first

MacOSX/codecs/include/FLAC/callback.h

@@ -0,0 +1,185 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2004-2009  Josh Coalson
+ * Copyright (C) 2011-2013  Xiph.Org Foundation
+ *
+ * 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.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * 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 FOUNDATION 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.
+ */
+
+#ifndef FLAC__CALLBACK_H
+#define FLAC__CALLBACK_H
+
+#include "ordinals.h"
+#include <stdlib.h> /* for size_t */
+
+/** \file include/FLAC/callback.h
+ *
+ *  \brief
+ *  This module defines the structures for describing I/O callbacks
+ *  to the other FLAC interfaces.
+ *
+ *  See the detailed documentation for callbacks in the
+ *  \link flac_callbacks callbacks \endlink module.
+ */
+
+/** \defgroup flac_callbacks FLAC/callback.h: I/O callback structures
+ *  \ingroup flac
+ *
+ *  \brief
+ *  This module defines the structures for describing I/O callbacks
+ *  to the other FLAC interfaces.
+ *
+ *  The purpose of the I/O callback functions is to create a common way
+ *  for the metadata interfaces to handle I/O.
+ *
+ *  Originally the metadata interfaces required filenames as the way of
+ *  specifying FLAC files to operate on.  This is problematic in some
+ *  environments so there is an additional option to specify a set of
+ *  callbacks for doing I/O on the FLAC file, instead of the filename.
+ *
+ *  In addition to the callbacks, a FLAC__IOHandle type is defined as an
+ *  opaque structure for a data source.
+ *
+ *  The callback function prototypes are similar (but not identical) to the
+ *  stdio functions fread, fwrite, fseek, ftell, feof, and fclose.  If you use
+ *  stdio streams to implement the callbacks, you can pass fread, fwrite, and
+ *  fclose anywhere a FLAC__IOCallback_Read, FLAC__IOCallback_Write, or
+ *  FLAC__IOCallback_Close is required, and a FILE* anywhere a FLAC__IOHandle
+ *  is required.  \warning You generally CANNOT directly use fseek or ftell
+ *  for FLAC__IOCallback_Seek or FLAC__IOCallback_Tell since on most systems
+ *  these use 32-bit offsets and FLAC requires 64-bit offsets to deal with
+ *  large files.  You will have to find an equivalent function (e.g. ftello),
+ *  or write a wrapper.  The same is true for feof() since this is usually
+ *  implemented as a macro, not as a function whose address can be taken.
+ *
+ * \{
+ */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** This is the opaque handle type used by the callbacks.  Typically
+ *  this is a \c FILE* or address of a file descriptor.
+ */
+typedef void* FLAC__IOHandle;
+
+/** Signature for the read callback.
+ *  The signature and semantics match POSIX fread() implementations
+ *  and can generally be used interchangeably.
+ *
+ * \param  ptr      The address of the read buffer.
+ * \param  size     The size of the records to be read.
+ * \param  nmemb    The number of records to be read.
+ * \param  handle   The handle to the data source.
+ * \retval size_t
+ *    The number of records read.
+ */
+typedef size_t (*FLAC__IOCallback_Read) (void *ptr, size_t size, size_t nmemb, FLAC__IOHandle handle);
+
+/** Signature for the write callback.
+ *  The signature and semantics match POSIX fwrite() implementations
+ *  and can generally be used interchangeably.
+ *
+ * \param  ptr      The address of the write buffer.
+ * \param  size     The size of the records to be written.
+ * \param  nmemb    The number of records to be written.
+ * \param  handle   The handle to the data source.
+ * \retval size_t
+ *    The number of records written.
+ */
+typedef size_t (*FLAC__IOCallback_Write) (const void *ptr, size_t size, size_t nmemb, FLAC__IOHandle handle);
+
+/** Signature for the seek callback.
+ *  The signature and semantics mostly match POSIX fseek() WITH ONE IMPORTANT
+ *  EXCEPTION: the offset is a 64-bit type whereas fseek() is generally 'long'
+ *  and 32-bits wide.
+ *
+ * \param  handle   The handle to the data source.
+ * \param  offset   The new position, relative to \a whence
+ * \param  whence   \c SEEK_SET, \c SEEK_CUR, or \c SEEK_END
+ * \retval int
+ *    \c 0 on success, \c -1 on error.
+ */
+typedef int (*FLAC__IOCallback_Seek) (FLAC__IOHandle handle, FLAC__int64 offset, int whence);
+
+/** Signature for the tell callback.
+ *  The signature and semantics mostly match POSIX ftell() WITH ONE IMPORTANT
+ *  EXCEPTION: the offset is a 64-bit type whereas ftell() is generally 'long'
+ *  and 32-bits wide.
+ *
+ * \param  handle   The handle to the data source.
+ * \retval FLAC__int64
+ *    The current position on success, \c -1 on error.
+ */
+typedef FLAC__int64 (*FLAC__IOCallback_Tell) (FLAC__IOHandle handle);
+
+/** Signature for the EOF callback.
+ *  The signature and semantics mostly match POSIX feof() but WATCHOUT:
+ *  on many systems, feof() is a macro, so in this case a wrapper function
+ *  must be provided instead.
+ *
+ * \param  handle   The handle to the data source.
+ * \retval int
+ *    \c 0 if not at end of file, nonzero if at end of file.
+ */
+typedef int (*FLAC__IOCallback_Eof) (FLAC__IOHandle handle);
+
+/** Signature for the close callback.
+ *  The signature and semantics match POSIX fclose() implementations
+ *  and can generally be used interchangeably.
+ *
+ * \param  handle   The handle to the data source.
+ * \retval int
+ *    \c 0 on success, \c EOF on error.
+ */
+typedef int (*FLAC__IOCallback_Close) (FLAC__IOHandle handle);
+
+/** A structure for holding a set of callbacks.
+ *  Each FLAC interface that requires a FLAC__IOCallbacks structure will
+ *  describe which of the callbacks are required.  The ones that are not
+ *  required may be set to NULL.
+ *
+ *  If the seek requirement for an interface is optional, you can signify that
+ *  a data sorce is not seekable by setting the \a seek field to \c NULL.
+ */
+typedef struct {
+	FLAC__IOCallback_Read read;
+	FLAC__IOCallback_Write write;
+	FLAC__IOCallback_Seek seek;
+	FLAC__IOCallback_Tell tell;
+	FLAC__IOCallback_Eof eof;
+	FLAC__IOCallback_Close close;
+} FLAC__IOCallbacks;
+
+/* \} */
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

MacOSX/codecs/include/FLAC/export.h

@@ -0,0 +1,97 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2013  Xiph.Org Foundation
+ *
+ * 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.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * 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 FOUNDATION 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.
+ */
+
+#ifndef FLAC__EXPORT_H
+#define FLAC__EXPORT_H
+
+/** \file include/FLAC/export.h
+ *
+ *  \brief
+ *  This module contains #defines and symbols for exporting function
+ *  calls, and providing version information and compiled-in features.
+ *
+ *  See the \link flac_export export \endlink module.
+ */
+
+/** \defgroup flac_export FLAC/export.h: export symbols
+ *  \ingroup flac
+ *
+ *  \brief
+ *  This module contains #defines and symbols for exporting function
+ *  calls, and providing version information and compiled-in features.
+ *
+ *  If you are compiling with MSVC and will link to the static library
+ *  (libFLAC.lib) you should define FLAC__NO_DLL in your project to
+ *  make sure the symbols are exported properly.
+ *
+ * \{
+ */
+
+#if defined(FLAC__NO_DLL)
+#define FLAC_API
+
+#elif defined(_WIN32) /*defined(_MSC_VER)*/
+#ifdef FLAC_API_EXPORTS
+#define FLAC_API __declspec(dllexport)
+#else
+#define FLAC_API __declspec(dllimport)
+#endif
+
+#elif defined(FLAC__USE_VISIBILITY_ATTR)
+#define FLAC_API __attribute__ ((visibility ("default")))
+
+#else
+#define FLAC_API
+
+#endif
+
+/** These #defines will mirror the libtool-based library version number, see
+ * http://www.gnu.org/software/libtool/manual/libtool.html#Libtool-versioning
+ */
+#define FLAC_API_VERSION_CURRENT 11
+#define FLAC_API_VERSION_REVISION 0 /**< see above */
+#define FLAC_API_VERSION_AGE 3 /**< see above */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** \c 1 if the library has been compiled with support for Ogg FLAC, else \c 0. */
+extern FLAC_API int FLAC_API_SUPPORTS_OGG_FLAC;
+
+#ifdef __cplusplus
+}
+#endif
+
+/* \} */
+
+#endif

MacOSX/codecs/include/FLAC/ordinals.h

@@ -0,0 +1,86 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2013  Xiph.Org Foundation
+ *
+ * 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.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * 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 FOUNDATION 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.
+ */
+
+#ifndef FLAC__ORDINALS_H
+#define FLAC__ORDINALS_H
+
+#if defined(_MSC_VER) && _MSC_VER < 1600
+
+/* Microsoft Visual Studio earlier than the 2010 version did not provide
+ * the 1999 ISO C Standard header file <stdint.h>.
+ */
+
+typedef __int8 FLAC__int8;
+typedef unsigned __int8 FLAC__uint8;
+
+typedef __int16 FLAC__int16;
+typedef __int32 FLAC__int32;
+typedef __int64 FLAC__int64;
+typedef unsigned __int16 FLAC__uint16;
+typedef unsigned __int32 FLAC__uint32;
+typedef unsigned __int64 FLAC__uint64;
+
+#else
+
+/* For MSVC 2010 and everything else which provides <stdint.h>. */
+
+#include <stdint.h>
+
+typedef int8_t FLAC__int8;
+typedef uint8_t FLAC__uint8;
+
+typedef int16_t FLAC__int16;
+typedef int32_t FLAC__int32;
+typedef int64_t FLAC__int64;
+typedef uint16_t FLAC__uint16;
+typedef uint32_t FLAC__uint32;
+typedef uint64_t FLAC__uint64;
+
+#endif
+
+typedef int FLAC__bool;
+
+typedef FLAC__uint8 FLAC__byte;
+
+
+#ifdef true
+#undef true
+#endif
+#ifdef false
+#undef false
+#endif
+#ifndef __cplusplus
+#define true 1
+#define false 0
+#endif
+
+#endif

MacOSX/codecs/include/flac_config.txt

@@ -0,0 +1,3 @@
+FLAC v1.3.0 release + several fixes from the flac git repo at xiph.org.
+Decoder-only functionality, which is what we need: the encoder stuff is
+left out of the build.

MacOSX/codecs/include/mikmod.h

@@ -0,0 +1,880 @@
+/*  MikMod sound library
+    (c) 1998-2014 Miodrag Vallat and others - see the AUTHORS file
+    for complete list.
+
+    This library is free software; you can redistribute it and/or modify
+    it under the terms of the GNU Library General Public License as
+    published by the Free Software Foundation; either version 2 of
+    the License, or (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU Library General Public License for more details.
+
+    You should have received a copy of the GNU Library General Public
+    License along with this library; if not, write to the Free Software
+    Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
+    02111-1307, USA.
+*/
+
+/*==============================================================================
+
+  MikMod sound library include file
+
+  ==============================================================================*/
+
+#ifndef _MIKMOD_H_
+#define _MIKMOD_H_
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/*
+ * ========== Compiler magic for shared libraries
+ *
+ * ========== NOTE TO WINDOWS DEVELOPERS:
+ * If you are compiling for Windows and will link to the static library
+ * (libmikmod.a with MinGW, or mikmod_static.lib with MSVC, Watcom, ..),
+ * you must define MIKMOD_STATIC in your project.  Otherwise, dllimport
+ * will be assumed.
+ */
+#if defined(_WIN32) || defined(__CYGWIN__)
+# if defined(MIKMOD_BUILD) && defined(DLL_EXPORT)       /* building libmikmod as a dll for windows */
+#   define MIKMODAPI __declspec(dllexport)
+# elif defined(MIKMOD_BUILD) || defined(MIKMOD_STATIC)  /* building or using static libmikmod for windows */
+#   define MIKMODAPI
+# else
+#   define MIKMODAPI __declspec(dllimport)                      /* using libmikmod dll for windows */
+# endif
+#elif defined(__OS2__) && defined(__WATCOMC__)
+# if defined(MIKMOD_BUILD) && defined(__SW_BD)          /* building libmikmod as a dll for os/2 */
+#   define MIKMODAPI __declspec(dllexport)
+# else
+#   define MIKMODAPI                                    /* using dll or static libmikmod for os/2 */
+# endif
+/* SYM_VISIBILITY should be defined if both the compiler
+ * and the target support the visibility attributes. the
+ * configury does that automatically. for the standalone
+ * makefiles, etc, the developer should add the required
+ * flags, i.e.:  -DSYM_VISIBILITY -fvisibility=hidden  */
+#elif defined(MIKMOD_BUILD) && defined(SYM_VISIBILITY)
+#   define MIKMODAPI __attribute__((visibility("default")))
+#else
+#   define MIKMODAPI
+#endif
+
+/*
+ *  ========== Library version
+ */
+
+#define LIBMIKMOD_VERSION_MAJOR 3L
+#define LIBMIKMOD_VERSION_MINOR 3L
+#define LIBMIKMOD_REVISION     13L
+
+#define LIBMIKMOD_VERSION \
+    ((LIBMIKMOD_VERSION_MAJOR<<16)| \
+     (LIBMIKMOD_VERSION_MINOR<< 8)| \
+     (LIBMIKMOD_REVISION))
+
+MIKMODAPI extern long MikMod_GetVersion(void);
+
+/*
+ *  ========== Dependency platform headers
+ */
+
+#if defined(_WIN32)||defined(__CYGWIN__)
+#ifndef WIN32_LEAN_AND_MEAN
+#define WIN32_LEAN_AND_MEAN
+#endif
+#include <windows.h>
+#include <io.h>
+#include <mmsystem.h>
+#define _MIKMOD_WIN32
+#endif
+
+#if defined(__DJGPP__) || defined(MSDOS) || defined(__MSDOS__) || defined(__DOS__)
+#define _MIKMOD_DOS
+#endif
+
+#if defined(__OS2__) || defined(__EMX__)
+#define INCL_DOSSEMAPHORES
+#include <os2.h>
+#include <io.h>
+#define _MIKMOD_OS2
+#endif
+
+#if defined(__MORPHOS__) || defined(__AROS__) || defined(_AMIGA) || defined(__AMIGA__) || defined(__amigaos__) || defined(AMIGAOS)
+#include <exec/types.h>
+#define _MIKMOD_AMIGA
+#endif
+
+/*
+ *  ========== Platform independent-type definitions
+ * (pain when it comes to cross-platform maintenance..)
+ */
+
+#if !(defined(_MIKMOD_OS2) || defined(_MIKMOD_WIN32))
+typedef char               CHAR;
+#endif
+
+/* BOOL:  0=false, <>0 true -- 16 bits on Amiga, int-wide on others. */
+#if !(defined(_MIKMOD_OS2) || defined(_MIKMOD_WIN32) || defined(_MIKMOD_AMIGA))
+typedef int                BOOL;
+#endif
+
+/* 1 byte, signed and unsigned: */
+typedef signed char        SBYTE;
+#ifndef _MIKMOD_AMIGA
+typedef unsigned char      UBYTE;
+#endif
+
+/* 2 bytes, signed and unsigned: */
+typedef signed short int   SWORD;
+#if !defined(_MIKMOD_AMIGA)
+typedef unsigned short int UWORD;
+#endif
+
+/* 4 bytes, signed and unsigned: */
+#if defined(_LP64) || defined(__LP64__) || defined(__arch64__) || defined(__alpha) || defined(__x86_64) || defined(__powerpc64__)
+        /* 64 bit architectures: */
+typedef signed int         SLONG;
+#if !(defined(_WIN32) || defined(_MIKMOD_AMIGA))
+typedef unsigned int       ULONG;
+#endif
+
+#else  /* 32 bit architectures: */
+typedef signed long int    SLONG;
+#if !(defined(_MIKMOD_OS2) || defined(_MIKMOD_WIN32) || defined(_MIKMOD_AMIGA))
+typedef unsigned long int  ULONG;
+#endif
+#endif
+
+/* make sure types are of correct sizes: */
+typedef int __mikmod_typetest [
+   (
+        (sizeof(SBYTE)==1) && (sizeof(UBYTE)==1)
+     && (sizeof(SWORD)==2) && (sizeof(UWORD)==2)
+     && (sizeof(SLONG)==4) && (sizeof(ULONG)==4)
+#ifndef _MIKMOD_AMIGA
+     && (sizeof(BOOL) == sizeof(int))
+#endif
+     && (sizeof(CHAR) == sizeof(char))
+   ) * 2 - 1 ];
+
+/*
+ *  ========== Error codes
+ */
+
+enum {
+    MMERR_OPENING_FILE = 1,
+    MMERR_OUT_OF_MEMORY,
+    MMERR_DYNAMIC_LINKING,
+
+    MMERR_SAMPLE_TOO_BIG,
+    MMERR_OUT_OF_HANDLES,
+    MMERR_UNKNOWN_WAVE_TYPE,
+
+    MMERR_LOADING_PATTERN,
+    MMERR_LOADING_TRACK,
+    MMERR_LOADING_HEADER,
+    MMERR_LOADING_SAMPLEINFO,
+    MMERR_NOT_A_MODULE,
+    MMERR_NOT_A_STREAM,
+    MMERR_MED_SYNTHSAMPLES,
+    MMERR_ITPACK_INVALID_DATA,
+
+    MMERR_DETECTING_DEVICE,
+    MMERR_INVALID_DEVICE,
+    MMERR_INITIALIZING_MIXER,
+    MMERR_OPENING_AUDIO,
+    MMERR_8BIT_ONLY,
+    MMERR_16BIT_ONLY,
+    MMERR_STEREO_ONLY,
+    MMERR_ULAW,
+    MMERR_NON_BLOCK,
+
+    MMERR_AF_AUDIO_PORT,
+
+    MMERR_AIX_CONFIG_INIT,
+    MMERR_AIX_CONFIG_CONTROL,
+    MMERR_AIX_CONFIG_START,
+
+    MMERR_GUS_SETTINGS,
+    MMERR_GUS_RESET,
+    MMERR_GUS_TIMER,
+
+    MMERR_HP_SETSAMPLESIZE,
+    MMERR_HP_SETSPEED,
+    MMERR_HP_CHANNELS,
+    MMERR_HP_AUDIO_OUTPUT,
+    MMERR_HP_AUDIO_DESC,
+    MMERR_HP_BUFFERSIZE,
+
+    MMERR_OSS_SETFRAGMENT,
+    MMERR_OSS_SETSAMPLESIZE,
+    MMERR_OSS_SETSTEREO,
+    MMERR_OSS_SETSPEED,
+
+    MMERR_SGI_SPEED,
+    MMERR_SGI_16BIT,
+    MMERR_SGI_8BIT,
+    MMERR_SGI_STEREO,
+    MMERR_SGI_MONO,
+
+    MMERR_SUN_INIT,
+
+    MMERR_OS2_MIXSETUP,
+    MMERR_OS2_SEMAPHORE,
+    MMERR_OS2_TIMER,
+    MMERR_OS2_THREAD,
+
+    MMERR_DS_PRIORITY,
+    MMERR_DS_BUFFER,
+    MMERR_DS_FORMAT,
+    MMERR_DS_NOTIFY,
+    MMERR_DS_EVENT,
+    MMERR_DS_THREAD,
+    MMERR_DS_UPDATE,
+
+    MMERR_WINMM_HANDLE,
+    MMERR_WINMM_ALLOCATED,
+    MMERR_WINMM_DEVICEID,
+    MMERR_WINMM_FORMAT,
+    MMERR_WINMM_UNKNOWN,
+
+    MMERR_MAC_SPEED,
+    MMERR_MAC_START,
+
+    MMERR_OSX_UNKNOWN_DEVICE, /* obsolete */
+    MMERR_OSX_BAD_PROPERTY,   /* obsolete */
+    MMERR_OSX_UNSUPPORTED_FORMAT,
+    MMERR_OSX_SET_STEREO,     /* obsolete */
+    MMERR_OSX_BUFFER_ALLOC,   /* obsolete */
+    MMERR_OSX_ADD_IO_PROC,    /* obsolete */
+    MMERR_OSX_DEVICE_START,
+    MMERR_OSX_PTHREAD,        /* obsolete */
+
+    MMERR_DOSWSS_STARTDMA,
+    MMERR_DOSSB_STARTDMA,
+
+    MMERR_NO_FLOAT32,/* should actually be after MMERR_ULAW or something */
+
+    MMERR_OPENAL_CREATECTX,
+    MMERR_OPENAL_CTXCURRENT,
+    MMERR_OPENAL_GENBUFFERS,
+    MMERR_OPENAL_GENSOURCES,
+    MMERR_OPENAL_SOURCE,
+    MMERR_OPENAL_QUEUEBUFFERS,
+    MMERR_OPENAL_UNQUEUEBUFFERS,
+    MMERR_OPENAL_BUFFERDATA,
+    MMERR_OPENAL_GETSOURCE,
+    MMERR_OPENAL_SOURCEPLAY,
+    MMERR_OPENAL_SOURCESTOP,
+
+    MMERR_ALSA_NOCONFIG,
+    MMERR_ALSA_SETPARAMS,
+    MMERR_ALSA_SETFORMAT,
+    MMERR_ALSA_SETRATE,
+    MMERR_ALSA_SETCHANNELS,
+    MMERR_ALSA_BUFFERSIZE,
+    MMERR_ALSA_PCM_START,
+    MMERR_ALSA_PCM_WRITE,
+    MMERR_ALSA_PCM_RECOVER,
+
+    MMERR_SNDIO_SETPARAMS,
+    MMERR_SNDIO_BADPARAMS,
+
+    MMERR_MAX
+};
+
+/*
+ *  ========== Error handling
+ */
+
+typedef void (MikMod_handler)(void);
+typedef MikMod_handler *MikMod_handler_t;
+
+MIKMODAPI extern int  MikMod_errno;
+MIKMODAPI extern BOOL MikMod_critical;
+MIKMODAPI extern const char *MikMod_strerror(int);
+
+MIKMODAPI extern MikMod_handler_t MikMod_RegisterErrorHandler(MikMod_handler_t);
+
+/*
+ *  ========== Library initialization and core functions
+ */
+
+struct MDRIVER;
+
+MIKMODAPI extern void   MikMod_RegisterAllDrivers(void);
+
+MIKMODAPI extern CHAR*  MikMod_InfoDriver(void);
+MIKMODAPI extern void   MikMod_RegisterDriver(struct MDRIVER*);
+MIKMODAPI extern int    MikMod_DriverFromAlias(const CHAR*);
+MIKMODAPI extern struct MDRIVER *MikMod_DriverByOrdinal(int);
+
+MIKMODAPI extern int    MikMod_Init(const CHAR*);
+MIKMODAPI extern void   MikMod_Exit(void);
+MIKMODAPI extern int    MikMod_Reset(const CHAR*);
+MIKMODAPI extern int    MikMod_SetNumVoices(int,int);
+MIKMODAPI extern BOOL   MikMod_Active(void);
+MIKMODAPI extern int    MikMod_EnableOutput(void);
+MIKMODAPI extern void   MikMod_DisableOutput(void);
+MIKMODAPI extern void   MikMod_Update(void);
+
+MIKMODAPI extern BOOL   MikMod_InitThreads(void);
+MIKMODAPI extern void   MikMod_Lock(void);
+MIKMODAPI extern void   MikMod_Unlock(void);
+
+MIKMODAPI extern void*  MikMod_malloc(size_t);
+MIKMODAPI extern void*  MikMod_calloc(size_t,size_t);
+MIKMODAPI extern void*  MikMod_realloc(void*,size_t);
+MIKMODAPI extern CHAR*  MikMod_strdup(const CHAR*);
+MIKMODAPI extern void   MikMod_free(void*);  /* frees if ptr != NULL */
+
+/*
+ *  ========== Reader, Writer
+ */
+
+typedef struct MREADER {
+    int  (*Seek)(struct MREADER*,long,int);
+    long (*Tell)(struct MREADER*);
+    BOOL (*Read)(struct MREADER*,void*,size_t);
+    int  (*Get)(struct MREADER*);
+    BOOL (*Eof)(struct MREADER*);
+    long iobase;
+    long prev_iobase;
+} MREADER;
+
+typedef struct MWRITER {
+    int  (*Seek)(struct MWRITER*, long, int);
+    long (*Tell)(struct MWRITER*);
+    BOOL (*Write)(struct MWRITER*, const void*, size_t);
+    int  (*Put)(struct MWRITER*, int);
+} MWRITER;
+
+/*
+ *  ========== Samples
+ */
+
+/* Sample playback should not be interrupted */
+#define SFX_CRITICAL 1
+
+/* Sample format [loading and in-memory] flags: */
+#define SF_16BITS       0x0001
+#define SF_STEREO       0x0002
+#define SF_SIGNED       0x0004
+#define SF_BIG_ENDIAN   0x0008
+#define SF_DELTA        0x0010
+#define SF_ITPACKED     0x0020
+#define SF_ADPCM4       0x0040
+
+#define SF_FORMATMASK   0x007F
+
+/* General Playback flags */
+
+#define SF_LOOP         0x0100
+#define SF_BIDI         0x0200
+#define SF_REVERSE      0x0400
+#define SF_SUSTAIN      0x0800
+
+#define SF_PLAYBACKMASK 0x0C00
+
+/* Module-only Playback Flags */
+
+#define SF_OWNPAN       0x1000
+#define SF_UST_LOOP     0x2000
+
+#define SF_EXTRAPLAYBACKMASK    0x3000
+
+/* Panning constants */
+#define PAN_LEFT        0
+#define PAN_HALFLEFT    64
+#define PAN_CENTER      128
+#define PAN_HALFRIGHT   192
+#define PAN_RIGHT       255
+#define PAN_SURROUND    512 /* panning value for Dolby Surround */
+
+typedef struct SAMPLE {
+    SWORD  panning;     /* panning (0-255 or PAN_SURROUND) */
+    ULONG  speed;       /* Base playing speed/frequency of note */
+    UBYTE  volume;      /* volume 0-64 */
+    UWORD  inflags;     /* sample format on disk */
+    UWORD  flags;       /* sample format in memory */
+    ULONG  length;      /* length of sample (in samples!) */
+    ULONG  loopstart;   /* repeat position (relative to start, in samples) */
+    ULONG  loopend;     /* repeat end */
+    ULONG  susbegin;    /* sustain loop begin (in samples) \  Not Supported */
+    ULONG  susend;      /* sustain loop end                /      Yet! */
+
+    /* Variables used by the module player only! (ignored for sound effects) */
+    UBYTE  globvol;     /* global volume */
+    UBYTE  vibflags;    /* autovibrato flag stuffs */
+    UBYTE  vibtype;     /* Vibratos moved from INSTRUMENT to SAMPLE */
+    UBYTE  vibsweep;
+    UBYTE  vibdepth;
+    UBYTE  vibrate;
+    CHAR*  samplename;  /* name of the sample */
+
+    /* Values used internally only */
+    UWORD  avibpos;     /* autovibrato pos [player use] */
+    UBYTE  divfactor;   /* for sample scaling, maintains proper period slides */
+    ULONG  seekpos;     /* seek position in file */
+    SWORD  handle;      /* sample handle used by individual drivers */
+    void (*onfree)(void *ctx); /* called from Sample_Free if not NULL */
+    void *ctx;          /* context passed to previous function*/
+} SAMPLE;
+
+/* Sample functions */
+
+MIKMODAPI extern SAMPLE *Sample_LoadRaw(const CHAR *,ULONG rate, ULONG channel, ULONG flags);
+MIKMODAPI extern SAMPLE *Sample_LoadRawFP(FILE *fp,ULONG rate,ULONG channel, ULONG flags);
+MIKMODAPI extern SAMPLE *Sample_LoadRawMem(const char *buf, int len, ULONG rate, ULONG channel, ULONG flags);
+MIKMODAPI extern SAMPLE *Sample_LoadRawGeneric(MREADER*reader,ULONG rate, ULONG channel, ULONG flags);
+
+MIKMODAPI extern SAMPLE *Sample_Load(const CHAR*);
+MIKMODAPI extern SAMPLE *Sample_LoadFP(FILE*);
+MIKMODAPI extern SAMPLE *Sample_LoadMem(const char *buf, int len);
+MIKMODAPI extern SAMPLE *Sample_LoadGeneric(MREADER*);
+MIKMODAPI extern void   Sample_Free(SAMPLE*);
+MIKMODAPI extern SBYTE  Sample_Play(SAMPLE*,ULONG,UBYTE);
+
+MIKMODAPI extern void   Voice_SetVolume(SBYTE,UWORD);
+MIKMODAPI extern UWORD  Voice_GetVolume(SBYTE);
+MIKMODAPI extern void   Voice_SetFrequency(SBYTE,ULONG);
+MIKMODAPI extern ULONG  Voice_GetFrequency(SBYTE);
+MIKMODAPI extern void   Voice_SetPanning(SBYTE,ULONG);
+MIKMODAPI extern ULONG  Voice_GetPanning(SBYTE);
+MIKMODAPI extern void   Voice_Play(SBYTE,SAMPLE*,ULONG);
+MIKMODAPI extern void   Voice_Stop(SBYTE);
+MIKMODAPI extern BOOL   Voice_Stopped(SBYTE);
+MIKMODAPI extern SLONG  Voice_GetPosition(SBYTE);
+MIKMODAPI extern ULONG  Voice_RealVolume(SBYTE);
+
+/*
+ *  ========== Internal module representation (UniMod)
+ */
+
+/*
+    Instrument definition - for information only, the only field which may be
+    of use in user programs is the name field
+*/
+
+/* Instrument note count */
+#define INSTNOTES 120
+
+/* Envelope point */
+typedef struct ENVPT {
+    SWORD pos;
+    SWORD val;
+} ENVPT;
+
+/* Envelope point count */
+#define ENVPOINTS 32
+
+/* Instrument structure */
+typedef struct INSTRUMENT {
+    CHAR* insname;
+
+    UBYTE flags;
+    UWORD samplenumber[INSTNOTES];
+    UBYTE samplenote[INSTNOTES];
+
+    UBYTE nnatype;
+    UBYTE dca;              /* duplicate check action */
+    UBYTE dct;              /* duplicate check type */
+    UBYTE globvol;
+    UWORD volfade;
+    SWORD panning;          /* instrument-based panning var */
+
+    UBYTE pitpansep;        /* pitch pan separation (0 to 255) */
+    UBYTE pitpancenter;     /* pitch pan center (0 to 119) */
+    UBYTE rvolvar;          /* random volume varations (0 - 100%) */
+    UBYTE rpanvar;          /* random panning varations (0 - 100%) */
+
+    /* volume envelope */
+    UBYTE volflg;           /* bit 0: on 1: sustain 2: loop */
+    UBYTE volpts;
+    UBYTE volsusbeg;
+    UBYTE volsusend;
+    UBYTE volbeg;
+    UBYTE volend;
+    ENVPT volenv[ENVPOINTS];
+    /* panning envelope */
+    UBYTE panflg;           /* bit 0: on 1: sustain 2: loop */
+    UBYTE panpts;
+    UBYTE pansusbeg;
+    UBYTE pansusend;
+    UBYTE panbeg;
+    UBYTE panend;
+    ENVPT panenv[ENVPOINTS];
+    /* pitch envelope */
+    UBYTE pitflg;           /* bit 0: on 1: sustain 2: loop */
+    UBYTE pitpts;
+    UBYTE pitsusbeg;
+    UBYTE pitsusend;
+    UBYTE pitbeg;
+    UBYTE pitend;
+    ENVPT pitenv[ENVPOINTS];
+} INSTRUMENT;
+
+struct MP_CONTROL;
+struct MP_VOICE;
+
+/*
+    Module definition
+*/
+
+/* maximum master channels supported */
+#define UF_MAXCHAN      64
+
+/* Module flags */
+#define UF_XMPERIODS    0x0001 /* XM periods / finetuning */
+#define UF_LINEAR       0x0002 /* LINEAR periods (UF_XMPERIODS must be set) */
+#define UF_INST         0x0004 /* Instruments are used */
+#define UF_NNA          0x0008 /* IT: NNA used, set numvoices rather
+                                  than numchn */
+#define UF_S3MSLIDES    0x0010 /* uses old S3M volume slides */
+#define UF_BGSLIDES     0x0020 /* continue volume slides in the background */
+#define UF_HIGHBPM      0x0040 /* MED: can use >255 bpm */
+#define UF_NOWRAP       0x0080 /* XM-type (i.e. illogical) pattern break
+                                  semantics */
+#define UF_ARPMEM       0x0100 /* IT: need arpeggio memory */
+#define UF_FT2QUIRKS    0x0200 /* emulate some FT2 replay quirks */
+#define UF_PANNING      0x0400 /* module uses panning effects or have
+                                  non-tracker default initial panning */
+#define UF_FARTEMPO     0x0800 /* Module uses Farandole tempo calculations */
+
+typedef struct MODULE {
+ /* general module information */
+    CHAR*       songname;    /* name of the song */
+    CHAR*       modtype;     /* string type of module loaded */
+    CHAR*       comment;     /* module comments */
+
+    UWORD       flags;       /* See module flags above */
+    UBYTE       numchn;      /* number of module channels */
+    UBYTE       numvoices;   /* max # voices used for full NNA playback */
+    UWORD       numpos;      /* number of positions in this song */
+    UWORD       numpat;      /* number of patterns in this song */
+    UWORD       numins;      /* number of instruments */
+    UWORD       numsmp;      /* number of samples */
+
+    struct INSTRUMENT* instruments; /* all instruments */
+    struct SAMPLE*     samples;     /* all samples */
+
+    UBYTE       realchn;     /* real number of channels used */
+    UBYTE       totalchn;    /* total number of channels used (incl NNAs) */
+
+ /* playback settings */
+    UWORD       reppos;      /* restart position */
+    UBYTE       initspeed;   /* initial song speed */
+    UWORD       inittempo;   /* initial song tempo */
+    UBYTE       initvolume;  /* initial global volume (0 - 128) */
+    UWORD       panning[UF_MAXCHAN]; /* panning positions */
+    UBYTE       chanvol[UF_MAXCHAN]; /* channel positions */
+    UWORD       bpm;         /* current beats-per-minute speed */
+    UWORD       sngspd;      /* current song speed */
+    SWORD       volume;      /* song volume (0-128) (or user volume) */
+
+    BOOL        extspd;      /* extended speed flag (default enabled) */
+    BOOL        panflag;     /* panning flag (default enabled) */
+    BOOL        wrap;        /* wrap module ? (default disabled) */
+    BOOL        loop;        /* allow module to loop ? (default enabled) */
+    BOOL        fadeout;     /* volume fade out during last pattern */
+
+    UWORD       patpos;      /* current row number */
+    SWORD       sngpos;      /* current song position */
+    ULONG       sngtime;     /* current song time in 2^-10 seconds */
+
+    SWORD       relspd;      /* relative speed factor */
+
+ /* internal module representation */
+    UWORD       numtrk;      /* number of tracks */
+    UBYTE**     tracks;      /* array of numtrk pointers to tracks */
+    UWORD*      patterns;    /* array of Patterns */
+    UWORD*      pattrows;    /* array of number of rows for each pattern */
+    UWORD*      positions;   /* all positions */
+
+    BOOL        forbid;      /* if true, no player update! */
+    UWORD       numrow;      /* number of rows on current pattern */
+    UWORD       vbtick;      /* tick counter (counts from 0 to sngspd) */
+    UWORD       sngremainder;/* used for song time computation */
+
+    struct MP_CONTROL* control; /* Effects Channel info (size pf->numchn) */
+    struct MP_VOICE*   voice;   /* Audio Voice information (size md_numchn) */
+
+    UBYTE       globalslide; /* global volume slide rate */
+    UBYTE       pat_repcrazy;/* module has just looped to position -1 */
+    UWORD       patbrk;      /* position where to start a new pattern */
+    UBYTE       patdly;      /* patterndelay counter (command memory) */
+    UBYTE       patdly2;     /* patterndelay counter (real one) */
+    SWORD       posjmp;      /* flag to indicate a jump is needed... */
+    UWORD       bpmlimit;    /* threshold to detect bpm or speed values */
+} MODULE;
+
+
+/* This structure is used to query current playing voices status */
+typedef struct VOICEINFO {
+    INSTRUMENT* i;            /* Current channel instrument */
+    SAMPLE*     s;            /* Current channel sample */
+    SWORD       panning;      /* panning position */
+    SBYTE       volume;       /* channel's "global" volume (0..64) */
+    UWORD       period;       /* period to play the sample at */
+    UBYTE       kick;         /* if true = sample has been restarted */
+} VOICEINFO;
+
+/*
+ *  ========== Module loaders
+ */
+
+struct MLOADER;
+
+MIKMODAPI extern CHAR*   MikMod_InfoLoader(void);
+MIKMODAPI extern void    MikMod_RegisterAllLoaders(void);
+MIKMODAPI extern void    MikMod_RegisterLoader(struct MLOADER*);
+
+MIKMODAPI extern struct MLOADER load_669; /* 669 and Extended-669 (by Tran/Renaissance) */
+MIKMODAPI extern struct MLOADER load_amf; /* DMP Advanced Module Format (by Otto Chrons) */
+MIKMODAPI extern struct MLOADER load_asy; /* ASYLUM Music Format 1.0 */
+MIKMODAPI extern struct MLOADER load_dsm; /* DSIK internal module format */
+MIKMODAPI extern struct MLOADER load_far; /* Farandole Composer (by Daniel Potter) */
+MIKMODAPI extern struct MLOADER load_gdm; /* General DigiMusic (by Edward Schlunder) */
+MIKMODAPI extern struct MLOADER load_gt2; /* Graoumf tracker */
+MIKMODAPI extern struct MLOADER load_it;  /* Impulse Tracker (by Jeffrey Lim) */
+MIKMODAPI extern struct MLOADER load_imf; /* Imago Orpheus (by Lutz Roeder) */
+MIKMODAPI extern struct MLOADER load_med; /* Amiga MED modules (by Teijo Kinnunen) */
+MIKMODAPI extern struct MLOADER load_m15; /* Soundtracker 15-instrument */
+MIKMODAPI extern struct MLOADER load_mod; /* Standard 31-instrument Module loader */
+MIKMODAPI extern struct MLOADER load_mtm; /* Multi-Tracker Module (by Renaissance) */
+MIKMODAPI extern struct MLOADER load_okt; /* Amiga Oktalyzer */
+MIKMODAPI extern struct MLOADER load_stm; /* ScreamTracker 2 (by Future Crew) */
+MIKMODAPI extern struct MLOADER load_stx; /* STMIK 0.2 (by Future Crew) */
+MIKMODAPI extern struct MLOADER load_s3m; /* ScreamTracker 3 (by Future Crew) */
+MIKMODAPI extern struct MLOADER load_ult; /* UltraTracker (by MAS) */
+MIKMODAPI extern struct MLOADER load_umx; /* Unreal UMX container of Epic Games */
+MIKMODAPI extern struct MLOADER load_uni; /* MikMod and APlayer internal module format */
+MIKMODAPI extern struct MLOADER load_xm;  /* FastTracker 2 (by Triton) */
+
+/*
+ *  ========== Module player
+ */
+
+MIKMODAPI extern MODULE* Player_Load(const CHAR*,int,BOOL);
+MIKMODAPI extern MODULE* Player_LoadFP(FILE*,int,BOOL);
+MIKMODAPI extern MODULE* Player_LoadMem(const char *buffer,int len,int maxchan,BOOL curious);
+MIKMODAPI extern MODULE* Player_LoadGeneric(MREADER*,int,BOOL);
+MIKMODAPI extern CHAR*   Player_LoadTitle(const CHAR*);
+MIKMODAPI extern CHAR*   Player_LoadTitleFP(FILE*);
+MIKMODAPI extern CHAR*   Player_LoadTitleMem(const char *buffer,int len);
+MIKMODAPI extern CHAR*   Player_LoadTitleGeneric(MREADER*);
+
+MIKMODAPI extern void    Player_Free(MODULE*);
+MIKMODAPI extern void    Player_Start(MODULE*);
+MIKMODAPI extern BOOL    Player_Active(void);
+MIKMODAPI extern void    Player_Stop(void);
+MIKMODAPI extern void    Player_TogglePause(void);
+MIKMODAPI extern BOOL    Player_Paused(void);
+MIKMODAPI extern void    Player_NextPosition(void);
+MIKMODAPI extern void    Player_PrevPosition(void);
+MIKMODAPI extern void    Player_SetPosition(UWORD);
+MIKMODAPI extern BOOL    Player_Muted(UBYTE);
+MIKMODAPI extern void    Player_SetVolume(SWORD);
+MIKMODAPI extern MODULE* Player_GetModule(void);
+MIKMODAPI extern void    Player_SetSpeed(UWORD);
+MIKMODAPI extern void    Player_SetTempo(UWORD);
+MIKMODAPI extern void    Player_Unmute(SLONG,...);
+MIKMODAPI extern void    Player_Mute(SLONG,...);
+MIKMODAPI extern void    Player_ToggleMute(SLONG,...);
+MIKMODAPI extern int     Player_GetChannelVoice(UBYTE);
+MIKMODAPI extern UWORD   Player_GetChannelPeriod(UBYTE);
+MIKMODAPI extern int     Player_QueryVoices(UWORD numvoices, VOICEINFO *vinfo);
+MIKMODAPI extern int     Player_GetRow(void);
+MIKMODAPI extern int     Player_GetOrder(void);
+
+typedef void (*MikMod_player_t)(void);
+typedef void (*MikMod_callback_t)(unsigned char *data, size_t len);
+
+MIKMODAPI extern MikMod_player_t MikMod_RegisterPlayer(MikMod_player_t);
+
+#define MUTE_EXCLUSIVE  32000
+#define MUTE_INCLUSIVE  32001
+
+/*
+ *  ========== Drivers
+ */
+
+enum {
+    MD_MUSIC = 0,
+    MD_SNDFX
+};
+
+enum {
+    MD_HARDWARE = 0,
+    MD_SOFTWARE
+};
+
+/* Mixing flags */
+
+/* These ones take effect only after MikMod_Init or MikMod_Reset */
+#define DMODE_16BITS     0x0001 /* enable 16 bit output */
+#define DMODE_STEREO     0x0002 /* enable stereo output */
+#define DMODE_SOFT_SNDFX 0x0004 /* Process sound effects via software mixer */
+#define DMODE_SOFT_MUSIC 0x0008 /* Process music via software mixer */
+#define DMODE_HQMIXER    0x0010 /* Use high-quality (slower) software mixer */
+#define DMODE_FLOAT      0x0020 /* enable float output */
+/* These take effect immediately. */
+#define DMODE_SURROUND   0x0100 /* enable surround sound */
+#define DMODE_INTERP     0x0200 /* enable interpolation */
+#define DMODE_REVERSE    0x0400 /* reverse stereo */
+#define DMODE_SIMDMIXER  0x0800 /* enable SIMD mixing */
+#define DMODE_NOISEREDUCTION 0x1000 /* Low pass filtering */
+
+
+struct SAMPLOAD;
+
+typedef struct MDRIVER {
+    struct MDRIVER* next;
+    const CHAR* Name;
+    const CHAR* Version;
+
+    UBYTE       HardVoiceLimit; /* Limit of hardware mixer voices */
+    UBYTE       SoftVoiceLimit; /* Limit of software mixer voices */
+
+    const CHAR* Alias;
+    const CHAR* CmdLineHelp;
+
+    void        (*CommandLine)      (const CHAR*);
+    BOOL        (*IsPresent)        (void);
+    SWORD       (*SampleLoad)       (struct SAMPLOAD*,int);
+    void        (*SampleUnload)     (SWORD);
+    ULONG       (*FreeSampleSpace)  (int);
+    ULONG       (*RealSampleLength) (int,struct SAMPLE*);
+    int         (*Init)             (void);
+    void        (*Exit)             (void);
+    int         (*Reset)            (void);
+    int         (*SetNumVoices)     (void);
+    int         (*PlayStart)        (void);
+    void        (*PlayStop)         (void);
+    void        (*Update)           (void);
+    void        (*Pause)            (void);
+    void        (*VoiceSetVolume)   (UBYTE,UWORD);
+    UWORD       (*VoiceGetVolume)   (UBYTE);
+    void        (*VoiceSetFrequency)(UBYTE,ULONG);
+    ULONG       (*VoiceGetFrequency)(UBYTE);
+    void        (*VoiceSetPanning)  (UBYTE,ULONG);
+    ULONG       (*VoiceGetPanning)  (UBYTE);
+    void        (*VoicePlay)        (UBYTE,SWORD,ULONG,ULONG,ULONG,ULONG,UWORD);
+    void        (*VoiceStop)        (UBYTE);
+    BOOL        (*VoiceStopped)     (UBYTE);
+    SLONG       (*VoiceGetPosition) (UBYTE);
+    ULONG       (*VoiceRealVolume)  (UBYTE);
+} MDRIVER;
+
+/* These variables can be changed at ANY time and results will be immediate */
+MIKMODAPI extern UBYTE md_volume;      /* global sound volume (0-128) */
+MIKMODAPI extern UBYTE md_musicvolume; /* volume of song */
+MIKMODAPI extern UBYTE md_sndfxvolume; /* volume of sound effects */
+MIKMODAPI extern UBYTE md_reverb;      /* 0 = none;  15 = chaos */
+MIKMODAPI extern UBYTE md_pansep;      /* 0 = mono;  128 == 100% (full left/right) */
+
+/* The variables below can be changed at any time, but changes will not be
+   implemented until MikMod_Reset is called. A call to MikMod_Reset may result
+   in a skip or pop in audio (depending on the soundcard driver and the settings
+   changed). */
+MIKMODAPI extern UWORD md_device;      /* device */
+MIKMODAPI extern UWORD md_mixfreq;     /* mixing frequency */
+MIKMODAPI extern UWORD md_mode;        /* mode. See DMODE_? flags above */
+
+/* The following variable should not be changed! */
+MIKMODAPI extern MDRIVER* md_driver;   /* Current driver in use. */
+
+/* Known drivers list */
+
+MIKMODAPI extern struct MDRIVER drv_nos;    /* no sound */
+MIKMODAPI extern struct MDRIVER drv_pipe;   /* piped output */
+MIKMODAPI extern struct MDRIVER drv_raw;    /* raw file disk writer [music.raw] */
+MIKMODAPI extern struct MDRIVER drv_stdout; /* output to stdout */
+MIKMODAPI extern struct MDRIVER drv_wav;    /* RIFF WAVE file disk writer [music.wav] */
+MIKMODAPI extern struct MDRIVER drv_aiff;   /* AIFF file disk writer [music.aiff] */
+
+MIKMODAPI extern struct MDRIVER drv_ultra;  /* Linux Ultrasound driver */
+MIKMODAPI extern struct MDRIVER drv_sam9407;/* Linux sam9407 driver */
+
+MIKMODAPI extern struct MDRIVER drv_AF;     /* Dec Alpha AudioFile */
+MIKMODAPI extern struct MDRIVER drv_ahi;    /* Amiga AHI */
+MIKMODAPI extern struct MDRIVER drv_aix;    /* AIX audio device */
+MIKMODAPI extern struct MDRIVER drv_alsa;   /* Advanced Linux Sound Architecture (ALSA) */
+MIKMODAPI extern struct MDRIVER drv_esd;    /* Enlightened sound daemon (EsounD) */
+MIKMODAPI extern struct MDRIVER drv_pulseaudio; /* PulseAudio  */
+MIKMODAPI extern struct MDRIVER drv_hp;     /* HP-UX audio device */
+MIKMODAPI extern struct MDRIVER drv_nas;    /* Network Audio System (NAS) */
+MIKMODAPI extern struct MDRIVER drv_oss;    /* OpenSound System (Linux,FreeBSD...) */
+MIKMODAPI extern struct MDRIVER drv_openal; /* OpenAL driver */
+MIKMODAPI extern struct MDRIVER drv_sdl;    /* SDL audio driver */
+MIKMODAPI extern struct MDRIVER drv_sgi;    /* SGI audio library */
+MIKMODAPI extern struct MDRIVER drv_sndio;  /* OpenBSD sndio */
+MIKMODAPI extern struct MDRIVER drv_sun;    /* Sun/NetBSD/OpenBSD audio device */
+
+MIKMODAPI extern struct MDRIVER drv_dart;   /* OS/2 Direct Audio RealTime */
+MIKMODAPI extern struct MDRIVER drv_os2;    /* OS/2 MMPM/2 */
+
+MIKMODAPI extern struct MDRIVER drv_ds;     /* Win32 DirectSound driver */
+MIKMODAPI extern struct MDRIVER drv_xaudio2;/* Win32 XAudio2 driver */
+MIKMODAPI extern struct MDRIVER drv_win;    /* Win32 multimedia API driver */
+
+MIKMODAPI extern struct MDRIVER drv_mac;    /* Macintosh Sound Manager driver */
+MIKMODAPI extern struct MDRIVER drv_osx;    /* MacOS X CoreAudio Driver */
+
+MIKMODAPI extern struct MDRIVER drv_dc;     /* Dreamcast driver */
+MIKMODAPI extern struct MDRIVER drv_gp32;   /* GP32 Sound driver */
+MIKMODAPI extern struct MDRIVER drv_psp;    /* PlayStation Portable driver */
+MIKMODAPI extern struct MDRIVER drv_n64;    /* Nintendo64 driver */
+
+MIKMODAPI extern struct MDRIVER drv_wss;    /* DOS WSS driver */
+MIKMODAPI extern struct MDRIVER drv_sb;     /* DOS S/B driver */
+
+MIKMODAPI extern struct MDRIVER drv_osles;  /* OpenSL ES driver for android */
+
+/*========== Virtual channel mixer interface (for user-supplied drivers only) */
+
+MIKMODAPI extern int   VC_Init(void);
+MIKMODAPI extern void  VC_Exit(void);
+MIKMODAPI extern void  VC_SetCallback(MikMod_callback_t callback);
+MIKMODAPI extern int   VC_SetNumVoices(void);
+MIKMODAPI extern ULONG VC_SampleSpace(int);
+MIKMODAPI extern ULONG VC_SampleLength(int,SAMPLE*);
+
+MIKMODAPI extern int   VC_PlayStart(void);
+MIKMODAPI extern void  VC_PlayStop(void);
+
+MIKMODAPI extern SWORD VC_SampleLoad(struct SAMPLOAD*,int);
+MIKMODAPI extern void  VC_SampleUnload(SWORD);
+
+MIKMODAPI extern ULONG VC_WriteBytes(SBYTE*,ULONG);
+MIKMODAPI extern ULONG VC_SilenceBytes(SBYTE*,ULONG);
+
+MIKMODAPI extern void  VC_VoiceSetVolume(UBYTE,UWORD);
+MIKMODAPI extern UWORD VC_VoiceGetVolume(UBYTE);
+MIKMODAPI extern void  VC_VoiceSetFrequency(UBYTE,ULONG);
+MIKMODAPI extern ULONG VC_VoiceGetFrequency(UBYTE);
+MIKMODAPI extern void  VC_VoiceSetPanning(UBYTE,ULONG);
+MIKMODAPI extern ULONG VC_VoiceGetPanning(UBYTE);
+MIKMODAPI extern void  VC_VoicePlay(UBYTE,SWORD,ULONG,ULONG,ULONG,ULONG,UWORD);
+
+MIKMODAPI extern void  VC_VoiceStop(UBYTE);
+MIKMODAPI extern BOOL  VC_VoiceStopped(UBYTE);
+MIKMODAPI extern SLONG VC_VoiceGetPosition(UBYTE);
+MIKMODAPI extern ULONG VC_VoiceRealVolume(UBYTE);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif
+
+/* ex:set ts=4: */

MacOSX/codecs/include/mikmod_config.txt

@@ -0,0 +1,2 @@
+libmikmod-3.3.12. only the "nosound" driver (drv_nos) is included:
+we only need/register/use drv_nos here and nothing else.

MacOSX/codecs/include/mpg123_config.txt

@@ -0,0 +1,5 @@
+mpg123-1.22.4, configured using:
+--disable-modules --disable-debug --disable-fifo --disable-ipv6 --disable-network --disable-messages --disable-lfs-alias --with-audio=dummy
+edited src/libmpg123/mpg123lib_intern.h and changed macros
+NOQUIET, VERBOSE* and PVERB() to be 0, in order to disable
+some debug messages from the library.

MacOSX/codecs/include/ogg/config_types.h

@@ -0,0 +1,25 @@
+#ifndef __CONFIG_TYPES_H__
+#define __CONFIG_TYPES_H__
+
+/* these are filled in by configure */
+#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;
+
+#endif

MacOSX/codecs/include/ogg/ogg.h

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

MacOSX/codecs/include/ogg/os_types.h

@@ -0,0 +1,147 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: #ifdef jail to whip a few platforms into the UNIX ideal.
+ last mod: $Id: os_types.h 19098 2014-02-26 19:06:45Z giles $
+
+ ********************************************************************/
+#ifndef _OS_TYPES_H
+#define _OS_TYPES_H
+
+/* make it easy on the folks that want to compile the libs with a
+   different malloc than stdlib */
+#define _ogg_malloc  malloc
+#define _ogg_calloc  calloc
+#define _ogg_realloc realloc
+#define _ogg_free    free
+
+#if defined(_WIN32)
+
+#  if defined(__CYGWIN__)
+#    include <stdint.h>
+     typedef int16_t ogg_int16_t;
+     typedef uint16_t ogg_uint16_t;
+     typedef int32_t ogg_int32_t;
+     typedef uint32_t ogg_uint32_t;
+     typedef int64_t ogg_int64_t;
+     typedef uint64_t ogg_uint64_t;
+#  elif defined(__MINGW32__)
+#    include <sys/types.h>
+     typedef short ogg_int16_t;
+     typedef unsigned short ogg_uint16_t;
+     typedef int ogg_int32_t;
+     typedef unsigned int ogg_uint32_t;
+     typedef long long ogg_int64_t;
+     typedef unsigned long long ogg_uint64_t;
+#  elif defined(__MWERKS__)
+     typedef long long ogg_int64_t;
+     typedef int ogg_int32_t;
+     typedef unsigned int ogg_uint32_t;
+     typedef short ogg_int16_t;
+     typedef unsigned short ogg_uint16_t;
+#  else
+     /* MSVC/Borland */
+     typedef __int64 ogg_int64_t;
+     typedef __int32 ogg_int32_t;
+     typedef unsigned __int32 ogg_uint32_t;
+     typedef __int16 ogg_int16_t;
+     typedef unsigned __int16 ogg_uint16_t;
+#  endif
+
+#elif defined(__MACOS__)
+
+#  include <sys/types.h>
+   typedef SInt16 ogg_int16_t;
+   typedef UInt16 ogg_uint16_t;
+   typedef SInt32 ogg_int32_t;
+   typedef UInt32 ogg_uint32_t;
+   typedef SInt64 ogg_int64_t;
+
+#elif (defined(__APPLE__) && defined(__MACH__)) /* MacOS X Framework build */
+
+#  include <inttypes.h>
+   typedef int16_t ogg_int16_t;
+   typedef uint16_t ogg_uint16_t;
+   typedef int32_t ogg_int32_t;
+   typedef uint32_t ogg_uint32_t;
+   typedef int64_t ogg_int64_t;
+
+#elif defined(__HAIKU__)
+
+  /* Haiku */
+#  include <sys/types.h>
+   typedef short ogg_int16_t;
+   typedef unsigned short ogg_uint16_t;
+   typedef int ogg_int32_t;
+   typedef unsigned int ogg_uint32_t;
+   typedef long long ogg_int64_t;
+
+#elif defined(__BEOS__)
+
+   /* Be */
+#  include <inttypes.h>
+   typedef int16_t ogg_int16_t;
+   typedef uint16_t ogg_uint16_t;
+   typedef int32_t ogg_int32_t;
+   typedef uint32_t ogg_uint32_t;
+   typedef int64_t ogg_int64_t;
+
+#elif defined (__EMX__)
+
+   /* OS/2 GCC */
+   typedef short ogg_int16_t;
+   typedef unsigned short ogg_uint16_t;
+   typedef int ogg_int32_t;
+   typedef unsigned int ogg_uint32_t;
+   typedef long long ogg_int64_t;
+
+#elif defined (DJGPP)
+
+   /* DJGPP */
+   typedef short ogg_int16_t;
+   typedef int ogg_int32_t;
+   typedef unsigned int ogg_uint32_t;
+   typedef long long ogg_int64_t;
+
+#elif defined(R5900)
+
+   /* PS2 EE */
+   typedef long ogg_int64_t;
+   typedef int ogg_int32_t;
+   typedef unsigned ogg_uint32_t;
+   typedef short ogg_int16_t;
+
+#elif defined(__SYMBIAN32__)
+
+   /* Symbian GCC */
+   typedef signed short ogg_int16_t;
+   typedef unsigned short ogg_uint16_t;
+   typedef signed int ogg_int32_t;
+   typedef unsigned int ogg_uint32_t;
+   typedef long long int ogg_int64_t;
+
+#elif defined(__TMS320C6X__)
+
+   /* TI C64x compiler */
+   typedef signed short ogg_int16_t;
+   typedef unsigned short ogg_uint16_t;
+   typedef signed int ogg_int32_t;
+   typedef unsigned int ogg_uint32_t;
+   typedef long long int ogg_int64_t;
+
+#else
+
+#  include <ogg/config_types.h>
+
+#endif
+
+#endif  /* _OS_TYPES_H */

MacOSX/codecs/include/opus/opus.h

@@ -0,0 +1,906 @@
+/* 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 1 byte, 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 (@ref OPUS_APPLICATION_VOIP/@ref OPUS_APPLICATION_AUDIO/@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 (OPUS_APPLICATION_VOIP/OPUS_APPLICATION_AUDIO/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);
+/**@}*/
+
+/** @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 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);
+
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_H */

MacOSX/codecs/include/opus/opus_defines.h

@@ -0,0 +1,659 @@
+/* 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
+/** The mode struct passed is invalid @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(__GNUC__) && !OPUS_GNUC_PREREQ(3,4))
+/* __restrict is broken with gcc < 3.4
+   https://gcc.gnu.org/bugzilla/show_bug.cgi?id=6392 */
+# define OPUS_RESTRICT
+#elif (!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
+
+/**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
+
+/* Don't use 4045, it's already taken by OPUS_GET_GAIN_REQUEST */
+
+/* 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)))
+/** @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*/
+
+/**@}*/
+
+
+/** @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.
+  * @warning Only the MDCT mode of Opus can provide hard CBR behavior.
+  * @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 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)
+
+/** 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>Enable inband FEC.</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.</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 with trigger progressively more loss resistant behavior in the encoder
+  * at the expense of quality at a given bitrate in the lossless case, 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.
+  * @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)
+
+/** 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)
+/**@}*/
+
+/** @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 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)
+
+/** 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)
+
+/**@}*/
+
+/** @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)
+
+/**@}*/
+
+/** @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.
+  *
+  * @returns Version string
+  */
+OPUS_EXPORT const char *opus_get_version_string(void);
+/**@}*/
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* OPUS_DEFINES_H */

MacOSX/codecs/include/opus/opus_multistream.h

@@ -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 the
+  * <a href="http://tools.ietf.org/html/draft-terriberry-oggopus">Ogg
+  * encapsulation specification</a> and is based on the self-delimited Opus
+  * framing described in Appendix B of <a href="http://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="http://www.xiph.org/vorbis/doc/Vorbis_I_spec.html#x1-800004.3.9">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(5);
+
+/** 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(6);
+
+/** 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 */

MacOSX/codecs/include/opus/opus_types.h

@@ -0,0 +1,159 @@
+/* (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
+
+/* Use the real stdint.h if it's there (taken from Paul Hsieh's pstdint.h) */
+#if (defined(__STDC__) && __STDC__ && __STDC_VERSION__ >= 199901L) || (defined(__GNUC__) && (defined(_STDINT_H) || defined(_STDINT_H_)) || defined (HAVE_STDINT_H))
+#include <stdint.h>
+
+   typedef int16_t opus_int16;
+   typedef uint16_t opus_uint16;
+   typedef int32_t opus_int32;
+   typedef uint32_t opus_uint32;
+#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
+
+#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
+
+#endif  /* OPUS_TYPES_H */

MacOSX/codecs/include/opusfile.h

@@ -0,0 +1,3 @@
+/* just a wrapper to bypass the pkg-config thingy: the
+ * headers under opus/ are edited accordingly for this */
+#include <opus/opusfile.h>

MacOSX/codecs/include/vorbis/codec.h

@@ -0,0 +1,242 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2001             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+
+ ********************************************************************
+
+ function: libvorbis codec headers
+
+ ********************************************************************/
+
+#ifndef _vorbis_codec_h_
+#define _vorbis_codec_h_
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+#include <ogg/ogg.h>
+
+typedef struct vorbis_info{
+  int version;
+  int channels;
+  long rate;
+
+  /* The below bitrate declarations are *hints*.
+     Combinations of the three values carry the following implications:
+
+     all three set to the same value:
+       implies a fixed rate bitstream
+     only nominal set:
+       implies a VBR stream that averages the nominal bitrate.  No hard
+       upper/lower limit
+     upper and or lower set:
+       implies a VBR bitstream that obeys the bitrate limits. nominal
+       may also be set to give a nominal rate.
+     none set:
+       the coder does not care to speculate.
+  */
+
+  long bitrate_upper;
+  long bitrate_nominal;
+  long bitrate_lower;
+  long bitrate_window;
+
+  void *codec_setup;
+} vorbis_info;
+
+/* vorbis_dsp_state buffers the current vorbis audio
+   analysis/synthesis state.  The DSP state belongs to a specific
+   logical bitstream ****************************************************/
+typedef struct vorbis_dsp_state{
+  int analysisp;
+  vorbis_info *vi;
+
+  float **pcm;
+  float **pcmret;
+  int      pcm_storage;
+  int      pcm_current;
+  int      pcm_returned;
+
+  int  preextrapolate;
+  int  eofflag;
+
+  long lW;
+  long W;
+  long nW;
+  long centerW;
+
+  ogg_int64_t granulepos;
+  ogg_int64_t sequence;
+
+  ogg_int64_t glue_bits;
+  ogg_int64_t time_bits;
+  ogg_int64_t floor_bits;
+  ogg_int64_t res_bits;
+
+  void       *backend_state;
+} vorbis_dsp_state;
+
+typedef struct vorbis_block{
+  /* necessary stream state for linking to the framing abstraction */
+  float  **pcm;       /* this is a pointer into local storage */
+  oggpack_buffer opb;
+
+  long  lW;
+  long  W;
+  long  nW;
+  int   pcmend;
+  int   mode;
+
+  int         eofflag;
+  ogg_int64_t granulepos;
+  ogg_int64_t sequence;
+  vorbis_dsp_state *vd; /* For read-only access of configuration */
+
+  /* local storage to avoid remallocing; it's up to the mapping to
+     structure it */
+  void               *localstore;
+  long                localtop;
+  long                localalloc;
+  long                totaluse;
+  struct alloc_chain *reap;
+
+  /* bitmetrics for the frame */
+  long glue_bits;
+  long time_bits;
+  long floor_bits;
+  long res_bits;
+
+  void *internal;
+
+} vorbis_block;
+
+/* vorbis_block is a single block of data to be processed as part of
+the analysis/synthesis stream; it belongs to a specific logical
+bitstream, but is independent from other vorbis_blocks belonging to
+that logical bitstream. *************************************************/
+
+struct alloc_chain{
+  void *ptr;
+  struct alloc_chain *next;
+};
+
+/* vorbis_info contains all the setup information specific to the
+   specific compression/decompression mode in progress (eg,
+   psychoacoustic settings, channel setup, options, codebook
+   etc). vorbis_info and substructures are in backends.h.
+*********************************************************************/
+
+/* the comments are not part of vorbis_info so that vorbis_info can be
+   static storage */
+typedef struct vorbis_comment{
+  /* unlimited user comment fields.  libvorbis writes 'libvorbis'
+     whatever vendor is set to in encode */
+  char **user_comments;
+  int   *comment_lengths;
+  int    comments;
+  char  *vendor;
+
+} vorbis_comment;
+
+
+/* libvorbis encodes in two abstraction layers; first we perform DSP
+   and produce a packet (see docs/analysis.txt).  The packet is then
+   coded into a framed OggSquish bitstream by the second layer (see
+   docs/framing.txt).  Decode is the reverse process; we sync/frame
+   the bitstream and extract individual packets, then decode the
+   packet back into PCM audio.
+
+   The extra framing/packetizing is used in streaming formats, such as
+   files.  Over the net (such as with UDP), the framing and
+   packetization aren't necessary as they're provided by the transport
+   and the streaming layer is not used */
+
+/* Vorbis PRIMITIVES: general ***************************************/
+
+extern void     vorbis_info_init(vorbis_info *vi);
+extern void     vorbis_info_clear(vorbis_info *vi);
+extern int      vorbis_info_blocksize(vorbis_info *vi,int zo);
+extern void     vorbis_comment_init(vorbis_comment *vc);
+extern void     vorbis_comment_add(vorbis_comment *vc, const char *comment);
+extern void     vorbis_comment_add_tag(vorbis_comment *vc,
+                                       const char *tag, const char *contents);
+extern char    *vorbis_comment_query(vorbis_comment *vc, const char *tag, int count);
+extern int      vorbis_comment_query_count(vorbis_comment *vc, const char *tag);
+extern void     vorbis_comment_clear(vorbis_comment *vc);
+
+extern int      vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb);
+extern int      vorbis_block_clear(vorbis_block *vb);
+extern void     vorbis_dsp_clear(vorbis_dsp_state *v);
+extern double   vorbis_granule_time(vorbis_dsp_state *v,
+                                    ogg_int64_t granulepos);
+
+extern const char *vorbis_version_string(void);
+
+/* Vorbis PRIMITIVES: analysis/DSP layer ****************************/
+
+extern int      vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi);
+extern int      vorbis_commentheader_out(vorbis_comment *vc, ogg_packet *op);
+extern int      vorbis_analysis_headerout(vorbis_dsp_state *v,
+                                          vorbis_comment *vc,
+                                          ogg_packet *op,
+                                          ogg_packet *op_comm,
+                                          ogg_packet *op_code);
+extern float  **vorbis_analysis_buffer(vorbis_dsp_state *v,int vals);
+extern int      vorbis_analysis_wrote(vorbis_dsp_state *v,int vals);
+extern int      vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb);
+extern int      vorbis_analysis(vorbis_block *vb,ogg_packet *op);
+
+extern int      vorbis_bitrate_addblock(vorbis_block *vb);
+extern int      vorbis_bitrate_flushpacket(vorbis_dsp_state *vd,
+                                           ogg_packet *op);
+
+/* Vorbis PRIMITIVES: synthesis layer *******************************/
+extern int      vorbis_synthesis_idheader(ogg_packet *op);
+extern int      vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc,
+                                          ogg_packet *op);
+
+extern int      vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi);
+extern int      vorbis_synthesis_restart(vorbis_dsp_state *v);
+extern int      vorbis_synthesis(vorbis_block *vb,ogg_packet *op);
+extern int      vorbis_synthesis_trackonly(vorbis_block *vb,ogg_packet *op);
+extern int      vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb);
+extern int      vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm);
+extern int      vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm);
+extern int      vorbis_synthesis_read(vorbis_dsp_state *v,int samples);
+extern long     vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op);
+
+extern int      vorbis_synthesis_halfrate(vorbis_info *v,int flag);
+extern int      vorbis_synthesis_halfrate_p(vorbis_info *v);
+
+/* Vorbis ERRORS and return codes ***********************************/
+
+#define OV_FALSE      -1
+#define OV_EOF        -2
+#define OV_HOLE       -3
+
+#define OV_EREAD      -128
+#define OV_EFAULT     -129
+#define OV_EIMPL      -130
+#define OV_EINVAL     -131
+#define OV_ENOTVORBIS -132
+#define OV_EBADHEADER -133
+#define OV_EVERSION   -134
+#define OV_ENOTAUDIO  -135
+#define OV_EBADPACKET -136
+#define OV_EBADLINK   -137
+#define OV_ENOSEEK    -138
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#endif
+

MacOSX/codecs/include/vorbis/vorbisfile.h

@@ -0,0 +1,205 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: stdio-based convenience library for opening/seeking/decoding
+
+ ********************************************************************/
+
+#ifndef _OV_FILE_H_
+#define _OV_FILE_H_
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+#include <stdio.h>
+#include "codec.h"
+
+/* The function prototypes for the callbacks are basically the same as for
+ * the stdio functions fread, fseek, fclose, ftell.
+ * The one difference is that the FILE * arguments have been replaced with
+ * a void * - this is to be used as a pointer to whatever internal data these
+ * functions might need. In the stdio case, it's just a FILE * cast to a void *
+ *
+ * If you use other functions, check the docs for these functions and return
+ * the right values. For seek_func(), you *MUST* return -1 if the stream is
+ * unseekable
+ */
+typedef struct {
+  size_t (*read_func)  (void *ptr, size_t size, size_t nmemb, void *datasource);
+  int    (*seek_func)  (void *datasource, ogg_int64_t offset, int whence);
+  int    (*close_func) (void *datasource);
+  long   (*tell_func)  (void *datasource);
+} ov_callbacks;
+
+#ifndef OV_EXCLUDE_STATIC_CALLBACKS
+
+/* a few sets of convenient callbacks, especially for use under
+ * Windows where ov_open_callbacks() should always be used instead of
+ * ov_open() to avoid problems with incompatible crt.o version linking
+ * issues. */
+
+static int _ov_header_fseek_wrap(FILE *f,ogg_int64_t off,int whence){
+  if(f==NULL)return(-1);
+
+#ifdef __MINGW32__
+  return fseeko64(f,off,whence);
+#elif defined (_WIN32)
+  return _fseeki64(f,off,whence);
+#else
+  return fseek(f,off,whence);
+#endif
+}
+
+/* These structs below (OV_CALLBACKS_DEFAULT etc) are defined here as
+ * static data. That means that every file which includes this header
+ * will get its own copy of these structs whether it uses them or
+ * not unless it #defines OV_EXCLUDE_STATIC_CALLBACKS.
+ * These static symbols are essential on platforms such as Windows on
+ * which several different versions of stdio support may be linked to
+ * by different DLLs, and we need to be certain we know which one
+ * we're using (the same one as the main application).
+ */
+
+static ov_callbacks OV_CALLBACKS_DEFAULT = {
+  (size_t (*)(void *, size_t, size_t, void *))  fread,
+  (int (*)(void *, ogg_int64_t, int))           _ov_header_fseek_wrap,
+  (int (*)(void *))                             fclose,
+  (long (*)(void *))                            ftell
+};
+
+static ov_callbacks OV_CALLBACKS_NOCLOSE = {
+  (size_t (*)(void *, size_t, size_t, void *))  fread,
+  (int (*)(void *, ogg_int64_t, int))           _ov_header_fseek_wrap,
+  (int (*)(void *))                             NULL,
+  (long (*)(void *))                            ftell
+};
+
+static ov_callbacks OV_CALLBACKS_STREAMONLY = {
+  (size_t (*)(void *, size_t, size_t, void *))  fread,
+  (int (*)(void *, ogg_int64_t, int))           NULL,
+  (int (*)(void *))                             fclose,
+  (long (*)(void *))                            NULL
+};
+
+static ov_callbacks OV_CALLBACKS_STREAMONLY_NOCLOSE = {
+  (size_t (*)(void *, size_t, size_t, void *))  fread,
+  (int (*)(void *, ogg_int64_t, int))           NULL,
+  (int (*)(void *))                             NULL,
+  (long (*)(void *))                            NULL
+};
+
+#endif
+
+#define  NOTOPEN   0
+#define  PARTOPEN  1
+#define  OPENED    2
+#define  STREAMSET 3
+#define  INITSET   4
+
+typedef struct OggVorbis_File {
+  void            *datasource; /* Pointer to a FILE *, etc. */
+  int              seekable;
+  ogg_int64_t      offset;
+  ogg_int64_t      end;
+  ogg_sync_state   oy;
+
+  /* If the FILE handle isn't seekable (eg, a pipe), only the current
+     stream appears */
+  int              links;
+  ogg_int64_t     *offsets;
+  ogg_int64_t     *dataoffsets;
+  long            *serialnos;
+  ogg_int64_t     *pcmlengths; /* overloaded to maintain binary
+                                  compatibility; x2 size, stores both
+                                  beginning and end values */
+  vorbis_info     *vi;
+  vorbis_comment  *vc;
+
+  /* Decoding working state local storage */
+  ogg_int64_t      pcm_offset;
+  int              ready_state;
+  long             current_serialno;
+  int              current_link;
+
+  double           bittrack;
+  double           samptrack;
+
+  ogg_stream_state os; /* take physical pages, weld into a logical
+                          stream of packets */
+  vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
+  vorbis_block     vb; /* local working space for packet->PCM decode */
+
+  ov_callbacks callbacks;
+
+} OggVorbis_File;
+
+
+extern int ov_clear(OggVorbis_File *vf);
+extern int ov_fopen(const char *path,OggVorbis_File *vf);
+extern int ov_open(FILE *f,OggVorbis_File *vf,const char *initial,long ibytes);
+extern int ov_open_callbacks(void *datasource, OggVorbis_File *vf,
+                const char *initial, long ibytes, ov_callbacks callbacks);
+
+extern int ov_test(FILE *f,OggVorbis_File *vf,const char *initial,long ibytes);
+extern int ov_test_callbacks(void *datasource, OggVorbis_File *vf,
+                const char *initial, long ibytes, ov_callbacks callbacks);
+extern int ov_test_open(OggVorbis_File *vf);
+
+extern long ov_bitrate(OggVorbis_File *vf,int i);
+extern long ov_bitrate_instant(OggVorbis_File *vf);
+extern long ov_streams(OggVorbis_File *vf);
+extern long ov_seekable(OggVorbis_File *vf);
+extern long ov_serialnumber(OggVorbis_File *vf,int i);
+
+extern ogg_int64_t ov_raw_total(OggVorbis_File *vf,int i);
+extern ogg_int64_t ov_pcm_total(OggVorbis_File *vf,int i);
+extern double ov_time_total(OggVorbis_File *vf,int i);
+
+extern int ov_raw_seek(OggVorbis_File *vf,ogg_int64_t pos);
+extern int ov_pcm_seek(OggVorbis_File *vf,ogg_int64_t pos);
+extern int ov_pcm_seek_page(OggVorbis_File *vf,ogg_int64_t pos);
+extern int ov_time_seek(OggVorbis_File *vf,double pos);
+extern int ov_time_seek_page(OggVorbis_File *vf,double pos);
+
+extern int ov_raw_seek_lap(OggVorbis_File *vf,ogg_int64_t pos);
+extern int ov_pcm_seek_lap(OggVorbis_File *vf,ogg_int64_t pos);
+extern int ov_pcm_seek_page_lap(OggVorbis_File *vf,ogg_int64_t pos);
+extern int ov_time_seek_lap(OggVorbis_File *vf,double pos);
+extern int ov_time_seek_page_lap(OggVorbis_File *vf,double pos);
+
+extern ogg_int64_t ov_raw_tell(OggVorbis_File *vf);
+extern ogg_int64_t ov_pcm_tell(OggVorbis_File *vf);
+extern double ov_time_tell(OggVorbis_File *vf);
+
+extern vorbis_info *ov_info(OggVorbis_File *vf,int link);
+extern vorbis_comment *ov_comment(OggVorbis_File *vf,int link);
+
+extern long ov_read_float(OggVorbis_File *vf,float ***pcm_channels,int samples,
+                          int *bitstream);
+extern long ov_read_filter(OggVorbis_File *vf,char *buffer,int length,
+                          int bigendianp,int word,int sgned,int *bitstream,
+                          void (*filter)(float **pcm,long channels,long samples,void *filter_param),void *filter_param);
+extern long ov_read(OggVorbis_File *vf,char *buffer,int length,
+                    int bigendianp,int word,int sgned,int *bitstream);
+extern int ov_crosslap(OggVorbis_File *vf1,OggVorbis_File *vf2);
+
+extern int ov_halfrate(OggVorbis_File *vf,int flag);
+extern int ov_halfrate_p(OggVorbis_File *vf);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#endif
+

MacOSX/codecs/include/xmp.h

@@ -0,0 +1,407 @@
+#ifndef XMP_H
+#define XMP_H
+
+#if defined(EMSCRIPTEN)
+# include <emscripten.h>
+#endif
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define XMP_VERSION "4.6.3"
+#define XMP_VERCODE 0x040603
+#define XMP_VER_MAJOR 4
+#define XMP_VER_MINOR 6
+#define XMP_VER_RELEASE 3
+
+#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 */
+
+/* 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		49170	/* 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 */
+	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 */
+	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 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_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 */