rate in Hz.
* This must be a multiple of 400, or
* inaccurate results will be returned.
* @returns Number of samples
* @retval OPUS_BAD_ARG Insufficient data was passed to the function
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_packet_get_nb_samples(const unsigned char packet[], opus_int32 len, opus_int32 Fs) OPUS_ARG_NONNULL(1);
/** Gets the number of samples of an Opus packet.
* @param [in] dec <tt>OpusDecoder*</tt>: Decoder state
* @param [in] packet <tt>char*</tt>: Opus packet
* @param [in] len <tt>opus_int32</tt>: Length of packet
* @returns Number of samples
* @retval OPUS_BAD_ARG Insufficient data was passed to the function
* @retval OPUS_INVALID_PACKET The compressed data passed is corrupted or of an unsupported type
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_decoder_get_nb_samples(const OpusDecoder *dec, const unsigned char packet[], opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
/** Applies soft-clipping to bring a float signal within the [-1,1] range. If
* the signal is already in that range, nothing is done. If there are values
* outside of [-1,1], then the signal is clipped as smoothly as possible to
* both fit in the range and avoid creating excessive distortion in the
* process.
* @param [in,out] pcm <tt>float*</tt>: Input PCM and modified PCM
* @param [in] frame_size <tt>int</tt> Number of samples per channel to process
* @param [in] channels <tt>int</tt>: Number of channels
* @param [in,out] softclip_mem <tt>float*</tt>: State memory for the soft clipping process (one float per channel, initialized to zero)
*/
OPUS_EXPORT void opus_pcm_soft_clip(float *pcm, int frame_size, int channels, float *softclip_mem);
/**@}*/
/** @defgroup opus_repacketizer Repacketizer
* @{
*
* The repacketizer can be used to merge multiple Opus packets into a single
* packet or alternatively to split Opus packets that have previously been
* merged. Splitting valid Opus packets is always guaranteed to succeed,
* whereas merging valid packets only succeeds if all frames have the same
* mode, bandwidth, and frame size, and when the total duration of the merged
* packet is no more than 120 ms. The 120 ms limit comes from the
* specification and limits decoder memory requirements at a point where
* framing overhead becomes negligible.
*
* The repacketizer currently only operates on elementary Opus
* streams. It will not manipualte multistream packets successfully, except in
* the degenerate case where they consist of data from a single stream.
*
* The repacketizing process starts with creating a repacketizer state, either
* by calling opus_repacketizer_create() or by allocating the memory yourself,
* e.g.,
* @code
* OpusRepacketizer *rp;
* rp = (OpusRepacketizer*)malloc(opus_repacketizer_get_size());
* if (rp != NULL)
* opus_repacketizer_init(rp);
* @endcode
*
* Then the application should submit packets with opus_repacketizer_cat(),
* extract new packets with opus_repacketizer_out() or
* opus_repacketizer_out_range(), and then reset the state for the next set of
* input packets via opus_repacketizer_init().
*
* For example, to split a sequence of packets into individual frames:
* @code
* unsigned char *data;
* int len;
* while (get_next_packet(&data, &len))
* {
* unsigned char out[1276];
* opus_int32 out_len;
* int nb_frames;
* int err;
* int i;
* err = opus_repacketizer_cat(rp, data, len);
* if (err != OPUS_OK)
* {
* release_packet(data);
* return err;
* }
* nb_frames = opus_repacketizer_get_nb_frames(rp);
* for (i = 0; i < nb_frames; i++)
* {
* out_len = opus_repacketizer_out_range(rp, i, i+1, out, sizeof(out));
* if (out_len < 0)
* {
* release_packet(data);
* return (int)out_len;
* }
* output_next_packet(out, out_len);
* }
* opus_repacketizer_init(rp);
* release_packet(data);
* }
* @endcode
*
* Alternatively, to combine a sequence of frames into packets that each
* contain up to <code>TARGET_DURATION_MS</code> milliseconds of data:
* @code
* // The maximum number of packets with duration TARGET_DURATION_MS occurs
* // when the frame size is 2.5 ms, for a total of (TARGET_DURATION_MS*2/5)
* // packets.
* unsigned char *data[(TARGET_DURATION_MS*2/5)+1];
* opus_int32 len[(TARGET_DURATION_MS*2/5)+1];
* int nb_packets;
* unsigned char out[1277*(TARGET_DURATION_MS*2/2)];
* opus_int32 out_len;
* int prev_toc;
* nb_packets = 0;
* while (get_next_packet(data+nb_packets, len+nb_packets))
* {
* int nb_frames;
* int err;
* nb_frames = opus_packet_get_nb_frames(data[nb_packets], len[nb_packets]);
* if (nb_frames < 1)
* {
* release_packets(data, nb_packets+1);
* return nb_frames;
* }
* nb_frames += opus_repacketizer_get_nb_frames(rp);
* // If adding the next packet would exceed our target, or it has an
* // incompatible TOC sequence, output the packets we already have before
* // submitting it.
* // N.B., The nb_packets > 0 check ensures we've submitted at least one
* // packet since the last call to opus_repacketizer_init(). Otherwise a
* // single packet longer than TARGET_DURATION_MS would cause us to try to
* // output an (invalid) empty packet. It also ensures that prev_toc has
* // been set to a valid value. Additionally, len[nb_packets] > 0 is
* // guaranteed by the call to opus_packet_get_nb_frames() above, so the
* // reference to data[nb_packets][0] should be valid.
* if (nb_packets > 0 && (
* ((prev_toc & 0xFC) != (data[nb_packets][0] & 0xFC)) ||
* opus_packet_get_samples_per_frame(data[nb_packets], 48000)*nb_frames >
* TARGET_DURATION_MS*48))
* {
* out_len = opus_repacketizer_out(rp, out, sizeof(out));
* if (out_len < 0)
* {
* release_packets(data, nb_packets+1);
* return (int)out_len;
* }
* output_next_packet(out, out_len);
* opus_repacketizer_init(rp);
* release_packets(data, nb_packets);
* data[0] = data[nb_packets];
* len[0] = len[nb_packets];
* nb_packets = 0;
* }
* err = opus_repacketizer_cat(rp, data[nb_packets], len[nb_packets]);
* if (err != OPUS_OK)
* {
* release_packets(data, nb_packets+1);
* return err;
* }
* prev_toc = data[nb_packets][0];
* nb_packets++;
* }
* // Output the final, partial packet.
* if (nb_packets > 0)
* {
* out_len = opus_repacketizer_out(rp, out, sizeof(out));
* release_packets(data, nb_packets);
* if (out_len < 0)
* return (int)out_len;
* output_next_packet(out, out_len);
* }
* @endcode
*
* An alternate way of merging packets is to simply call opus_repacketizer_cat()
* unconditionally until it fails. At that point, the merged packet can be
* obtained with opus_repacketizer_out() and the input packet for which
* opus_repacketizer_cat() needs to be re-added to a newly reinitialized
* repacketizer state.
*/
typedef struct OpusRepacketizer OpusRepacketizer;
/** Gets the size of an <code>OpusRepacketizer</code> structure.
* @returns The size in bytes.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_size(void);
/** (Re)initializes a previously allocated repacketizer state.
* The state must be at least the size returned by opus_repacketizer_get_size().
* This can be used for applications which use their own allocator instead of
* malloc().
* It must also be called to reset the queue of packets waiting to be
* repacketized, which is necessary if the maximum packet duration of 120 ms
* is reached or if you wish to submit packets with a different Opus
* configuration (coding mode, audio bandwidth, frame size, or channel count).
* Failure to do so will prevent a new packet from being added with
* opus_repacketizer_cat().
* @see opus_repacketizer_create
* @see opus_repacketizer_get_size
* @see opus_repacketizer_cat
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to
* (re)initialize.
* @returns A pointer to the same repacketizer state that was passed in.
*/
OPUS_EXPORT OpusRepacketizer *opus_repacketizer_init(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
/** Allocates memory and initializes the new repacketizer with
* opus_repacketizer_init().
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusRepacketizer *opus_repacketizer_create(void);
/** Frees an <code>OpusRepacketizer</code> allocated by
* opus_repacketizer_create().
* @param[in] rp <tt>OpusRepacketizer*</tt>: State to be freed.
*/
OPUS_EXPORT void opus_repacketizer_destroy(OpusRepacketizer *rp);
/** Add a packet to the current repacketizer state.
* This packet must match the configuration of any packets already submitted
* for repacketization since the last call to opus_repacketizer_init().
* This means that it must have the same coding mode, audio bandwidth, frame
* size, and channel count.
* This can be checked in advance by examining the top 6 bits of the first
* byte of the packet, and ensuring they match the top 6 bits of the first
* byte of any previously submitted packet.
* The total duration of audio in the repacketizer state also must not exceed
* 120 ms, the maximum duration of a single packet, after adding this packet.
*
* The contents of the current repacketizer state can be extracted into new
* packets using opus_repacketizer_out() or opus_repacketizer_out_range().
*
* In order to add a packet with a different configuration or to add more
* audio beyond 120 ms, you must clear the repacketizer state by calling
* opus_repacketizer_init().
* If a packet is too large to add to the current repacketizer state, no part
* of it is added, even if it contains multiple frames, some of which might
* fit.
* If you wish to be able to add parts of such packets, you should first use
* another repacketizer to split the packet into pieces and add them
* individually.
* @see opus_repacketizer_out_range
* @see opus_repacketizer_out
* @see opus_repacketizer_init
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state to which to
* add the packet.
* @param[in] data <tt>const unsigned char*</tt>: The packet data.
* The application must ensure
* this pointer remains valid
* until the next call to
* opus_repacketizer_init() or
* opus_repacketizer_destroy().
* @param len <tt>opus_int32</tt>: The number of bytes in the packet data.
* @returns An error code indicating whether or not the operation succeeded.
* @retval #OPUS_OK The packet's contents have been added to the repacketizer
* state.
* @retval #OPUS_INVALID_PACKET The packet did not have a valid TOC sequence,
* the packet's TOC sequence was not compatible
* with previously submitted packets (because
* the coding mode, audio bandwidth, frame size,
* or channel count did not match), or adding
* this packet would increase the total amount of
* audio stored in the repacketizer state to more
* than 120 ms.
*/
OPUS_EXPORT int opus_repacketizer_cat(OpusRepacketizer *rp, const unsigned char *data, opus_int32 len) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2);
/** Construct a new packet from data previously submitted to the repacketizer
* state via opus_repacketizer_cat().
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
* construct the new packet.
* @param begin <tt>int</tt>: The index of the first frame in the current
* repacketizer state to include in the output.
* @param end <tt>int</tt>: One past the index of the last frame in the
* current repacketizer state to include in the
* output.
* @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
* store the output packet.
* @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
* the output buffer. In order to guarantee
* success, this should be at least
* <code>1276</code> for a single frame,
* or for multiple frames,
* <code>1277*(end-begin)</code>.
* However, <code>1*(end-begin)</code> plus
* the size of all packet data submitted to
* the repacketizer since the last call to
* opus_repacketizer_init() or
* opus_repacketizer_create() is also
* sufficient, and possibly much smaller.
* @returns The total size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BAD_ARG <code>[begin,end)</code> was an invalid range of
* frames (begin < 0, begin >= end, or end >
* opus_repacketizer_get_nb_frames()).
* @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
* complete output packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out_range(OpusRepacketizer *rp, int begin, int end, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Return the total number of frames contained in packet data submitted to
* the repacketizer state so far via opus_repacketizer_cat() since the last
* call to opus_repacketizer_init() or opus_repacketizer_create().
* This defines the valid range of packets that can be extracted with
* opus_repacketizer_out_range() or opus_repacketizer_out().
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state containing the
* frames.
* @returns The total number of frames contained in the packet data submitted
* to the repacketizer state.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_repacketizer_get_nb_frames(OpusRepacketizer *rp) OPUS_ARG_NONNULL(1);
/** Construct a new packet from data previously submitted to the repacketizer
* state via opus_repacketizer_cat().
* This is a convenience routine that returns all the data submitted so far
* in a single packet.
* It is equivalent to calling
* @code
* opus_repacketizer_out_range(rp, 0, opus_repacketizer_get_nb_frames(rp),
* data, maxlen)
* @endcode
* @param rp <tt>OpusRepacketizer*</tt>: The repacketizer state from which to
* construct the new packet.
* @param[out] data <tt>const unsigned char*</tt>: The buffer in which to
* store the output packet.
* @param maxlen <tt>opus_int32</tt>: The maximum number of bytes to store in
* the output buffer. In order to guarantee
* success, this should be at least
* <code>1277*opus_repacketizer_get_nb_frames(rp)</code>.
* However,
* <code>1*opus_repacketizer_get_nb_frames(rp)</code>
* plus the size of all packet data
* submitted to the repacketizer since the
* last call to opus_repacketizer_init() or
* opus_repacketizer_create() is also
* sufficient, and possibly much smaller.
* @returns The total size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BUFFER_TOO_SMALL \a maxlen was insufficient to contain the
* complete output packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_repacketizer_out(OpusRepacketizer *rp, unsigned char *data, opus_int32 maxlen) OPUS_ARG_NONNULL(1);
/** Pads a given Opus packet to a larger size (possibly changing the TOC sequence).
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to pad.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
* This must be at least as large as len.
* @returns an error code
* @retval #OPUS_OK \a on success.
* @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT int opus_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len);
/** Remove all padding from a given Opus packet and rewrite the TOC sequence to
* minimize space usage.
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to strip.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @returns The new size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BAD_ARG \a len was less than 1.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_packet_unpad(unsigned char *data, opus_int32 len);
/** Pads a given Opus multi-stream packet to a larger size (possibly changing the TOC sequence).
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to pad.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @param new_len <tt>opus_int32</tt>: The desired size of the packet after padding.
* This must be at least 1.
* @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
* This must be at least as large as len.
* @returns an error code
* @retval #OPUS_OK \a on success.
* @retval #OPUS_BAD_ARG \a len was less than 1.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT int opus_multistream_packet_pad(unsigned char *data, opus_int32 len, opus_int32 new_len, int nb_streams);
/** Remove all padding from a given Opus multi-stream packet and rewrite the TOC sequence to
* minimize space usage.
* @param[in,out] data <tt>const unsigned char*</tt>: The buffer containing the
* packet to strip.
* @param len <tt>opus_int32</tt>: The size of the packet.
* This must be at least 1.
* @param nb_streams <tt>opus_int32</tt>: The number of streams (not channels) in the packet.
* This must be at least 1.
* @returns The new size of the output packet on success, or an error code
* on failure.
* @retval #OPUS_BAD_ARG \a len was less than 1 or new_len was less than len.
* @retval #OPUS_INVALID_PACKET \a data did not contain a valid Opus packet.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_packet_unpad(unsigned char *data, opus_int32 len, int nb_streams);
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_H */
libs/SDL3_mixer/VisualC/external/include/opus/opus_defines.h
Vendored
+801
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@@ -0,0 +1,801 @@
/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
Written by Jean-Marc Valin and Koen Vos */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/**
* @file opus_defines.h
* @brief Opus reference implementation constants
*/
#ifndef OPUS_DEFINES_H
#define OPUS_DEFINES_H
#include <opus/opus_types.h>
#ifdef __cplusplus
extern "C" {
#endif
/** @defgroup opus_errorcodes Error codes
* @{
*/
/** No error @hideinitializer*/
#define OPUS_OK 0
/** One or more invalid/out of range arguments @hideinitializer*/
#define OPUS_BAD_ARG -1
/** Not enough bytes allocated in the buffer @hideinitializer*/
#define OPUS_BUFFER_TOO_SMALL -2
/** An internal error was detected @hideinitializer*/
#define OPUS_INTERNAL_ERROR -3
/** The compressed data passed is corrupted @hideinitializer*/
#define OPUS_INVALID_PACKET -4
/** Invalid/unsupported request number @hideinitializer*/
#define OPUS_UNIMPLEMENTED -5
/** An encoder or decoder structure is invalid or already freed @hideinitializer*/
#define OPUS_INVALID_STATE -6
/** Memory allocation has failed @hideinitializer*/
#define OPUS_ALLOC_FAIL -7
/**@}*/
/** @cond OPUS_INTERNAL_DOC */
/**Export control for opus functions */
#ifndef OPUS_EXPORT
# if defined(_WIN32)
# if defined(OPUS_BUILD) && defined(DLL_EXPORT)
# define OPUS_EXPORT __declspec(dllexport)
# else
# define OPUS_EXPORT
# endif
# elif defined(__GNUC__) && defined(OPUS_BUILD)
# define OPUS_EXPORT __attribute__ ((visibility ("default")))
# else
# define OPUS_EXPORT
# endif
#endif
# if !defined(OPUS_GNUC_PREREQ)
# if defined(__GNUC__)&&defined(__GNUC_MINOR__)
# define OPUS_GNUC_PREREQ(_maj,_min) \
((__GNUC__<<16)+__GNUC_MINOR__>=((_maj)<<16)+(_min))
# else
# define OPUS_GNUC_PREREQ(_maj,_min) 0
# endif
# endif
#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
# if OPUS_GNUC_PREREQ(3,0)
# define OPUS_RESTRICT __restrict__
# elif (defined(_MSC_VER) && _MSC_VER >= 1400)
# define OPUS_RESTRICT __restrict
# else
# define OPUS_RESTRICT
# endif
#else
# define OPUS_RESTRICT restrict
#endif
#if (!defined(__STDC_VERSION__) || (__STDC_VERSION__ < 199901L) )
# if OPUS_GNUC_PREREQ(2,7)
# define OPUS_INLINE __inline__
# elif (defined(_MSC_VER))
# define OPUS_INLINE __inline
# else
# define OPUS_INLINE
# endif
#else
# define OPUS_INLINE inline
#endif
/**Warning attributes for opus functions
* NONNULL is not used in OPUS_BUILD to avoid the compiler optimizing out
* some paranoid null checks. */
#if defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
# define OPUS_WARN_UNUSED_RESULT __attribute__ ((__warn_unused_result__))
#else
# define OPUS_WARN_UNUSED_RESULT
#endif
#if !defined(OPUS_BUILD) && defined(__GNUC__) && OPUS_GNUC_PREREQ(3, 4)
# define OPUS_ARG_NONNULL(_x) __attribute__ ((__nonnull__(_x)))
#else
# define OPUS_ARG_NONNULL(_x)
#endif
/** These are the actual Encoder CTL ID numbers.
* They should not be used directly by applications.
* In general, SETs should be even and GETs should be odd.*/
#define OPUS_SET_APPLICATION_REQUEST 4000
#define OPUS_GET_APPLICATION_REQUEST 4001
#define OPUS_SET_BITRATE_REQUEST 4002
#define OPUS_GET_BITRATE_REQUEST 4003
#define OPUS_SET_MAX_BANDWIDTH_REQUEST 4004
#define OPUS_GET_MAX_BANDWIDTH_REQUEST 4005
#define OPUS_SET_VBR_REQUEST 4006
#define OPUS_GET_VBR_REQUEST 4007
#define OPUS_SET_BANDWIDTH_REQUEST 4008
#define OPUS_GET_BANDWIDTH_REQUEST 4009
#define OPUS_SET_COMPLEXITY_REQUEST 4010
#define OPUS_GET_COMPLEXITY_REQUEST 4011
#define OPUS_SET_INBAND_FEC_REQUEST 4012
#define OPUS_GET_INBAND_FEC_REQUEST 4013
#define OPUS_SET_PACKET_LOSS_PERC_REQUEST 4014
#define OPUS_GET_PACKET_LOSS_PERC_REQUEST 4015
#define OPUS_SET_DTX_REQUEST 4016
#define OPUS_GET_DTX_REQUEST 4017
#define OPUS_SET_VBR_CONSTRAINT_REQUEST 4020
#define OPUS_GET_VBR_CONSTRAINT_REQUEST 4021
#define OPUS_SET_FORCE_CHANNELS_REQUEST 4022
#define OPUS_GET_FORCE_CHANNELS_REQUEST 4023
#define OPUS_SET_SIGNAL_REQUEST 4024
#define OPUS_GET_SIGNAL_REQUEST 4025
#define OPUS_GET_LOOKAHEAD_REQUEST 4027
/* #define OPUS_RESET_STATE 4028 */
#define OPUS_GET_SAMPLE_RATE_REQUEST 4029
#define OPUS_GET_FINAL_RANGE_REQUEST 4031
#define OPUS_GET_PITCH_REQUEST 4033
#define OPUS_SET_GAIN_REQUEST 4034
#define OPUS_GET_GAIN_REQUEST 4045 /* Should have been 4035 */
#define OPUS_SET_LSB_DEPTH_REQUEST 4036
#define OPUS_GET_LSB_DEPTH_REQUEST 4037
#define OPUS_GET_LAST_PACKET_DURATION_REQUEST 4039
#define OPUS_SET_EXPERT_FRAME_DURATION_REQUEST 4040
#define OPUS_GET_EXPERT_FRAME_DURATION_REQUEST 4041
#define OPUS_SET_PREDICTION_DISABLED_REQUEST 4042
#define OPUS_GET_PREDICTION_DISABLED_REQUEST 4043
/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
#define OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST 4046
#define OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST 4047
#define OPUS_GET_IN_DTX_REQUEST 4049
/** Defines for the presence of extended APIs. */
#define OPUS_HAVE_OPUS_PROJECTION_H
/* Macros to trigger compilation errors when the wrong types are provided to a CTL */
#define __opus_check_int(x) (((void)((x) == (opus_int32)0)), (opus_int32)(x))
#define __opus_check_int_ptr(ptr) ((ptr) + ((ptr) - (opus_int32*)(ptr)))
#define __opus_check_uint_ptr(ptr) ((ptr) + ((ptr) - (opus_uint32*)(ptr)))
#define __opus_check_val16_ptr(ptr) ((ptr) + ((ptr) - (opus_val16*)(ptr)))
/** @endcond */
/** @defgroup opus_ctlvalues Pre-defined values for CTL interface
* @see opus_genericctls, opus_encoderctls
* @{
*/
/* Values for the various encoder CTLs */
#define OPUS_AUTO -1000 /**<Auto/default setting @hideinitializer*/
#define OPUS_BITRATE_MAX -1 /**<Maximum bitrate @hideinitializer*/
/** Best for most VoIP/videoconference applications where listening quality and intelligibility matter most
* @hideinitializer */
#define OPUS_APPLICATION_VOIP 2048
/** Best for broadcast/high-fidelity application where the decoded audio should be as close as possible to the input
* @hideinitializer */
#define OPUS_APPLICATION_AUDIO 2049
/** Only use when lowest-achievable latency is what matters most. Voice-optimized modes cannot be used.
* @hideinitializer */
#define OPUS_APPLICATION_RESTRICTED_LOWDELAY 2051
#define OPUS_SIGNAL_VOICE 3001 /**< Signal being encoded is voice */
#define OPUS_SIGNAL_MUSIC 3002 /**< Signal being encoded is music */
#define OPUS_BANDWIDTH_NARROWBAND 1101 /**< 4 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_MEDIUMBAND 1102 /**< 6 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_WIDEBAND 1103 /**< 8 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_SUPERWIDEBAND 1104 /**<12 kHz bandpass @hideinitializer*/
#define OPUS_BANDWIDTH_FULLBAND 1105 /**<20 kHz bandpass @hideinitializer*/
#define OPUS_FRAMESIZE_ARG 5000 /**< Select frame size from the argument (default) */
#define OPUS_FRAMESIZE_2_5_MS 5001 /**< Use 2.5 ms frames */
#define OPUS_FRAMESIZE_5_MS 5002 /**< Use 5 ms frames */
#define OPUS_FRAMESIZE_10_MS 5003 /**< Use 10 ms frames */
#define OPUS_FRAMESIZE_20_MS 5004 /**< Use 20 ms frames */
#define OPUS_FRAMESIZE_40_MS 5005 /**< Use 40 ms frames */
#define OPUS_FRAMESIZE_60_MS 5006 /**< Use 60 ms frames */
#define OPUS_FRAMESIZE_80_MS 5007 /**< Use 80 ms frames */
#define OPUS_FRAMESIZE_100_MS 5008 /**< Use 100 ms frames */
#define OPUS_FRAMESIZE_120_MS 5009 /**< Use 120 ms frames */
/**@}*/
/** @defgroup opus_encoderctls Encoder related CTLs
*
* These are convenience macros for use with the \c opus_encode_ctl
* interface. They are used to generate the appropriate series of
* arguments for that call, passing the correct type, size and so
* on as expected for each particular request.
*
* Some usage examples:
*
* @code
* int ret;
* ret = opus_encoder_ctl(enc_ctx, OPUS_SET_BANDWIDTH(OPUS_AUTO));
* if (ret != OPUS_OK) return ret;
*
* opus_int32 rate;
* opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&rate));
*
* opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
* @endcode
*
* @see opus_genericctls, opus_encoder
* @{
*/
/** Configures the encoder's computational complexity.
* The supported range is 0-10 inclusive with 10 representing the highest complexity.
* @see OPUS_GET_COMPLEXITY
* @param[in] x <tt>opus_int32</tt>: Allowed values: 0-10, inclusive.
*
* @hideinitializer */
#define OPUS_SET_COMPLEXITY(x) OPUS_SET_COMPLEXITY_REQUEST, __opus_check_int(x)
/** Gets the encoder's complexity configuration.
* @see OPUS_SET_COMPLEXITY
* @param[out] x <tt>opus_int32 *</tt>: Returns a value in the range 0-10,
* inclusive.
* @hideinitializer */
#define OPUS_GET_COMPLEXITY(x) OPUS_GET_COMPLEXITY_REQUEST, __opus_check_int_ptr(x)
/** Configures the bitrate in the encoder.
* Rates from 500 to 512000 bits per second are meaningful, as well as the
* special values #OPUS_AUTO and #OPUS_BITRATE_MAX.
* The value #OPUS_BITRATE_MAX can be used to cause the codec to use as much
* rate as it can, which is useful for controlling the rate by adjusting the
* output buffer size.
* @see OPUS_GET_BITRATE
* @param[in] x <tt>opus_int32</tt>: Bitrate in bits per second. The default
* is determined based on the number of
* channels and the input sampling rate.
* @hideinitializer */
#define OPUS_SET_BITRATE(x) OPUS_SET_BITRATE_REQUEST, __opus_check_int(x)
/** Gets the encoder's bitrate configuration.
* @see OPUS_SET_BITRATE
* @param[out] x <tt>opus_int32 *</tt>: Returns the bitrate in bits per second.
* The default is determined based on the
* number of channels and the input
* sampling rate.
* @hideinitializer */
#define OPUS_GET_BITRATE(x) OPUS_GET_BITRATE_REQUEST, __opus_check_int_ptr(x)
/** Enables or disables variable bitrate (VBR) in the encoder.
* The configured bitrate may not be met exactly because frames must
* be an integer number of bytes in length.
* @see OPUS_GET_VBR
* @see OPUS_SET_VBR_CONSTRAINT
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Hard CBR. For LPC/hybrid modes at very low bit-rate, this can
* cause noticeable quality degradation.</dd>
* <dt>1</dt><dd>VBR (default). The exact type of VBR is controlled by
* #OPUS_SET_VBR_CONSTRAINT.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_VBR(x) OPUS_SET_VBR_REQUEST, __opus_check_int(x)
/** Determine if variable bitrate (VBR) is enabled in the encoder.
* @see OPUS_SET_VBR
* @see OPUS_GET_VBR_CONSTRAINT
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Hard CBR.</dd>
* <dt>1</dt><dd>VBR (default). The exact type of VBR may be retrieved via
* #OPUS_GET_VBR_CONSTRAINT.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_VBR(x) OPUS_GET_VBR_REQUEST, __opus_check_int_ptr(x)
/** Enables or disables constrained VBR in the encoder.
* This setting is ignored when the encoder is in CBR mode.
* @warning Only the MDCT mode of Opus currently heeds the constraint.
* Speech mode ignores it completely, hybrid mode may fail to obey it
* if the LPC layer uses more bitrate than the constraint would have
* permitted.
* @see OPUS_GET_VBR_CONSTRAINT
* @see OPUS_SET_VBR
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Unconstrained VBR.</dd>
* <dt>1</dt><dd>Constrained VBR (default). This creates a maximum of one
* frame of buffering delay assuming a transport with a
* serialization speed of the nominal bitrate.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_VBR_CONSTRAINT(x) OPUS_SET_VBR_CONSTRAINT_REQUEST, __opus_check_int(x)
/** Determine if constrained VBR is enabled in the encoder.
* @see OPUS_SET_VBR_CONSTRAINT
* @see OPUS_GET_VBR
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Unconstrained VBR.</dd>
* <dt>1</dt><dd>Constrained VBR (default).</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_VBR_CONSTRAINT(x) OPUS_GET_VBR_CONSTRAINT_REQUEST, __opus_check_int_ptr(x)
/** Configures mono/stereo forcing in the encoder.
* This can force the encoder to produce packets encoded as either mono or
* stereo, regardless of the format of the input audio. This is useful when
* the caller knows that the input signal is currently a mono source embedded
* in a stereo stream.
* @see OPUS_GET_FORCE_CHANNELS
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
* <dt>1</dt> <dd>Forced mono</dd>
* <dt>2</dt> <dd>Forced stereo</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_FORCE_CHANNELS(x) OPUS_SET_FORCE_CHANNELS_REQUEST, __opus_check_int(x)
/** Gets the encoder's forced channel configuration.
* @see OPUS_SET_FORCE_CHANNELS
* @param[out] x <tt>opus_int32 *</tt>:
* <dl>
* <dt>#OPUS_AUTO</dt><dd>Not forced (default)</dd>
* <dt>1</dt> <dd>Forced mono</dd>
* <dt>2</dt> <dd>Forced stereo</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_FORCE_CHANNELS(x) OPUS_GET_FORCE_CHANNELS_REQUEST, __opus_check_int_ptr(x)
/** Configures the maximum bandpass that the encoder will select automatically.
* Applications should normally use this instead of #OPUS_SET_BANDWIDTH
* (leaving that set to the default, #OPUS_AUTO). This allows the
* application to set an upper bound based on the type of input it is
* providing, but still gives the encoder the freedom to reduce the bandpass
* when the bitrate becomes too low, for better overall quality.
* @see OPUS_GET_MAX_BANDWIDTH
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband (default)</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_MAX_BANDWIDTH(x) OPUS_SET_MAX_BANDWIDTH_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured maximum allowed bandpass.
* @see OPUS_SET_MAX_BANDWIDTH
* @param[out] x <tt>opus_int32 *</tt>: Allowed values:
* <dl>
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband (default)</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_MAX_BANDWIDTH(x) OPUS_GET_MAX_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
/** Sets the encoder's bandpass to a specific value.
* This prevents the encoder from automatically selecting the bandpass based
* on the available bitrate. If an application knows the bandpass of the input
* audio it is providing, it should normally use #OPUS_SET_MAX_BANDWIDTH
* instead, which still gives the encoder the freedom to reduce the bandpass
* when the bitrate becomes too low, for better overall quality.
* @see OPUS_GET_BANDWIDTH
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_BANDWIDTH(x) OPUS_SET_BANDWIDTH_REQUEST, __opus_check_int(x)
/** Configures the type of signal being encoded.
* This is a hint which helps the encoder's mode selection.
* @see OPUS_GET_SIGNAL
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
* <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_SIGNAL(x) OPUS_SET_SIGNAL_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured signal type.
* @see OPUS_SET_SIGNAL
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_SIGNAL_VOICE</dt><dd>Bias thresholds towards choosing LPC or Hybrid modes.</dd>
* <dt>#OPUS_SIGNAL_MUSIC</dt><dd>Bias thresholds towards choosing MDCT modes.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_SIGNAL(x) OPUS_GET_SIGNAL_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's intended application.
* The initial value is a mandatory argument to the encoder_create function.
* @see OPUS_GET_APPLICATION
* @param[in] x <tt>opus_int32</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_APPLICATION(x) OPUS_SET_APPLICATION_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured application.
* @see OPUS_SET_APPLICATION
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_APPLICATION(x) OPUS_GET_APPLICATION_REQUEST, __opus_check_int_ptr(x)
/** Gets the total samples of delay added by the entire codec.
* This can be queried by the encoder and then the provided number of samples can be
* skipped on from the start of the decoder's output to provide time aligned input
* and output. From the perspective of a decoding application the real data begins this many
* samples late.
*
* The decoder contribution to this delay is identical for all decoders, but the
* encoder portion of the delay may vary from implementation to implementation,
* version to version, or even depend on the encoder's initial configuration.
* Applications needing delay compensation should call this CTL rather than
* hard-coding a value.
* @param[out] x <tt>opus_int32 *</tt>: Number of lookahead samples
* @hideinitializer */
#define OPUS_GET_LOOKAHEAD(x) OPUS_GET_LOOKAHEAD_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's use of inband forward error correction (FEC).
* @note This is only applicable to the LPC layer
* @see OPUS_GET_INBAND_FEC
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Disable inband FEC (default).</dd>
* <dt>1</dt><dd>Inband FEC enabled. If the packet loss rate is sufficiently high, Opus will automatically switch to SILK even at high rates to enable use of that FEC.</dd>
* <dt>2</dt><dd>Inband FEC enabled, but does not necessarily switch to SILK if we have music.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_INBAND_FEC(x) OPUS_SET_INBAND_FEC_REQUEST, __opus_check_int(x)
/** Gets encoder's configured use of inband forward error correction.
* @see OPUS_SET_INBAND_FEC
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Inband FEC disabled (default).</dd>
* <dt>1</dt><dd>Inband FEC enabled. If the packet loss rate is sufficiently high, Opus will automatically switch to SILK even at high rates to enable use of that FEC.</dd>
* <dt>2</dt><dd>Inband FEC enabled, but does not necessarily switch to SILK if we have music.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_INBAND_FEC(x) OPUS_GET_INBAND_FEC_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's expected packet loss percentage.
* Higher values trigger progressively more loss resistant behavior in the encoder
* at the expense of quality at a given bitrate in the absence of packet loss, but
* greater quality under loss.
* @see OPUS_GET_PACKET_LOSS_PERC
* @param[in] x <tt>opus_int32</tt>: Loss percentage in the range 0-100, inclusive (default: 0).
* @hideinitializer */
#define OPUS_SET_PACKET_LOSS_PERC(x) OPUS_SET_PACKET_LOSS_PERC_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured packet loss percentage.
* @see OPUS_SET_PACKET_LOSS_PERC
* @param[out] x <tt>opus_int32 *</tt>: Returns the configured loss percentage
* in the range 0-100, inclusive (default: 0).
* @hideinitializer */
#define OPUS_GET_PACKET_LOSS_PERC(x) OPUS_GET_PACKET_LOSS_PERC_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's use of discontinuous transmission (DTX).
* @note This is only applicable to the LPC layer
* @see OPUS_GET_DTX
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Disable DTX (default).</dd>
* <dt>1</dt><dd>Enabled DTX.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_DTX(x) OPUS_SET_DTX_REQUEST, __opus_check_int(x)
/** Gets encoder's configured use of discontinuous transmission.
* @see OPUS_SET_DTX
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>DTX disabled (default).</dd>
* <dt>1</dt><dd>DTX enabled.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_DTX(x) OPUS_GET_DTX_REQUEST, __opus_check_int_ptr(x)
/** Configures the depth of signal being encoded.
*
* This is a hint which helps the encoder identify silence and near-silence.
* It represents the number of significant bits of linear intensity below
* which the signal contains ignorable quantization or other noise.
*
* For example, OPUS_SET_LSB_DEPTH(14) would be an appropriate setting
* for G.711 u-law input. OPUS_SET_LSB_DEPTH(16) would be appropriate
* for 16-bit linear pcm input with opus_encode_float().
*
* When using opus_encode() instead of opus_encode_float(), or when libopus
* is compiled for fixed-point, the encoder uses the minimum of the value
* set here and the value 16.
*
* @see OPUS_GET_LSB_DEPTH
* @param[in] x <tt>opus_int32</tt>: Input precision in bits, between 8 and 24
* (default: 24).
* @hideinitializer */
#define OPUS_SET_LSB_DEPTH(x) OPUS_SET_LSB_DEPTH_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured signal depth.
* @see OPUS_SET_LSB_DEPTH
* @param[out] x <tt>opus_int32 *</tt>: Input precision in bits, between 8 and
* 24 (default: 24).
* @hideinitializer */
#define OPUS_GET_LSB_DEPTH(x) OPUS_GET_LSB_DEPTH_REQUEST, __opus_check_int_ptr(x)
/** Configures the encoder's use of variable duration frames.
* When variable duration is enabled, the encoder is free to use a shorter frame
* size than the one requested in the opus_encode*() call.
* It is then the user's responsibility
* to verify how much audio was encoded by checking the ToC byte of the encoded
* packet. The part of the audio that was not encoded needs to be resent to the
* encoder for the next call. Do not use this option unless you <b>really</b>
* know what you are doing.
* @see OPUS_GET_EXPERT_FRAME_DURATION
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
* <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_EXPERT_FRAME_DURATION(x) OPUS_SET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured use of variable duration frames.
* @see OPUS_SET_EXPERT_FRAME_DURATION
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>OPUS_FRAMESIZE_ARG</dt><dd>Select frame size from the argument (default).</dd>
* <dt>OPUS_FRAMESIZE_2_5_MS</dt><dd>Use 2.5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_5_MS</dt><dd>Use 5 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_10_MS</dt><dd>Use 10 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_20_MS</dt><dd>Use 20 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_40_MS</dt><dd>Use 40 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_60_MS</dt><dd>Use 60 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_80_MS</dt><dd>Use 80 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_100_MS</dt><dd>Use 100 ms frames.</dd>
* <dt>OPUS_FRAMESIZE_120_MS</dt><dd>Use 120 ms frames.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_EXPERT_FRAME_DURATION(x) OPUS_GET_EXPERT_FRAME_DURATION_REQUEST, __opus_check_int_ptr(x)
/** If set to 1, disables almost all use of prediction, making frames almost
* completely independent. This reduces quality.
* @see OPUS_GET_PREDICTION_DISABLED
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Enable prediction (default).</dd>
* <dt>1</dt><dd>Disable prediction.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_PREDICTION_DISABLED(x) OPUS_SET_PREDICTION_DISABLED_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured prediction status.
* @see OPUS_SET_PREDICTION_DISABLED
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Prediction enabled (default).</dd>
* <dt>1</dt><dd>Prediction disabled.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_PREDICTION_DISABLED(x) OPUS_GET_PREDICTION_DISABLED_REQUEST, __opus_check_int_ptr(x)
/**@}*/
/** @defgroup opus_genericctls Generic CTLs
*
* These macros are used with the \c opus_decoder_ctl and
* \c opus_encoder_ctl calls to generate a particular
* request.
*
* When called on an \c OpusDecoder they apply to that
* particular decoder instance. When called on an
* \c OpusEncoder they apply to the corresponding setting
* on that encoder instance, if present.
*
* Some usage examples:
*
* @code
* int ret;
* opus_int32 pitch;
* ret = opus_decoder_ctl(dec_ctx, OPUS_GET_PITCH(&pitch));
* if (ret == OPUS_OK) return ret;
*
* opus_encoder_ctl(enc_ctx, OPUS_RESET_STATE);
* opus_decoder_ctl(dec_ctx, OPUS_RESET_STATE);
*
* opus_int32 enc_bw, dec_bw;
* opus_encoder_ctl(enc_ctx, OPUS_GET_BANDWIDTH(&enc_bw));
* opus_decoder_ctl(dec_ctx, OPUS_GET_BANDWIDTH(&dec_bw));
* if (enc_bw != dec_bw) {
* printf("packet bandwidth mismatch!\n");
* }
* @endcode
*
* @see opus_encoder, opus_decoder_ctl, opus_encoder_ctl, opus_decoderctls, opus_encoderctls
* @{
*/
/** Resets the codec state to be equivalent to a freshly initialized state.
* This should be called when switching streams in order to prevent
* the back to back decoding from giving different results from
* one at a time decoding.
* @hideinitializer */
#define OPUS_RESET_STATE 4028
/** Gets the final state of the codec's entropy coder.
* This is used for testing purposes,
* The encoder and decoder state should be identical after coding a payload
* (assuming no data corruption or software bugs)
*
* @param[out] x <tt>opus_uint32 *</tt>: Entropy coder state
*
* @hideinitializer */
#define OPUS_GET_FINAL_RANGE(x) OPUS_GET_FINAL_RANGE_REQUEST, __opus_check_uint_ptr(x)
/** Gets the encoder's configured bandpass or the decoder's last bandpass.
* @see OPUS_SET_BANDWIDTH
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>#OPUS_AUTO</dt> <dd>(default)</dd>
* <dt>#OPUS_BANDWIDTH_NARROWBAND</dt> <dd>4 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_MEDIUMBAND</dt> <dd>6 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_WIDEBAND</dt> <dd>8 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_SUPERWIDEBAND</dt><dd>12 kHz passband</dd>
* <dt>#OPUS_BANDWIDTH_FULLBAND</dt> <dd>20 kHz passband</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_BANDWIDTH(x) OPUS_GET_BANDWIDTH_REQUEST, __opus_check_int_ptr(x)
/** Gets the sampling rate the encoder or decoder was initialized with.
* This simply returns the <code>Fs</code> value passed to opus_encoder_init()
* or opus_decoder_init().
* @param[out] x <tt>opus_int32 *</tt>: Sampling rate of encoder or decoder.
* @hideinitializer
*/
#define OPUS_GET_SAMPLE_RATE(x) OPUS_GET_SAMPLE_RATE_REQUEST, __opus_check_int_ptr(x)
/** If set to 1, disables the use of phase inversion for intensity stereo,
* improving the quality of mono downmixes, but slightly reducing normal
* stereo quality. Disabling phase inversion in the decoder does not comply
* with RFC 6716, although it does not cause any interoperability issue and
* is expected to become part of the Opus standard once RFC 6716 is updated
* by draft-ietf-codec-opus-update.
* @see OPUS_GET_PHASE_INVERSION_DISABLED
* @param[in] x <tt>opus_int32</tt>: Allowed values:
* <dl>
* <dt>0</dt><dd>Enable phase inversion (default).</dd>
* <dt>1</dt><dd>Disable phase inversion.</dd>
* </dl>
* @hideinitializer */
#define OPUS_SET_PHASE_INVERSION_DISABLED(x) OPUS_SET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int(x)
/** Gets the encoder's configured phase inversion status.
* @see OPUS_SET_PHASE_INVERSION_DISABLED
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>Stereo phase inversion enabled (default).</dd>
* <dt>1</dt><dd>Stereo phase inversion disabled.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_PHASE_INVERSION_DISABLED(x) OPUS_GET_PHASE_INVERSION_DISABLED_REQUEST, __opus_check_int_ptr(x)
/** Gets the DTX state of the encoder.
* Returns whether the last encoded frame was either a comfort noise update
* during DTX or not encoded because of DTX.
* @param[out] x <tt>opus_int32 *</tt>: Returns one of the following values:
* <dl>
* <dt>0</dt><dd>The encoder is not in DTX.</dd>
* <dt>1</dt><dd>The encoder is in DTX.</dd>
* </dl>
* @hideinitializer */
#define OPUS_GET_IN_DTX(x) OPUS_GET_IN_DTX_REQUEST, __opus_check_int_ptr(x)
/**@}*/
/** @defgroup opus_decoderctls Decoder related CTLs
* @see opus_genericctls, opus_encoderctls, opus_decoder
* @{
*/
/** Configures decoder gain adjustment.
* Scales the decoded output by a factor specified in Q8 dB units.
* This has a maximum range of -32768 to 32767 inclusive, and returns
* OPUS_BAD_ARG otherwise. The default is zero indicating no adjustment.
* This setting survives decoder reset.
*
* gain = pow(10, x/(20.0*256))
*
* @param[in] x <tt>opus_int32</tt>: Amount to scale PCM signal by in Q8 dB units.
* @hideinitializer */
#define OPUS_SET_GAIN(x) OPUS_SET_GAIN_REQUEST, __opus_check_int(x)
/** Gets the decoder's configured gain adjustment. @see OPUS_SET_GAIN
*
* @param[out] x <tt>opus_int32 *</tt>: Amount to scale PCM signal by in Q8 dB units.
* @hideinitializer */
#define OPUS_GET_GAIN(x) OPUS_GET_GAIN_REQUEST, __opus_check_int_ptr(x)
/** Gets the duration (in samples) of the last packet successfully decoded or concealed.
* @param[out] x <tt>opus_int32 *</tt>: Number of samples (at current sampling rate).
* @hideinitializer */
#define OPUS_GET_LAST_PACKET_DURATION(x) OPUS_GET_LAST_PACKET_DURATION_REQUEST, __opus_check_int_ptr(x)
/** Gets the pitch of the last decoded frame, if available.
* This can be used for any post-processing algorithm requiring the use of pitch,
* e.g. time stretching/shortening. If the last frame was not voiced, or if the
* pitch was not coded in the frame, then zero is returned.
*
* This CTL is only implemented for decoder instances.
*
* @param[out] x <tt>opus_int32 *</tt>: pitch period at 48 kHz (or 0 if not available)
*
* @hideinitializer */
#define OPUS_GET_PITCH(x) OPUS_GET_PITCH_REQUEST, __opus_check_int_ptr(x)
/**@}*/
/** @defgroup opus_libinfo Opus library information functions
* @{
*/
/** Converts an opus error code into a human readable string.
*
* @param[in] error <tt>int</tt>: Error number
* @returns Error string
*/
OPUS_EXPORT const char *opus_strerror(int error);
/** Gets the libopus version string.
*
* Applications may look for the substring "-fixed" in the version string to
* determine whether they have a fixed-point or floating-point build at
* runtime.
*
* @returns Version string
*/
OPUS_EXPORT const char *opus_get_version_string(void);
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_DEFINES_H */
libs/SDL3_mixer/VisualC/external/include/opus/opus_multistream.h
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+660
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@@ -0,0 +1,660 @@
/* Copyright (c) 2011 Xiph.Org Foundation
Written by Jean-Marc Valin */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/**
* @file opus_multistream.h
* @brief Opus reference implementation multistream API
*/
#ifndef OPUS_MULTISTREAM_H
#define OPUS_MULTISTREAM_H
#include <opus/opus.h>
#ifdef __cplusplus
extern "C" {
#endif
/** @cond OPUS_INTERNAL_DOC */
/** Macros to trigger compilation errors when the wrong types are provided to a
* CTL. */
/**@{*/
#define __opus_check_encstate_ptr(ptr) ((ptr) + ((ptr) - (OpusEncoder**)(ptr)))
#define __opus_check_decstate_ptr(ptr) ((ptr) + ((ptr) - (OpusDecoder**)(ptr)))
/**@}*/
/** These are the actual encoder and decoder CTL ID numbers.
* They should not be used directly by applications.
* In general, SETs should be even and GETs should be odd.*/
/**@{*/
#define OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST 5120
#define OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST 5122
/**@}*/
/** @endcond */
/** @defgroup opus_multistream_ctls Multistream specific encoder and decoder CTLs
*
* These are convenience macros that are specific to the
* opus_multistream_encoder_ctl() and opus_multistream_decoder_ctl()
* interface.
* The CTLs from @ref opus_genericctls, @ref opus_encoderctls, and
* @ref opus_decoderctls may be applied to a multistream encoder or decoder as
* well.
* In addition, you may retrieve the encoder or decoder state for an specific
* stream via #OPUS_MULTISTREAM_GET_ENCODER_STATE or
* #OPUS_MULTISTREAM_GET_DECODER_STATE and apply CTLs to it individually.
*/
/**@{*/
/** Gets the encoder state for an individual stream of a multistream encoder.
* @param[in] x <tt>opus_int32</tt>: The index of the stream whose encoder you
* wish to retrieve.
* This must be non-negative and less than
* the <code>streams</code> parameter used
* to initialize the encoder.
* @param[out] y <tt>OpusEncoder**</tt>: Returns a pointer to the given
* encoder state.
* @retval OPUS_BAD_ARG The index of the requested stream was out of range.
* @hideinitializer
*/
#define OPUS_MULTISTREAM_GET_ENCODER_STATE(x,y) OPUS_MULTISTREAM_GET_ENCODER_STATE_REQUEST, __opus_check_int(x), __opus_check_encstate_ptr(y)
/** Gets the decoder state for an individual stream of a multistream decoder.
* @param[in] x <tt>opus_int32</tt>: The index of the stream whose decoder you
* wish to retrieve.
* This must be non-negative and less than
* the <code>streams</code> parameter used
* to initialize the decoder.
* @param[out] y <tt>OpusDecoder**</tt>: Returns a pointer to the given
* decoder state.
* @retval OPUS_BAD_ARG The index of the requested stream was out of range.
* @hideinitializer
*/
#define OPUS_MULTISTREAM_GET_DECODER_STATE(x,y) OPUS_MULTISTREAM_GET_DECODER_STATE_REQUEST, __opus_check_int(x), __opus_check_decstate_ptr(y)
/**@}*/
/** @defgroup opus_multistream Opus Multistream API
* @{
*
* The multistream API allows individual Opus streams to be combined into a
* single packet, enabling support for up to 255 channels. Unlike an
* elementary Opus stream, the encoder and decoder must negotiate the channel
* configuration before the decoder can successfully interpret the data in the
* packets produced by the encoder. Some basic information, such as packet
* duration, can be computed without any special negotiation.
*
* The format for multistream Opus packets is defined in
* <a href="https://tools.ietf.org/html/rfc7845">RFC 7845</a>
* and is based on the self-delimited Opus framing described in Appendix B of
* <a href="https://tools.ietf.org/html/rfc6716">RFC 6716</a>.
* Normal Opus packets are just a degenerate case of multistream Opus packets,
* and can be encoded or decoded with the multistream API by setting
* <code>streams</code> to <code>1</code> when initializing the encoder or
* decoder.
*
* Multistream Opus streams can contain up to 255 elementary Opus streams.
* These may be either "uncoupled" or "coupled", indicating that the decoder
* is configured to decode them to either 1 or 2 channels, respectively.
* The streams are ordered so that all coupled streams appear at the
* beginning.
*
* A <code>mapping</code> table defines which decoded channel <code>i</code>
* should be used for each input/output (I/O) channel <code>j</code>. This table is
* typically provided as an unsigned char array.
* Let <code>i = mapping[j]</code> be the index for I/O channel <code>j</code>.
* If <code>i < 2*coupled_streams</code>, then I/O channel <code>j</code> is
* encoded as the left channel of stream <code>(i/2)</code> if <code>i</code>
* is even, or as the right channel of stream <code>(i/2)</code> if
* <code>i</code> is odd. Otherwise, I/O channel <code>j</code> is encoded as
* mono in stream <code>(i - coupled_streams)</code>, unless it has the special
* value 255, in which case it is omitted from the encoding entirely (the
* decoder will reproduce it as silence). Each value <code>i</code> must either
* be the special value 255 or be less than <code>streams + coupled_streams</code>.
*
* The output channels specified by the encoder
* should use the
* <a href="https://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-810004.3.9">Vorbis
* channel ordering</a>. A decoder may wish to apply an additional permutation
* to the mapping the encoder used to achieve a different output channel
* order (e.g. for outputing in WAV order).
*
* Each multistream packet contains an Opus packet for each stream, and all of
* the Opus packets in a single multistream packet must have the same
* duration. Therefore the duration of a multistream packet can be extracted
* from the TOC sequence of the first stream, which is located at the
* beginning of the packet, just like an elementary Opus stream:
*
* @code
* int nb_samples;
* int nb_frames;
* nb_frames = opus_packet_get_nb_frames(data, len);
* if (nb_frames < 1)
* return nb_frames;
* nb_samples = opus_packet_get_samples_per_frame(data, 48000) * nb_frames;
* @endcode
*
* The general encoding and decoding process proceeds exactly the same as in
* the normal @ref opus_encoder and @ref opus_decoder APIs.
* See their documentation for an overview of how to use the corresponding
* multistream functions.
*/
/** Opus multistream encoder state.
* This contains the complete state of a multistream Opus encoder.
* It is position independent and can be freely copied.
* @see opus_multistream_encoder_create
* @see opus_multistream_encoder_init
*/
typedef struct OpusMSEncoder OpusMSEncoder;
/** Opus multistream decoder state.
* This contains the complete state of a multistream Opus decoder.
* It is position independent and can be freely copied.
* @see opus_multistream_decoder_create
* @see opus_multistream_decoder_init
*/
typedef struct OpusMSDecoder OpusMSDecoder;
/**\name Multistream encoder functions */
/**@{*/
/** Gets the size of an OpusMSEncoder structure.
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_encoder_get_size(
int streams,
int coupled_streams
);
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_surround_encoder_get_size(
int channels,
int mapping_family
);
/** Allocates and initializes a multistream encoder state.
* Call opus_multistream_encoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than the number of channels.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than the number of input channels.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* encoded channels to input channels, as described in
* @ref opus_multistream. As an extra constraint, the
* multistream encoder does not allow encoding coupled
* streams for which one channel is unused since this
* is never a good idea.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_encoder_create(
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping,
int application,
int *error
) OPUS_ARG_NONNULL(5);
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSEncoder *opus_multistream_surround_encoder_create(
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
unsigned char *mapping,
int application,
int *error
) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
/** Initialize a previously allocated multistream encoder state.
* The memory pointed to by \a st must be at least the size returned by
* opus_multistream_encoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_multistream_encoder_create
* @see opus_multistream_encoder_get_size
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than the number of channels.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than the number of input channels.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* encoded channels to input channels, as described in
* @ref opus_multistream. As an extra constraint, the
* multistream encoder does not allow encoding coupled
* streams for which one channel is unused since this
* is never a good idea.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_multistream_encoder_init(
OpusMSEncoder *st,
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping,
int application
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
OPUS_EXPORT int opus_multistream_surround_encoder_init(
OpusMSEncoder *st,
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
unsigned char *mapping,
int application
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6) OPUS_ARG_NONNULL(7);
/** Encodes a multistream Opus frame.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
* samples.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode(
OpusMSEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a multistream Opus frame from floating point input.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
* samples with a normal range of
* +/-1.0.
* Samples with a range beyond +/-1.0
* are supported but will be clipped by
* decoders using the integer API and
* should only be used if it is known
* that the far end supports extended
* dynamic range.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_encode_float(
OpusMSEncoder *st,
const float *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Frees an <code>OpusMSEncoder</code> allocated by
* opus_multistream_encoder_create().
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to be freed.
*/
OPUS_EXPORT void opus_multistream_encoder_destroy(OpusMSEncoder *st);
/** Perform a CTL function on a multistream Opus encoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_encoderctls, or @ref opus_multistream_ctls.
* @see opus_genericctls
* @see opus_encoderctls
* @see opus_multistream_ctls
*/
OPUS_EXPORT int opus_multistream_encoder_ctl(OpusMSEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
/**\name Multistream decoder functions */
/**@{*/
/** Gets the size of an <code>OpusMSDecoder</code> structure.
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_multistream_decoder_get_size(
int streams,
int coupled_streams
);
/** Allocates and initializes a multistream decoder state.
* Call opus_multistream_decoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* coded channels to output channels, as described in
* @ref opus_multistream.
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusMSDecoder *opus_multistream_decoder_create(
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping,
int *error
) OPUS_ARG_NONNULL(5);
/** Intialize a previously allocated decoder state object.
* The memory pointed to by \a st must be at least the size returned by
* opus_multistream_encoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_multistream_decoder_create
* @see opus_multistream_deocder_get_size
* @param st <tt>OpusMSEncoder*</tt>: Multistream encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] mapping <code>const unsigned char[channels]</code>: Mapping from
* coded channels to output channels, as described in
* @ref opus_multistream.
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_multistream_decoder_init(
OpusMSDecoder *st,
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
const unsigned char *mapping
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
/** Decode a multistream Opus packet.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode(
OpusMSDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int16 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode a multistream Opus packet with floating point output.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_multistream_decode_float(
OpusMSDecoder *st,
const unsigned char *data,
opus_int32 len,
float *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on a multistream Opus decoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusMSDecoder*</tt>: Multistream decoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_decoderctls, or @ref opus_multistream_ctls.
* @see opus_genericctls
* @see opus_decoderctls
* @see opus_multistream_ctls
*/
OPUS_EXPORT int opus_multistream_decoder_ctl(OpusMSDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/** Frees an <code>OpusMSDecoder</code> allocated by
* opus_multistream_decoder_create().
* @param st <tt>OpusMSDecoder</tt>: Multistream decoder state to be freed.
*/
OPUS_EXPORT void opus_multistream_decoder_destroy(OpusMSDecoder *st);
/**@}*/
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_MULTISTREAM_H */
libs/SDL3_mixer/VisualC/external/include/opus/opus_projection.h
Vendored
+568
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@@ -0,0 +1,568 @@
/* Copyright (c) 2017 Google Inc.
Written by Andrew Allen */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/**
* @file opus_projection.h
* @brief Opus projection reference API
*/
#ifndef OPUS_PROJECTION_H
#define OPUS_PROJECTION_H
#include "opus_multistream.h"
#ifdef __cplusplus
extern "C" {
#endif
/** @cond OPUS_INTERNAL_DOC */
/** These are the actual encoder and decoder CTL ID numbers.
* They should not be used directly by applications.c
* In general, SETs should be even and GETs should be odd.*/
/**@{*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST 6001
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST 6003
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST 6005
/**@}*/
/** @endcond */
/** @defgroup opus_projection_ctls Projection specific encoder and decoder CTLs
*
* These are convenience macros that are specific to the
* opus_projection_encoder_ctl() and opus_projection_decoder_ctl()
* interface.
* The CTLs from @ref opus_genericctls, @ref opus_encoderctls,
* @ref opus_decoderctls, and @ref opus_multistream_ctls may be applied to a
* projection encoder or decoder as well.
*/
/**@{*/
/** Gets the gain (in dB. S7.8-format) of the demixing matrix from the encoder.
* @param[out] x <tt>opus_int32 *</tt>: Returns the gain (in dB. S7.8-format)
* of the demixing matrix.
* @hideinitializer
*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_GAIN_REQUEST, __opus_check_int_ptr(x)
/** Gets the size in bytes of the demixing matrix from the encoder.
* @param[out] x <tt>opus_int32 *</tt>: Returns the size in bytes of the
* demixing matrix.
* @hideinitializer
*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE(x) OPUS_PROJECTION_GET_DEMIXING_MATRIX_SIZE_REQUEST, __opus_check_int_ptr(x)
/** Copies the demixing matrix to the supplied pointer location.
* @param[out] x <tt>unsigned char *</tt>: Returns the demixing matrix to the
* supplied pointer location.
* @param y <tt>opus_int32</tt>: The size in bytes of the reserved memory at the
* pointer location.
* @hideinitializer
*/
#define OPUS_PROJECTION_GET_DEMIXING_MATRIX(x,y) OPUS_PROJECTION_GET_DEMIXING_MATRIX_REQUEST, x, __opus_check_int(y)
/**@}*/
/** Opus projection encoder state.
* This contains the complete state of a projection Opus encoder.
* It is position independent and can be freely copied.
* @see opus_projection_ambisonics_encoder_create
*/
typedef struct OpusProjectionEncoder OpusProjectionEncoder;
/** Opus projection decoder state.
* This contains the complete state of a projection Opus decoder.
* It is position independent and can be freely copied.
* @see opus_projection_decoder_create
* @see opus_projection_decoder_init
*/
typedef struct OpusProjectionDecoder OpusProjectionDecoder;
/**\name Projection encoder functions */
/**@{*/
/** Gets the size of an OpusProjectionEncoder structure.
* @param channels <tt>int</tt>: The total number of input channels to encode.
* This must be no more than 255.
* @param mapping_family <tt>int</tt>: The mapping family to use for selecting
* the appropriate projection.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_ambisonics_encoder_get_size(
int channels,
int mapping_family
);
/** Allocates and initializes a projection encoder state.
* Call opus_projection_encoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param mapping_family <tt>int</tt>: The mapping family to use for selecting
* the appropriate projection.
* @param[out] streams <tt>int *</tt>: The total number of streams that will
* be encoded from the input.
* @param[out] coupled_streams <tt>int *</tt>: Number of coupled (2 channel)
* streams that will be encoded from the input.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionEncoder *opus_projection_ambisonics_encoder_create(
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
int application,
int *error
) OPUS_ARG_NONNULL(4) OPUS_ARG_NONNULL(5);
/** Initialize a previously allocated projection encoder state.
* The memory pointed to by \a st must be at least the size returned by
* opus_projection_ambisonics_encoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_projection_ambisonics_encoder_create
* @see opus_projection_ambisonics_encoder_get_size
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate of the input signal (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels in the input signal.
* This must be at most 255.
* It may be greater than the number of
* coded channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams to encode from the
* input.
* This must be no more than the number of channels.
* @param coupled_streams <tt>int</tt>: Number of coupled (2 channel) streams
* to encode.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* encoded channels (<code>streams +
* coupled_streams</code>) must be no
* more than the number of input channels.
* @param application <tt>int</tt>: The target encoder application.
* This must be one of the following:
* <dl>
* <dt>#OPUS_APPLICATION_VOIP</dt>
* <dd>Process signal for improved speech intelligibility.</dd>
* <dt>#OPUS_APPLICATION_AUDIO</dt>
* <dd>Favor faithfulness to the original input.</dd>
* <dt>#OPUS_APPLICATION_RESTRICTED_LOWDELAY</dt>
* <dd>Configure the minimum possible coding delay by disabling certain modes
* of operation.</dd>
* </dl>
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_projection_ambisonics_encoder_init(
OpusProjectionEncoder *st,
opus_int32 Fs,
int channels,
int mapping_family,
int *streams,
int *coupled_streams,
int application
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(5) OPUS_ARG_NONNULL(6);
/** Encodes a projection Opus frame.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
* @param[in] pcm <tt>const opus_int16*</tt>: The input signal as interleaved
* samples.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode(
OpusProjectionEncoder *st,
const opus_int16 *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Encodes a projection Opus frame from floating point input.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
* @param[in] pcm <tt>const float*</tt>: The input signal as interleaved
* samples with a normal range of
* +/-1.0.
* Samples with a range beyond +/-1.0
* are supported but will be clipped by
* decoders using the integer API and
* should only be used if it is known
* that the far end supports extended
* dynamic range.
* This must contain
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: Number of samples per channel in the input
* signal.
* This must be an Opus frame size for the
* encoder's sampling rate.
* For example, at 48 kHz the permitted values
* are 120, 240, 480, 960, 1920, and 2880.
* Passing in a duration of less than 10 ms
* (480 samples at 48 kHz) will prevent the
* encoder from using the LPC or hybrid modes.
* @param[out] data <tt>unsigned char*</tt>: Output payload.
* This must contain storage for at
* least \a max_data_bytes.
* @param [in] max_data_bytes <tt>opus_int32</tt>: Size of the allocated
* memory for the output
* payload. This may be
* used to impose an upper limit on
* the instant bitrate, but should
* not be used as the only bitrate
* control. Use #OPUS_SET_BITRATE to
* control the bitrate.
* @returns The length of the encoded packet (in bytes) on success or a
* negative error code (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_encode_float(
OpusProjectionEncoder *st,
const float *pcm,
int frame_size,
unsigned char *data,
opus_int32 max_data_bytes
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(2) OPUS_ARG_NONNULL(4);
/** Frees an <code>OpusProjectionEncoder</code> allocated by
* opus_projection_ambisonics_encoder_create().
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state to be freed.
*/
OPUS_EXPORT void opus_projection_encoder_destroy(OpusProjectionEncoder *st);
/** Perform a CTL function on a projection Opus encoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusProjectionEncoder*</tt>: Projection encoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_encoderctls, @ref opus_multistream_ctls, or
* @ref opus_projection_ctls
* @see opus_genericctls
* @see opus_encoderctls
* @see opus_multistream_ctls
* @see opus_projection_ctls
*/
OPUS_EXPORT int opus_projection_encoder_ctl(OpusProjectionEncoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/**@}*/
/**\name Projection decoder functions */
/**@{*/
/** Gets the size of an <code>OpusProjectionDecoder</code> structure.
* @param channels <tt>int</tt>: The total number of output channels.
* This must be no more than 255.
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @returns The size in bytes on success, or a negative error code
* (see @ref opus_errorcodes) on error.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT opus_int32 opus_projection_decoder_get_size(
int channels,
int streams,
int coupled_streams
);
/** Allocates and initializes a projection decoder state.
* Call opus_projection_decoder_destroy() to release
* this object when finished.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
* that mapping from coded channels to output channels,
* as described in @ref opus_projection and
* @ref opus_projection_ctls.
* @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
* demixing matrix, as
* described in @ref
* opus_projection_ctls.
* @param[out] error <tt>int *</tt>: Returns #OPUS_OK on success, or an error
* code (see @ref opus_errorcodes) on
* failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT OpusProjectionDecoder *opus_projection_decoder_create(
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
unsigned char *demixing_matrix,
opus_int32 demixing_matrix_size,
int *error
) OPUS_ARG_NONNULL(5);
/** Intialize a previously allocated projection decoder state object.
* The memory pointed to by \a st must be at least the size returned by
* opus_projection_decoder_get_size().
* This is intended for applications which use their own allocator instead of
* malloc.
* To reset a previously initialized state, use the #OPUS_RESET_STATE CTL.
* @see opus_projection_decoder_create
* @see opus_projection_deocder_get_size
* @param st <tt>OpusProjectionDecoder*</tt>: Projection encoder state to initialize.
* @param Fs <tt>opus_int32</tt>: Sampling rate to decode at (in Hz).
* This must be one of 8000, 12000, 16000,
* 24000, or 48000.
* @param channels <tt>int</tt>: Number of channels to output.
* This must be at most 255.
* It may be different from the number of coded
* channels (<code>streams +
* coupled_streams</code>).
* @param streams <tt>int</tt>: The total number of streams coded in the
* input.
* This must be no more than 255.
* @param coupled_streams <tt>int</tt>: Number of streams to decode as coupled
* (2 channel) streams.
* This must be no larger than the total
* number of streams.
* Additionally, The total number of
* coded channels (<code>streams +
* coupled_streams</code>) must be no
* more than 255.
* @param[in] demixing_matrix <tt>const unsigned char[demixing_matrix_size]</tt>: Demixing matrix
* that mapping from coded channels to output channels,
* as described in @ref opus_projection and
* @ref opus_projection_ctls.
* @param demixing_matrix_size <tt>opus_int32</tt>: The size in bytes of the
* demixing matrix, as
* described in @ref
* opus_projection_ctls.
* @returns #OPUS_OK on success, or an error code (see @ref opus_errorcodes)
* on failure.
*/
OPUS_EXPORT int opus_projection_decoder_init(
OpusProjectionDecoder *st,
opus_int32 Fs,
int channels,
int streams,
int coupled_streams,
unsigned char *demixing_matrix,
opus_int32 demixing_matrix_size
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(6);
/** Decode a projection Opus packet.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode(
OpusProjectionDecoder *st,
const unsigned char *data,
opus_int32 len,
opus_int16 *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Decode a projection Opus packet with floating point output.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
* @param[in] data <tt>const unsigned char*</tt>: Input payload.
* Use a <code>NULL</code>
* pointer to indicate packet
* loss.
* @param len <tt>opus_int32</tt>: Number of bytes in payload.
* @param[out] pcm <tt>opus_int16*</tt>: Output signal, with interleaved
* samples.
* This must contain room for
* <code>frame_size*channels</code>
* samples.
* @param frame_size <tt>int</tt>: The number of samples per channel of
* available space in \a pcm.
* If this is less than the maximum packet duration
* (120 ms; 5760 for 48kHz), this function will not be capable
* of decoding some packets. In the case of PLC (data==NULL)
* or FEC (decode_fec=1), then frame_size needs to be exactly
* the duration of audio that is missing, otherwise the
* decoder will not be in the optimal state to decode the
* next incoming packet. For the PLC and FEC cases, frame_size
* <b>must</b> be a multiple of 2.5 ms.
* @param decode_fec <tt>int</tt>: Flag (0 or 1) to request that any in-band
* forward error correction data be decoded.
* If no such data is available, the frame is
* decoded as if it were lost.
* @returns Number of samples decoded on success or a negative error code
* (see @ref opus_errorcodes) on failure.
*/
OPUS_EXPORT OPUS_WARN_UNUSED_RESULT int opus_projection_decode_float(
OpusProjectionDecoder *st,
const unsigned char *data,
opus_int32 len,
float *pcm,
int frame_size,
int decode_fec
) OPUS_ARG_NONNULL(1) OPUS_ARG_NONNULL(4);
/** Perform a CTL function on a projection Opus decoder.
*
* Generally the request and subsequent arguments are generated by a
* convenience macro.
* @param st <tt>OpusProjectionDecoder*</tt>: Projection decoder state.
* @param request This and all remaining parameters should be replaced by one
* of the convenience macros in @ref opus_genericctls,
* @ref opus_decoderctls, @ref opus_multistream_ctls, or
* @ref opus_projection_ctls.
* @see opus_genericctls
* @see opus_decoderctls
* @see opus_multistream_ctls
* @see opus_projection_ctls
*/
OPUS_EXPORT int opus_projection_decoder_ctl(OpusProjectionDecoder *st, int request, ...) OPUS_ARG_NONNULL(1);
/** Frees an <code>OpusProjectionDecoder</code> allocated by
* opus_projection_decoder_create().
* @param st <tt>OpusProjectionDecoder</tt>: Projection decoder state to be freed.
*/
OPUS_EXPORT void opus_projection_decoder_destroy(OpusProjectionDecoder *st);
/**@}*/
/**@}*/
#ifdef __cplusplus
}
#endif
#endif /* OPUS_PROJECTION_H */
libs/SDL3_mixer/VisualC/external/include/opus/opus_types.h
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/* (C) COPYRIGHT 1994-2002 Xiph.Org Foundation */
/* Modified by Jean-Marc Valin */
/*
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/* opus_types.h based on ogg_types.h from libogg */
/**
@file opus_types.h
@brief Opus reference implementation types
*/
#ifndef OPUS_TYPES_H
#define OPUS_TYPES_H
#define opus_int int /* used for counters etc; at least 16 bits */
#define opus_int64 long long
#define opus_int8 signed char
#define opus_uint unsigned int /* used for counters etc; at least 16 bits */
#define opus_uint64 unsigned long long
#define opus_uint8 unsigned char
/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
#if (defined(__STDC__) && __STDC__ && defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
#include <stdint.h>
# undef opus_int64
# undef opus_int8
# undef opus_uint64
# undef opus_uint8
typedef int8_t opus_int8;
typedef uint8_t opus_uint8;
typedef int16_t opus_int16;
typedef uint16_t opus_uint16;
typedef int32_t opus_int32;
typedef uint32_t opus_uint32;
typedef int64_t opus_int64;
typedef uint64_t opus_uint64;
#elif defined(_WIN32)
# if defined(__CYGWIN__)
# include <_G_config.h>
typedef _G_int32_t opus_int32;
typedef _G_uint32_t opus_uint32;
typedef _G_int16 opus_int16;
typedef _G_uint16 opus_uint16;
# elif defined(__MINGW32__)
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
# elif defined(__MWERKS__)
typedef int opus_int32;
typedef unsigned int opus_uint32;
typedef short opus_int16;
typedef unsigned short opus_uint16;
# else
/* MSVC/Borland */
typedef __int32 opus_int32;
typedef unsigned __int32 opus_uint32;
typedef __int16 opus_int16;
typedef unsigned __int16 opus_uint16;
# endif
#elif defined(__MACOS__)
# include <sys/types.h>
typedef SInt16 opus_int16;
typedef UInt16 opus_uint16;
typedef SInt32 opus_int32;
typedef UInt32 opus_uint32;
#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
# include <sys/types.h>
typedef int16_t opus_int16;
typedef u_int16_t opus_uint16;
typedef int32_t opus_int32;
typedef u_int32_t opus_uint32;
#elif defined(__BEOS__)
/* Be */
# include <inttypes.h>
typedef int16 opus_int16;
typedef u_int16 opus_uint16;
typedef int32_t opus_int32;
typedef u_int32_t opus_uint32;
#elif defined (__EMX__)
/* OS/2 GCC */
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#elif defined (DJGPP)
/* DJGPP */
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#elif defined(R5900)
/* PS2 EE */
typedef int opus_int32;
typedef unsigned opus_uint32;
typedef short opus_int16;
typedef unsigned short opus_uint16;
#elif defined(__SYMBIAN32__)
/* Symbian GCC */
typedef signed short opus_int16;
typedef unsigned short opus_uint16;
typedef signed int opus_int32;
typedef unsigned int opus_uint32;
#elif defined(CONFIG_TI_C54X) || defined (CONFIG_TI_C55X)
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef long opus_int32;
typedef unsigned long opus_uint32;
#elif defined(CONFIG_TI_C6X)
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#else
/* Give up, take a reasonable guess */
typedef short opus_int16;
typedef unsigned short opus_uint16;
typedef int opus_int32;
typedef unsigned int opus_uint32;
#endif
#endif /* OPUS_TYPES_H */
libs/SDL3_mixer/VisualC/external/include/opus/opusfile.h
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libs/SDL3_mixer/VisualC/external/include/opusfile.h
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/* wrapper to avoid an additional -Iinclude/opus directive.
* headers under opus/ are edited accordingly for this */
#include <opus/opusfile.h>
libs/SDL3_mixer/VisualC/external/include/wavpack/wavpack.h
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////////////////////////////////////////////////////////////////////////////
// **** WAVPACK **** //
// Hybrid Lossless Wavefile Compressor //
// Copyright (c) 1998 - 2024 David Bryant. //
// All Rights Reserved. //
// Distributed under the BSD Software License (see license.txt) //
////////////////////////////////////////////////////////////////////////////
// wavpack.h
#ifndef WAVPACK_H
#define WAVPACK_H
// This header file contains all the definitions required to use the
// functions in "wputils.c" to read and write WavPack files and streams.
#include <sys/types.h>
#if defined(_MSC_VER) && _MSC_VER < 1600
typedef unsigned __int64 uint64_t;
typedef unsigned __int32 uint32_t;
typedef unsigned __int16 uint16_t;
typedef unsigned __int8 uint8_t;
typedef __int64 int64_t;
typedef __int32 int32_t;
typedef __int16 int16_t;
typedef __int8 int8_t;
#else
#include <stdint.h>
#endif
// RIFF / wav header formats (these occur at the beginning of both wav files
// and pre-4.0 WavPack files that are not in the "raw" mode). Generally, an
// application using the library to read or write WavPack files will not be
// concerned with any of these.
typedef struct {
char ckID [4];
uint32_t ckSize;
char formType [4];
} RiffChunkHeader;
typedef struct {
char ckID [4];
uint32_t ckSize;
} ChunkHeader;
#define ChunkHeaderFormat "4L"
typedef struct {
uint16_t FormatTag, NumChannels;
uint32_t SampleRate, BytesPerSecond;
uint16_t BlockAlign, BitsPerSample;
uint16_t cbSize, ValidBitsPerSample;
int32_t ChannelMask;
uint16_t SubFormat;
char GUID [14];
} WaveHeader;
#define WaveHeaderFormat "SSLLSSSSLS"
// This is the ONLY structure that occurs in WavPack files (as of version
// 4.0), and is the preamble to every block in both the .wv and .wvc
// files (in little-endian format). Normally, this structure has no use
// to an application using the library to read or write WavPack files,
// but if an application needs to manually parse WavPack files then this
// would be used (with appropriate endian correction).
typedef struct {
char ckID [4];
uint32_t ckSize;
int16_t version;
unsigned char block_index_u8;
unsigned char total_samples_u8;
uint32_t total_samples, block_index, block_samples, flags, crc;
} WavpackHeader;
#define WavpackHeaderFormat "4LS2LLLLL"
// Macros to access the 40-bit block_index field
#define GET_BLOCK_INDEX(hdr) ( (int64_t) (hdr).block_index + ((int64_t) (hdr).block_index_u8 << 32) )
#define SET_BLOCK_INDEX(hdr,value) do { \
int64_t tmp = (value); \
(hdr).block_index = (uint32_t) tmp; \
(hdr).block_index_u8 = \
(unsigned char) (tmp >> 32); \
} while (0)
// Macros to access the 40-bit total_samples field, which is complicated by the fact that
// all 1's in the lower 32 bits indicates "unknown" (regardless of upper 8 bits)
#define GET_TOTAL_SAMPLES(hdr) ( ((hdr).total_samples == (uint32_t) -1) ? -1 : \
(int64_t) (hdr).total_samples + ((int64_t) (hdr).total_samples_u8 << 32) - (hdr).total_samples_u8 )
#define SET_TOTAL_SAMPLES(hdr,value) do { \
int64_t tmp = (value); \
if (tmp < 0) \
(hdr).total_samples = (uint32_t) -1; \
else { \
tmp += (tmp / 0xffffffffLL); \
(hdr).total_samples = (uint32_t) tmp; \
(hdr).total_samples_u8 = \
(unsigned char) (tmp >> 32); \
} \
} while (0)
// or-values for WavpackHeader.flags
#define BYTES_STORED 3 // 1-4 bytes/sample
#define MONO_FLAG 4 // not stereo
#define HYBRID_FLAG 8 // hybrid mode
#define JOINT_STEREO 0x10 // joint stereo
#define CROSS_DECORR 0x20 // no-delay cross decorrelation
#define HYBRID_SHAPE 0x40 // noise shape (hybrid mode only)
#define FLOAT_DATA 0x80 // ieee 32-bit floating point data
#define INT32_DATA 0x100 // special extended int handling
#define HYBRID_BITRATE 0x200 // bitrate noise (hybrid mode only)
#define HYBRID_BALANCE 0x400 // balance noise (hybrid stereo mode only)
#define INITIAL_BLOCK 0x800 // initial block of multichannel segment
#define FINAL_BLOCK 0x1000 // final block of multichannel segment
#define SHIFT_LSB 13
#define SHIFT_MASK (0x1fL << SHIFT_LSB)
#define MAG_LSB 18
#define MAG_MASK (0x1fL << MAG_LSB)
#define SRATE_LSB 23
#define SRATE_MASK (0xfL << SRATE_LSB)
#define FALSE_STEREO 0x40000000 // block is stereo, but data is mono
#define NEW_SHAPING 0x20000000 // use IIR filter for negative shaping
#define MONO_DATA (MONO_FLAG | FALSE_STEREO)
// Introduced in WavPack 5.0:
#define HAS_CHECKSUM 0x10000000 // block contains a trailing checksum
#define DSD_FLAG 0x80000000 // block is encoded DSD (1-bit PCM)
#define IGNORED_FLAGS 0x08000000 // reserved, but ignore if encountered
#define UNKNOWN_FLAGS 0x00000000 // we no longer have any of these spares
#define MIN_STREAM_VERS 0x402 // lowest stream version we'll decode
#define MAX_STREAM_VERS 0x410 // highest stream version we'll decode or encode
#define WAVPACK_MAX_CHANS 4096 // max channels handled by WavPack format & library
// This sets the maximum number of channels that the current WavPack CLI applications
// accept. It's somewhat arbitrary because the actual WavPack format and library can
// handle up to 4096 channels. However, anything beyond 256 channels is obviously
// a niche case and is not well tested, so this lower limit is defined for now.
#define WAVPACK_MAX_CLI_CHANS 256
// These are the mask bit definitions for the metadata chunk id byte (see format.txt)
#define ID_UNIQUE 0x3f
#define ID_OPTIONAL_DATA 0x20
#define ID_ODD_SIZE 0x40
#define ID_LARGE 0x80
#define ID_DUMMY 0x0
#define ID_ENCODER_INFO 0x1
#define ID_DECORR_TERMS 0x2
#define ID_DECORR_WEIGHTS 0x3
#define ID_DECORR_SAMPLES 0x4
#define ID_ENTROPY_VARS 0x5
#define ID_HYBRID_PROFILE 0x6
#define ID_SHAPING_WEIGHTS 0x7
#define ID_FLOAT_INFO 0x8
#define ID_INT32_INFO 0x9
#define ID_WV_BITSTREAM 0xa
#define ID_WVC_BITSTREAM 0xb
#define ID_WVX_BITSTREAM 0xc
#define ID_CHANNEL_INFO 0xd
#define ID_DSD_BLOCK 0xe
#define ID_RIFF_HEADER (ID_OPTIONAL_DATA | 0x1)
#define ID_RIFF_TRAILER (ID_OPTIONAL_DATA | 0x2)
#define ID_ALT_HEADER (ID_OPTIONAL_DATA | 0x3)
#define ID_ALT_TRAILER (ID_OPTIONAL_DATA | 0x4)
#define ID_CONFIG_BLOCK (ID_OPTIONAL_DATA | 0x5)
#define ID_MD5_CHECKSUM (ID_OPTIONAL_DATA | 0x6)
#define ID_SAMPLE_RATE (ID_OPTIONAL_DATA | 0x7)
#define ID_ALT_EXTENSION (ID_OPTIONAL_DATA | 0x8)
#define ID_ALT_MD5_CHECKSUM (ID_OPTIONAL_DATA | 0x9)
#define ID_NEW_CONFIG_BLOCK (ID_OPTIONAL_DATA | 0xa)
#define ID_CHANNEL_IDENTITIES (ID_OPTIONAL_DATA | 0xb)
#define ID_WVX_NEW_BITSTREAM (ID_OPTIONAL_DATA | ID_WVX_BITSTREAM)
#define ID_BLOCK_CHECKSUM (ID_OPTIONAL_DATA | 0xf)
///////////////////////// WavPack Configuration ///////////////////////////////
// This external structure is used during encode to provide configuration to
// the encoding engine and during decoding to provide fle information back to
// the higher level functions. Not all fields are used in both modes.
typedef struct {
float bitrate, shaping_weight;
int bits_per_sample, bytes_per_sample;
int qmode, flags, xmode, num_channels, float_norm_exp;
int32_t block_samples, worker_threads, sample_rate, channel_mask;
unsigned char md5_checksum [16], md5_read;
int num_tag_strings; // this field is not used
char **tag_strings; // this field is not used
} WavpackConfig;
#define CONFIG_HYBRID_FLAG 8 // hybrid mode
#define CONFIG_JOINT_STEREO 0x10 // joint stereo
#define CONFIG_CROSS_DECORR 0x20 // no-delay cross decorrelation
#define CONFIG_HYBRID_SHAPE 0x40 // noise shape (hybrid mode only)
#define CONFIG_FAST_FLAG 0x200 // fast mode
#define CONFIG_HIGH_FLAG 0x800 // high quality mode
#define CONFIG_VERY_HIGH_FLAG 0x1000 // very high
#define CONFIG_BITRATE_KBPS 0x2000 // bitrate is kbps, not bits / sample
#define CONFIG_SHAPE_OVERRIDE 0x8000 // shaping mode specified
#define CONFIG_JOINT_OVERRIDE 0x10000 // joint-stereo mode specified
#define CONFIG_DYNAMIC_SHAPING 0x20000 // dynamic noise shaping
#define CONFIG_CREATE_EXE 0x40000 // create executable
#define CONFIG_CREATE_WVC 0x80000 // create correction file
#define CONFIG_OPTIMIZE_WVC 0x100000 // maximize hybrid compression
#define CONFIG_COMPATIBLE_WRITE 0x400000 // write files for decoders < 4.3
#define CONFIG_CALC_NOISE 0x800000 // calc noise in hybrid mode
#define CONFIG_EXTRA_MODE 0x2000000 // extra processing mode
#define CONFIG_SKIP_WVX 0x4000000 // no wvx stream w/ floats & big ints
#define CONFIG_MD5_CHECKSUM 0x8000000 // store MD5 signature
#define CONFIG_MERGE_BLOCKS 0x10000000 // merge blocks of equal redundancy (for lossyWAV)
#define CONFIG_PAIR_UNDEF_CHANS 0x20000000 // encode undefined channels in stereo pairs
#define CONFIG_OPTIMIZE_32BIT 0x40000000 // new optimizations for 32-bit integer files
#define CONFIG_OPTIMIZE_MONO 0x80000000 // optimize for mono streams posing as stereo
// The lower 8 bits of qmode indicate the use of new features in version 5 that (presently)
// only apply to Core Audio Files (CAF) and DSD files, but could apply to other things too.
// These flags are stored in the file and can be retrieved by a decoder that is aware of
// them, but the individual bits are meaningless to the library. If ANY of these bits are
// set then the MD5 sum is written with a new ID so that old decoders will not see it
// (because these features will cause the MD5 sum to be different and fail).
#define QMODE_BIG_ENDIAN 0x1 // big-endian data format (opposite of WAV format)
#define QMODE_SIGNED_BYTES 0x2 // 8-bit audio data is signed (opposite of WAV format)
#define QMODE_UNSIGNED_WORDS 0x4 // audio data (other than 8-bit) is unsigned (opposite of WAV format)
#define QMODE_REORDERED_CHANS 0x8 // source channels were not Microsoft order, so they were reordered
#define QMODE_DSD_LSB_FIRST 0x10 // DSD bytes, LSB first (most Sony .dsf files)
#define QMODE_DSD_MSB_FIRST 0x20 // DSD bytes, MSB first (Philips .dff files)
#define QMODE_DSD_IN_BLOCKS 0x40 // DSD data is blocked by channels (Sony .dsf only)
#define QMODE_DSD_AUDIO (QMODE_DSD_LSB_FIRST | QMODE_DSD_MSB_FIRST)
// The rest of the qmode word is reserved for the private use of the command-line programs
// and are ignored by the library (and not stored either). They really should not be defined
// here, but I thought it would be a good idea to have all the definitions together.
#define QMODE_ADOBE_MODE 0x100 // user specified Adobe mode
#define QMODE_NO_STORE_WRAPPER 0x200 // user specified to not store audio file wrapper (RIFF, CAFF, etc.)
#define QMODE_CHANS_UNASSIGNED 0x400 // user specified "..." in --channel-order option
#define QMODE_IGNORE_LENGTH 0x800 // user specified to ignore length in file header
#define QMODE_RAW_PCM 0x1000 // user specified raw PCM format (no header present)
#define QMODE_EVEN_BYTE_DEPTH 0x2000 // user specified to force even byte bit-depth
////////////// Callbacks used for reading & writing WavPack streams //////////
typedef struct {
int32_t (*read_bytes)(void *id, void *data, int32_t bcount);
uint32_t (*get_pos)(void *id);
int (*set_pos_abs)(void *id, uint32_t pos);
int (*set_pos_rel)(void *id, int32_t delta, int mode);
int (*push_back_byte)(void *id, int c);
uint32_t (*get_length)(void *id);
int (*can_seek)(void *id);
// this callback is for writing edited tags only
int32_t (*write_bytes)(void *id, void *data, int32_t bcount);
} WavpackStreamReader;
// Extended version of structure for handling large files and added
// functionality for truncating and closing files
typedef struct {
int32_t (*read_bytes)(void *id, void *data, int32_t bcount);
int32_t (*write_bytes)(void *id, void *data, int32_t bcount);
int64_t (*get_pos)(void *id); // new signature for large files
int (*set_pos_abs)(void *id, int64_t pos); // new signature for large files
int (*set_pos_rel)(void *id, int64_t delta, int mode); // new signature for large files
int (*push_back_byte)(void *id, int c);
int64_t (*get_length)(void *id); // new signature for large files
int (*can_seek)(void *id);
int (*truncate_here)(void *id); // new function to truncate file at current position
int (*close)(void *id); // new function to close file
} WavpackStreamReader64;
typedef int (*WavpackBlockOutput)(void *id, void *data, int32_t bcount);
//////////////////////////// function prototypes /////////////////////////////
typedef struct WavpackContext WavpackContext;
#ifdef __cplusplus
extern "C" {
#endif
#define MAX_WAVPACK_SAMPLES ((1LL << 40) - 257)
WavpackContext *WavpackOpenRawDecoder (
void *main_data, int32_t main_size,
void *corr_data, int32_t corr_size,
int16_t version, char *error, int flags, int norm_offset);
WavpackContext *WavpackOpenFileInputEx64 (WavpackStreamReader64 *reader, void *wv_id, void *wvc_id, char *error, int flags, int norm_offset);
WavpackContext *WavpackOpenFileInputEx (WavpackStreamReader *reader, void *wv_id, void *wvc_id, char *error, int flags, int norm_offset);
WavpackContext *WavpackOpenFileInput (const char *infilename, char *error, int flags, int norm_offset);
#define OPEN_WVC 0x1 // open/read "correction" file
#define OPEN_TAGS 0x2 // read ID3v1 / APEv2 tags (seekable file)
#define OPEN_WRAPPER 0x4 // make audio wrapper available (i.e. RIFF)
#define OPEN_2CH_MAX 0x8 // open multichannel as stereo (no downmix)
#define OPEN_NORMALIZE 0x10 // normalize floating point data to +/- 1.0
#define OPEN_STREAMING 0x20 // "streaming" mode blindly unpacks blocks
// w/o regard to header file position info
#define OPEN_EDIT_TAGS 0x40 // allow editing of tags
#define OPEN_FILE_UTF8 0x80 // assume filenames are UTF-8 encoded, not ANSI (Windows only)
// new for version 5
#define OPEN_DSD_NATIVE 0x100 // open DSD files as bitstreams
// (returned as 8-bit "samples" stored in 32-bit words)
#define OPEN_DSD_AS_PCM 0x200 // open DSD files as 24-bit PCM (decimated 8x)
#define OPEN_ALT_TYPES 0x400 // application is aware of alternate file types & qmode
// (just affects retrieving wrappers & MD5 checksums)
#define OPEN_NO_CHECKSUM 0x800 // don't verify block checksums before decoding
// new for multithreaded
#define OPEN_THREADS_SHFT 12 // specify number of additional worker threads here for
#define OPEN_THREADS_MASK 0xF000 // decode; 0 to disable, otherwise 1-15 added threads
int WavpackGetMode (WavpackContext *wpc);
#define MODE_WVC 0x1
#define MODE_LOSSLESS 0x2
#define MODE_HYBRID 0x4
#define MODE_FLOAT 0x8
#define MODE_VALID_TAG 0x10
#define MODE_HIGH 0x20
#define MODE_FAST 0x40
#define MODE_EXTRA 0x80 // extra mode used, see MODE_XMODE for possible level
#define MODE_APETAG 0x100
#define MODE_SFX 0x200
#define MODE_VERY_HIGH 0x400
#define MODE_MD5 0x800
#define MODE_XMODE 0x7000 // mask for extra level (1-6, 0=unknown)
#define MODE_DNS 0x8000
int WavpackVerifySingleBlock (unsigned char *buffer, int verify_checksum);
int WavpackGetQualifyMode (WavpackContext *wpc);
char *WavpackGetErrorMessage (WavpackContext *wpc);
int WavpackGetVersion (WavpackContext *wpc);
char *WavpackGetFileExtension (WavpackContext *wpc);
unsigned char WavpackGetFileFormat (WavpackContext *wpc);
uint32_t WavpackUnpackSamples (WavpackContext *wpc, int32_t *buffer, uint32_t samples);
uint32_t WavpackGetNumSamples (WavpackContext *wpc);
int64_t WavpackGetNumSamples64 (WavpackContext *wpc);
uint32_t WavpackGetNumSamplesInFrame (WavpackContext *wpc);
uint32_t WavpackGetSampleIndex (WavpackContext *wpc);
int64_t WavpackGetSampleIndex64 (WavpackContext *wpc);
int WavpackGetNumErrors (WavpackContext *wpc);
int WavpackLossyBlocks (WavpackContext *wpc);
int WavpackSeekSample (WavpackContext *wpc, uint32_t sample);
int WavpackSeekSample64 (WavpackContext *wpc, int64_t sample);
WavpackContext *WavpackCloseFile (WavpackContext *wpc);
uint32_t WavpackGetSampleRate (WavpackContext *wpc);
uint32_t WavpackGetNativeSampleRate (WavpackContext *wpc);
int WavpackGetBitsPerSample (WavpackContext *wpc);
int WavpackGetBytesPerSample (WavpackContext *wpc);
int WavpackGetNumChannels (WavpackContext *wpc);
int WavpackGetChannelMask (WavpackContext *wpc);
int WavpackGetReducedChannels (WavpackContext *wpc);
int WavpackGetFloatNormExp (WavpackContext *wpc);
int WavpackGetMD5Sum (WavpackContext *wpc, unsigned char data [16]);
void WavpackGetChannelIdentities (WavpackContext *wpc, unsigned char *identities);
uint32_t WavpackGetChannelLayout (WavpackContext *wpc, unsigned char *reorder);
uint32_t WavpackGetWrapperBytes (WavpackContext *wpc);
unsigned char *WavpackGetWrapperData (WavpackContext *wpc);
void WavpackFreeWrapper (WavpackContext *wpc);
void WavpackSeekTrailingWrapper (WavpackContext *wpc);
double WavpackGetProgress (WavpackContext *wpc);
uint32_t WavpackGetFileSize (WavpackContext *wpc);
int64_t WavpackGetFileSize64 (WavpackContext *wpc);
double WavpackGetRatio (WavpackContext *wpc);
double WavpackGetAverageBitrate (WavpackContext *wpc, int count_wvc);
double WavpackGetInstantBitrate (WavpackContext *wpc);
int WavpackGetNumTagItems (WavpackContext *wpc);
int WavpackGetTagItem (WavpackContext *wpc, const char *item, char *value, int size);
int WavpackGetTagItemIndexed (WavpackContext *wpc, int index, char *item, int size);
int WavpackGetNumBinaryTagItems (WavpackContext *wpc);
int WavpackGetBinaryTagItem (WavpackContext *wpc, const char *item, char *value, int size);
int WavpackGetBinaryTagItemIndexed (WavpackContext *wpc, int index, char *item, int size);
int WavpackAppendTagItem (WavpackContext *wpc, const char *item, const char *value, int vsize);
int WavpackAppendBinaryTagItem (WavpackContext *wpc, const char *item, const char *value, int vsize);
int WavpackDeleteTagItem (WavpackContext *wpc, const char *item);
int WavpackWriteTag (WavpackContext *wpc);
WavpackContext *WavpackOpenFileOutput (WavpackBlockOutput blockout, void *wv_id, void *wvc_id);
void WavpackSetFileInformation (WavpackContext *wpc, char *file_extension, unsigned char file_format);
#define WP_FORMAT_WAV 0 // Microsoft RIFF, including BWF and RF64 variants
#define WP_FORMAT_W64 1 // Sony Wave64
#define WP_FORMAT_CAF 2 // Apple CoreAudio
#define WP_FORMAT_DFF 3 // Philips DSDIFF
#define WP_FORMAT_DSF 4 // Sony DSD Format
#define WP_FORMAT_AIF 5 // Apple AIFF
int WavpackSetConfiguration (WavpackContext *wpc, WavpackConfig *config, uint32_t total_samples);
int WavpackSetConfiguration64 (WavpackContext *wpc, WavpackConfig *config, int64_t total_samples, const unsigned char *chan_ids);
int WavpackSetChannelLayout (WavpackContext *wpc, uint32_t layout_tag, const unsigned char *reorder);
int WavpackAddWrapper (WavpackContext *wpc, void *data, uint32_t bcount);
int WavpackStoreMD5Sum (WavpackContext *wpc, unsigned char data [16]);
int WavpackPackInit (WavpackContext *wpc);
int WavpackPackSamples (WavpackContext *wpc, int32_t *sample_buffer, uint32_t sample_count);
int WavpackFlushSamples (WavpackContext *wpc);
void WavpackUpdateNumSamples (WavpackContext *wpc, void *first_block);
void *WavpackGetWrapperLocation (void *first_block, uint32_t *size);
double WavpackGetEncodedNoise (WavpackContext *wpc, double *peak);
void WavpackFloatNormalize (int32_t *values, int32_t num_values, int delta_exp);
void WavpackLittleEndianToNative (void *data, char *format);
void WavpackNativeToLittleEndian (void *data, char *format);
void WavpackBigEndianToNative (void *data, char *format);
void WavpackNativeToBigEndian (void *data, char *format);
uint32_t WavpackGetLibraryVersion (void);
const char *WavpackGetLibraryVersionString (void);
#ifdef __cplusplus
}
#endif
#endif
libs/SDL3_mixer/VisualC/external/include/xmp.h
Vendored
+416
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@@ -0,0 +1,416 @@
#ifndef XMP_H
#define XMP_H
#if defined(EMSCRIPTEN)
# include <emscripten.h>
#endif
#ifdef __cplusplus
extern "C" {
#endif
#define XMP_VERSION "4.7.0"
#define XMP_VERCODE 0x040700
#define XMP_VER_MAJOR 4
#define XMP_VER_MINOR 7
#define XMP_VER_RELEASE 0
#if defined(_WIN32) && !defined(__CYGWIN__)
# if defined(LIBXMP_STATIC)
# define LIBXMP_EXPORT
# elif defined(BUILDING_DLL)
# define LIBXMP_EXPORT __declspec(dllexport)
# else
# define LIBXMP_EXPORT __declspec(dllimport)
# endif
#elif defined(__OS2__) && defined(__WATCOMC__)
# if defined(LIBXMP_STATIC)
# define LIBXMP_EXPORT
# elif defined(BUILDING_DLL)
# define LIBXMP_EXPORT __declspec(dllexport)
# else
# define LIBXMP_EXPORT
# endif
#elif (defined(__GNUC__) || defined(__clang__) || defined(__HP_cc)) && defined(XMP_SYM_VISIBILITY)
# if defined(LIBXMP_STATIC)
# define LIBXMP_EXPORT
# else
# define LIBXMP_EXPORT __attribute__((visibility("default")))
# endif
#elif defined(__SUNPRO_C) && defined(XMP_LDSCOPE_GLOBAL)
# if defined(LIBXMP_STATIC)
# define LIBXMP_EXPORT
# else
# define LIBXMP_EXPORT __global
# endif
#elif defined(EMSCRIPTEN)
# define LIBXMP_EXPORT EMSCRIPTEN_KEEPALIVE
# define LIBXMP_EXPORT_VAR
#else
# define LIBXMP_EXPORT
#endif
#if !defined(LIBXMP_EXPORT_VAR)
# define LIBXMP_EXPORT_VAR LIBXMP_EXPORT
#endif
#define XMP_NAME_SIZE 64 /* Size of module name and type */
#define XMP_KEY_OFF 0x81 /* Note number for key off event */
#define XMP_KEY_CUT 0x82 /* Note number for key cut event */
#define XMP_KEY_FADE 0x83 /* Note number for fade event */
/* mixer parameter macros */
/* sample format flags */
#define XMP_FORMAT_8BIT (1 << 0) /* Mix to 8-bit instead of 16 */
#define XMP_FORMAT_UNSIGNED (1 << 1) /* Mix to unsigned samples */
#define XMP_FORMAT_MONO (1 << 2) /* Mix to mono instead of stereo */
#define XMP_FORMAT_32BIT (1 << 3) /* Mix to 32-bit int instead of 16 */
/* player parameters */
#define XMP_PLAYER_AMP 0 /* Amplification factor */
#define XMP_PLAYER_MIX 1 /* Stereo mixing */
#define XMP_PLAYER_INTERP 2 /* Interpolation type */
#define XMP_PLAYER_DSP 3 /* DSP effect flags */
#define XMP_PLAYER_FLAGS 4 /* Player flags */
#define XMP_PLAYER_CFLAGS 5 /* Player flags for current module */
#define XMP_PLAYER_SMPCTL 6 /* Sample control flags */
#define XMP_PLAYER_VOLUME 7 /* Player module volume */
#define XMP_PLAYER_STATE 8 /* Internal player state (read only) */
#define XMP_PLAYER_SMIX_VOLUME 9 /* SMIX volume */
#define XMP_PLAYER_DEFPAN 10 /* Default pan setting */
#define XMP_PLAYER_MODE 11 /* Player personality */
#define XMP_PLAYER_MIXER_TYPE 12 /* Current mixer (read only) */
#define XMP_PLAYER_VOICES 13 /* Maximum number of mixer voices */
/* interpolation types */
#define XMP_INTERP_NEAREST 0 /* Nearest neighbor */
#define XMP_INTERP_LINEAR 1 /* Linear (default) */
#define XMP_INTERP_SPLINE 2 /* Cubic spline */
/* dsp effect types */
#define XMP_DSP_LOWPASS (1 << 0) /* Lowpass filter effect */
#define XMP_DSP_ALL (XMP_DSP_LOWPASS)
/* player state */
#define XMP_STATE_UNLOADED 0 /* Context created */
#define XMP_STATE_LOADED 1 /* Module loaded */
#define XMP_STATE_PLAYING 2 /* Module playing */
/* player flags */
#define XMP_FLAGS_VBLANK (1 << 0) /* Use vblank timing */
#define XMP_FLAGS_FX9BUG (1 << 1) /* Emulate FX9 bug */
#define XMP_FLAGS_FIXLOOP (1 << 2) /* Emulate sample loop bug */
#define XMP_FLAGS_A500 (1 << 3) /* Use Paula mixer in Amiga modules */
/* player modes */
#define XMP_MODE_AUTO 0 /* Autodetect mode (default) */
#define XMP_MODE_MOD 1 /* Play as a generic MOD player */
#define XMP_MODE_NOISETRACKER 2 /* Play using Noisetracker quirks */
#define XMP_MODE_PROTRACKER 3 /* Play using Protracker quirks */
#define XMP_MODE_S3M 4 /* Play as a generic S3M player */
#define XMP_MODE_ST3 5 /* Play using ST3 bug emulation */
#define XMP_MODE_ST3GUS 6 /* Play using ST3+GUS quirks */
#define XMP_MODE_XM 7 /* Play as a generic XM player */
#define XMP_MODE_FT2 8 /* Play using FT2 bug emulation */
#define XMP_MODE_IT 9 /* Play using IT quirks */
#define XMP_MODE_ITSMP 10 /* Play using IT sample mode quirks */
/* mixer types */
#define XMP_MIXER_STANDARD 0 /* Standard mixer */
#define XMP_MIXER_A500 1 /* Amiga 500 */
#define XMP_MIXER_A500F 2 /* Amiga 500 with led filter */
/* sample flags */
#define XMP_SMPCTL_SKIP (1 << 0) /* Don't load samples */
/* limits */
#define XMP_MAX_KEYS 121 /* Number of valid keys */
#define XMP_MAX_ENV_POINTS 32 /* Max number of envelope points */
#define XMP_MAX_MOD_LENGTH 256 /* Max number of patterns in module */
#define XMP_MAX_CHANNELS 64 /* Max number of channels in module */
#define XMP_MAX_SRATE 768000 /* max sampling rate (Hz) */
#define XMP_MIN_SRATE 4000 /* min sampling rate (Hz) */
#define XMP_MIN_BPM 20 /* min BPM */
/* frame rate = (50 * bpm / 125) Hz */
/* frame size = (sampling rate * channels * size) / frame rate */
#define XMP_MAX_FRAMESIZE (5 * XMP_MAX_SRATE * 2 / XMP_MIN_BPM)
/* error codes */
#define XMP_END 1
#define XMP_ERROR_INTERNAL 2 /* Internal error */
#define XMP_ERROR_FORMAT 3 /* Unsupported module format */
#define XMP_ERROR_LOAD 4 /* Error loading file */
#define XMP_ERROR_DEPACK 5 /* Error depacking file */
#define XMP_ERROR_SYSTEM 6 /* System error */
#define XMP_ERROR_INVALID 7 /* Invalid parameter */
#define XMP_ERROR_STATE 8 /* Invalid player state */
struct xmp_channel {
int pan; /* Channel pan (0x80 is center) */
int vol; /* Channel volume */
#define XMP_CHANNEL_SYNTH (1 << 0) /* Channel is synthesized */
#define XMP_CHANNEL_MUTE (1 << 1) /* Channel is muted */
#define XMP_CHANNEL_SPLIT (1 << 2) /* Split Amiga channel in bits 5-4 */
#define XMP_CHANNEL_SURROUND (1 << 4) /* Surround channel */
int flg; /* Channel flags */
};
struct xmp_pattern {
int rows; /* Number of rows */
int index[1]; /* Track index */
};
struct xmp_event {
unsigned char note; /* Note number (0 means no note) */
unsigned char ins; /* Patch number */
unsigned char vol; /* Volume (0 to basevol) */
unsigned char fxt; /* Effect type */
unsigned char fxp; /* Effect parameter */
unsigned char f2t; /* Secondary effect type */
unsigned char f2p; /* Secondary effect parameter */
unsigned char _flag; /* Internal (reserved) flags */
};
struct xmp_track {
int rows; /* Number of rows */
struct xmp_event event[1]; /* Event data */
};
struct xmp_envelope {
#define XMP_ENVELOPE_ON (1 << 0) /* Envelope is enabled */
#define XMP_ENVELOPE_SUS (1 << 1) /* Envelope has sustain point */
#define XMP_ENVELOPE_LOOP (1 << 2) /* Envelope has loop */
#define XMP_ENVELOPE_FLT (1 << 3) /* Envelope is used for filter */
#define XMP_ENVELOPE_SLOOP (1 << 4) /* Envelope has sustain loop */
#define XMP_ENVELOPE_CARRY (1 << 5) /* Don't reset envelope position */
int flg; /* Flags */
int npt; /* Number of envelope points */
int scl; /* Envelope scaling */
int sus; /* Sustain start point */
int sue; /* Sustain end point */
int lps; /* Loop start point */
int lpe; /* Loop end point */
short data[XMP_MAX_ENV_POINTS * 2];
};
struct xmp_subinstrument {
int vol; /* Default volume */
int gvl; /* Global volume */
#define XMP_INST_NO_DEFAULT_PAN -1
int pan; /* Pan */
int xpo; /* Transpose */
int fin; /* Finetune */
int vwf; /* Vibrato waveform */
int vde; /* Vibrato depth */
int vra; /* Vibrato rate */
int vsw; /* Vibrato sweep */
int rvv; /* Random volume/pan variation (IT) */
int sid; /* Sample number */
#define XMP_INST_NNA_CUT 0x00
#define XMP_INST_NNA_CONT 0x01
#define XMP_INST_NNA_OFF 0x02
#define XMP_INST_NNA_FADE 0x03
int nna; /* New note action */
#define XMP_INST_DCT_OFF 0x00
#define XMP_INST_DCT_NOTE 0x01
#define XMP_INST_DCT_SMP 0x02
#define XMP_INST_DCT_INST 0x03
int dct; /* Duplicate check type */
#define XMP_INST_DCA_CUT XMP_INST_NNA_CUT
#define XMP_INST_DCA_OFF XMP_INST_NNA_OFF
#define XMP_INST_DCA_FADE XMP_INST_NNA_FADE
int dca; /* Duplicate check action */
int ifc; /* Initial filter cutoff */
int ifr; /* Initial filter resonance */
};
struct xmp_instrument {
char name[32]; /* Instrument name */
int vol; /* Instrument volume */
int nsm; /* Number of samples */
int rls; /* Release (fadeout) */
struct xmp_envelope aei; /* Amplitude envelope info */
struct xmp_envelope pei; /* Pan envelope info */
struct xmp_envelope fei; /* Frequency envelope info */
struct {
unsigned char ins; /* Instrument number for each key */
signed char xpo; /* Instrument transpose for each key */
} map[XMP_MAX_KEYS];
struct xmp_subinstrument *sub;
void *extra; /* Extra fields */
};
struct xmp_sample {
char name[32]; /* Sample name */
int len; /* Sample length */
int lps; /* Loop start */
int lpe; /* Loop end */
#define XMP_SAMPLE_16BIT (1 << 0) /* 16bit sample */
#define XMP_SAMPLE_LOOP (1 << 1) /* Sample is looped */
#define XMP_SAMPLE_LOOP_BIDIR (1 << 2) /* Bidirectional sample loop */
#define XMP_SAMPLE_LOOP_REVERSE (1 << 3) /* Backwards sample loop */
#define XMP_SAMPLE_LOOP_FULL (1 << 4) /* Play full sample before looping */
#define XMP_SAMPLE_SLOOP (1 << 5) /* Sample has sustain loop */
#define XMP_SAMPLE_SLOOP_BIDIR (1 << 6) /* Bidirectional sustain loop */
#define XMP_SAMPLE_STEREO (1 << 7) /* Interlaced stereo sample */
#define XMP_SAMPLE_SYNTH (1 << 15) /* Data contains synth patch */
int flg; /* Flags */
unsigned char *data; /* Sample data */
};
struct xmp_sequence {
int entry_point;
int duration;
};
struct xmp_module {
char name[XMP_NAME_SIZE]; /* Module title */
char type[XMP_NAME_SIZE]; /* Module format */
int pat; /* Number of patterns */
int trk; /* Number of tracks */
int chn; /* Tracks per pattern */
int ins; /* Number of instruments */
int smp; /* Number of samples */
int spd; /* Initial speed */
int bpm; /* Initial BPM */
int len; /* Module length in patterns */
int rst; /* Restart position */
int gvl; /* Global volume */
struct xmp_pattern **xxp; /* Patterns */
struct xmp_track **xxt; /* Tracks */
struct xmp_instrument *xxi; /* Instruments */
struct xmp_sample *xxs; /* Samples */
struct xmp_channel xxc[XMP_MAX_CHANNELS]; /* Channel info */
#define XMP_MARK_SKIP 0xfe /* S3M/IT-only: skip position */
#define XMP_MARK_END 0xff /* S3M/IT-only: end position */
unsigned char xxo[XMP_MAX_MOD_LENGTH]; /* Orders */
};
struct xmp_test_info {
char name[XMP_NAME_SIZE]; /* Module title */
char type[XMP_NAME_SIZE]; /* Module format */
};
struct xmp_module_info {
unsigned char md5[16]; /* MD5 message digest */
int vol_base; /* Volume scale */
struct xmp_module *mod; /* Pointer to module data */
char *comment; /* Comment text, if any */
int num_sequences; /* Number of valid sequences */
struct xmp_sequence *seq_data; /* Pointer to sequence data */
};
struct xmp_channel_info {
unsigned int period; /* Sample period (* 4096) */
unsigned int position; /* Sample position */
short pitchbend; /* Linear bend from base note*/
unsigned char note; /* Current base note number */
unsigned char instrument; /* Current instrument number */
unsigned char sample; /* Current sample number */
unsigned char volume; /* Current volume */
unsigned char pan; /* Current stereo pan */
unsigned char reserved; /* Reserved */
struct xmp_event event; /* Current track event */
};
struct xmp_frame_info { /* Current frame information */
int pos; /* Current position */
int pattern; /* Current pattern */
int row; /* Current row in pattern */
int num_rows; /* Number of rows in current pattern */
int frame; /* Current frame */
int speed; /* Current replay speed */
int bpm; /* Current bpm */
int time; /* Current module time in ms */
int total_time; /* Estimated replay time in ms*/
int frame_time; /* Frame replay time in us */
void *buffer; /* Pointer to sound buffer */
int buffer_size; /* Used buffer size */
int total_size; /* Total buffer size */
int volume; /* Current master volume */
int loop_count; /* Loop counter */
int virt_channels; /* Number of virtual channels */
int virt_used; /* Used virtual channels */
int sequence; /* Current sequence */
struct xmp_channel_info channel_info[XMP_MAX_CHANNELS]; /* Current channel information */
};
struct xmp_callbacks {
unsigned long (*read_func)(void *dest, unsigned long len,
unsigned long nmemb, void *priv);
int (*seek_func)(void *priv, long offset, int whence);
long (*tell_func)(void *priv);
int (*close_func)(void *priv);
};
typedef char *xmp_context;
LIBXMP_EXPORT_VAR extern const char *xmp_version;
LIBXMP_EXPORT_VAR extern const unsigned int xmp_vercode;
LIBXMP_EXPORT int xmp_syserrno (void);
LIBXMP_EXPORT xmp_context xmp_create_context (void);
LIBXMP_EXPORT void xmp_free_context (xmp_context);
LIBXMP_EXPORT int xmp_load_module (xmp_context, const char *);
LIBXMP_EXPORT int xmp_load_module_from_memory (xmp_context, const void *, long);
LIBXMP_EXPORT int xmp_load_module_from_file (xmp_context, void *, long);
LIBXMP_EXPORT int xmp_load_module_from_callbacks (xmp_context, void *, struct xmp_callbacks);
LIBXMP_EXPORT int xmp_test_module (const char *, struct xmp_test_info *);
LIBXMP_EXPORT int xmp_test_module_from_memory (const void *, long, struct xmp_test_info *);
LIBXMP_EXPORT int xmp_test_module_from_file (void *, struct xmp_test_info *);
LIBXMP_EXPORT int xmp_test_module_from_callbacks (void *, struct xmp_callbacks, struct xmp_test_info *);
LIBXMP_EXPORT void xmp_scan_module (xmp_context);
LIBXMP_EXPORT void xmp_release_module (xmp_context);
LIBXMP_EXPORT int xmp_start_player (xmp_context, int, int);
LIBXMP_EXPORT int xmp_play_frame (xmp_context);
LIBXMP_EXPORT int xmp_play_buffer (xmp_context, void *, int, int);
LIBXMP_EXPORT void xmp_get_frame_info (xmp_context, struct xmp_frame_info *);
LIBXMP_EXPORT void xmp_end_player (xmp_context);
LIBXMP_EXPORT void xmp_inject_event (xmp_context, int, struct xmp_event *);
LIBXMP_EXPORT void xmp_get_module_info (xmp_context, struct xmp_module_info *);
LIBXMP_EXPORT const char *const *xmp_get_format_list (void);
LIBXMP_EXPORT int xmp_next_position (xmp_context);
LIBXMP_EXPORT int xmp_prev_position (xmp_context);
LIBXMP_EXPORT int xmp_set_position (xmp_context, int);
LIBXMP_EXPORT int xmp_set_row (xmp_context, int);
LIBXMP_EXPORT int xmp_set_tempo_factor(xmp_context, double);
LIBXMP_EXPORT int xmp_set_tempo_factor_relative(xmp_context, double);
LIBXMP_EXPORT double xmp_get_tempo_factor(xmp_context);
LIBXMP_EXPORT double xmp_get_tempo_factor_relative(xmp_context);
LIBXMP_EXPORT void xmp_stop_module (xmp_context);
LIBXMP_EXPORT void xmp_restart_module (xmp_context);
LIBXMP_EXPORT int xmp_seek_time (xmp_context, int);
LIBXMP_EXPORT int xmp_seek_time_frame (xmp_context, int);
LIBXMP_EXPORT int xmp_channel_mute (xmp_context, int, int);
LIBXMP_EXPORT int xmp_channel_vol (xmp_context, int, int);
LIBXMP_EXPORT int xmp_set_player (xmp_context, int, int);
LIBXMP_EXPORT int xmp_get_player (xmp_context, int);
LIBXMP_EXPORT int xmp_set_instrument_path (xmp_context, const char *);
/* External sample mixer API */
LIBXMP_EXPORT int xmp_start_smix (xmp_context, int, int);
LIBXMP_EXPORT void xmp_end_smix (xmp_context);
LIBXMP_EXPORT int xmp_smix_play_instrument(xmp_context, int, int, int, int);
LIBXMP_EXPORT int xmp_smix_play_sample (xmp_context, int, int, int, int);
LIBXMP_EXPORT int xmp_smix_channel_pan (xmp_context, int, int);
LIBXMP_EXPORT int xmp_smix_load_sample (xmp_context, int, const char *);
LIBXMP_EXPORT int xmp_smix_release_sample (xmp_context, int);
#ifdef __cplusplus
}
#endif
#endif /* XMP_H */
libs/SDL3_mixer/VisualC/external/optional/x64/LICENSE.gme.txt
Vendored
+504
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@@ -0,0 +1,504 @@
GNU LESSER GENERAL PUBLIC LICENSE
Version 2.1, February 1999
Copyright (C) 1991, 1999 Free Software Foundation, Inc.
51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
Everyone is permitted to copy and distribute verbatim copies
of this license document, but changing it is not allowed.
[This is the first released version of the Lesser GPL. It also counts
as the successor of the GNU Library Public License, version 2, hence
the version number 2.1.]
Preamble
The licenses for most software are designed to take away your
freedom to share and change it. By contrast, the GNU General Public
Licenses are intended to guarantee your freedom to share and change
free software--to make sure the software is free for all its users.
This license, the Lesser General Public License, applies to some
specially designated software packages--typically libraries--of the
Free Software Foundation and other authors who decide to use it. You
can use it too, but we suggest you first think carefully about whether
this license or the ordinary General Public License is the better
strategy to use in any particular case, based on the explanations below.
When we speak of free software, we are referring to freedom of use,
not price. Our General Public Licenses are designed to make sure that
you have the freedom to distribute copies of free software (and charge
for this service if you wish); that you receive source code or can get
it if you want it; that you can change the software and use pieces of
it in new free programs; and that you are informed that you can do
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Copyright 2001-2011 Xiph.Org, Skype Limited, Octasic,
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Copyright (c) 1998 - 2025 David Bryant
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Extended Module Player
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GNU LESSER GENERAL PUBLIC LICENSE
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Copyright (c) 2002-2020 Xiph.org Foundation
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Copyright 2001-2011 Xiph.Org, Skype Limited, Octasic,
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Copyright (c) 1998 - 2025 David Bryant
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Extended Module Player
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libs/SDL3_mixer/VisualC/external/optional/x86/libxmp.dll
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