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diff --git a/3rdparty/include/SDL2/SDL_audio.h b/3rdparty/include/SDL2/SDL_audio.h
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+++ b/3rdparty/include/SDL2/SDL_audio.h
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+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2020 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+/**
+ *  \file SDL_audio.h
+ *
+ *  Access to the raw audio mixing buffer for the SDL library.
+ */
+
+#ifndef SDL_audio_h_
+#define SDL_audio_h_
+
+#include "SDL_stdinc.h"
+#include "SDL_error.h"
+#include "SDL_endian.h"
+#include "SDL_mutex.h"
+#include "SDL_thread.h"
+#include "SDL_rwops.h"
+
+#include "begin_code.h"
+/* Set up for C function definitions, even when using C++ */
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ *  \brief Audio format flags.
+ *
+ *  These are what the 16 bits in SDL_AudioFormat currently mean...
+ *  (Unspecified bits are always zero).
+ *
+ *  \verbatim
+    ++-----------------------sample is signed if set
+    ||
+    ||       ++-----------sample is bigendian if set
+    ||       ||
+    ||       ||          ++---sample is float if set
+    ||       ||          ||
+    ||       ||          || +---sample bit size---+
+    ||       ||          || |                     |
+    15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
+    \endverbatim
+ *
+ *  There are macros in SDL 2.0 and later to query these bits.
+ */
+typedef Uint16 SDL_AudioFormat;
+
+/**
+ *  \name Audio flags
+ */
+/* @{ */
+
+#define SDL_AUDIO_MASK_BITSIZE       (0xFF)
+#define SDL_AUDIO_MASK_DATATYPE      (1<<8)
+#define SDL_AUDIO_MASK_ENDIAN        (1<<12)
+#define SDL_AUDIO_MASK_SIGNED        (1<<15)
+#define SDL_AUDIO_BITSIZE(x)         (x & SDL_AUDIO_MASK_BITSIZE)
+#define SDL_AUDIO_ISFLOAT(x)         (x & SDL_AUDIO_MASK_DATATYPE)
+#define SDL_AUDIO_ISBIGENDIAN(x)     (x & SDL_AUDIO_MASK_ENDIAN)
+#define SDL_AUDIO_ISSIGNED(x)        (x & SDL_AUDIO_MASK_SIGNED)
+#define SDL_AUDIO_ISINT(x)           (!SDL_AUDIO_ISFLOAT(x))
+#define SDL_AUDIO_ISLITTLEENDIAN(x)  (!SDL_AUDIO_ISBIGENDIAN(x))
+#define SDL_AUDIO_ISUNSIGNED(x)      (!SDL_AUDIO_ISSIGNED(x))
+
+/**
+ *  \name Audio format flags
+ *
+ *  Defaults to LSB byte order.
+ */
+/* @{ */
+#define AUDIO_U8        0x0008  /**< Unsigned 8-bit samples */
+#define AUDIO_S8        0x8008  /**< Signed 8-bit samples */
+#define AUDIO_U16LSB    0x0010  /**< Unsigned 16-bit samples */
+#define AUDIO_S16LSB    0x8010  /**< Signed 16-bit samples */
+#define AUDIO_U16MSB    0x1010  /**< As above, but big-endian byte order */
+#define AUDIO_S16MSB    0x9010  /**< As above, but big-endian byte order */
+#define AUDIO_U16       AUDIO_U16LSB
+#define AUDIO_S16       AUDIO_S16LSB
+/* @} */
+
+/**
+ *  \name int32 support
+ */
+/* @{ */
+#define AUDIO_S32LSB    0x8020  /**< 32-bit integer samples */
+#define AUDIO_S32MSB    0x9020  /**< As above, but big-endian byte order */
+#define AUDIO_S32       AUDIO_S32LSB
+/* @} */
+
+/**
+ *  \name float32 support
+ */
+/* @{ */
+#define AUDIO_F32LSB    0x8120  /**< 32-bit floating point samples */
+#define AUDIO_F32MSB    0x9120  /**< As above, but big-endian byte order */
+#define AUDIO_F32       AUDIO_F32LSB
+/* @} */
+
+/**
+ *  \name Native audio byte ordering
+ */
+/* @{ */
+#if SDL_BYTEORDER == SDL_LIL_ENDIAN
+#define AUDIO_U16SYS    AUDIO_U16LSB
+#define AUDIO_S16SYS    AUDIO_S16LSB
+#define AUDIO_S32SYS    AUDIO_S32LSB
+#define AUDIO_F32SYS    AUDIO_F32LSB
+#else
+#define AUDIO_U16SYS    AUDIO_U16MSB
+#define AUDIO_S16SYS    AUDIO_S16MSB
+#define AUDIO_S32SYS    AUDIO_S32MSB
+#define AUDIO_F32SYS    AUDIO_F32MSB
+#endif
+/* @} */
+
+/**
+ *  \name Allow change flags
+ *
+ *  Which audio format changes are allowed when opening a device.
+ */
+/* @{ */
+#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE    0x00000001
+#define SDL_AUDIO_ALLOW_FORMAT_CHANGE       0x00000002
+#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE     0x00000004
+#define SDL_AUDIO_ALLOW_SAMPLES_CHANGE      0x00000008
+#define SDL_AUDIO_ALLOW_ANY_CHANGE          (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE|SDL_AUDIO_ALLOW_SAMPLES_CHANGE)
+/* @} */
+
+/* @} *//* Audio flags */
+
+/**
+ *  This function is called when the audio device needs more data.
+ *
+ *  \param userdata An application-specific parameter saved in
+ *                  the SDL_AudioSpec structure
+ *  \param stream A pointer to the audio data buffer.
+ *  \param len    The length of that buffer in bytes.
+ *
+ *  Once the callback returns, the buffer will no longer be valid.
+ *  Stereo samples are stored in a LRLRLR ordering.
+ *
+ *  You can choose to avoid callbacks and use SDL_QueueAudio() instead, if
+ *  you like. Just open your audio device with a NULL callback.
+ */
+typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
+                                            int len);
+
+/**
+ *  The calculated values in this structure are calculated by SDL_OpenAudio().
+ *
+ *  For multi-channel audio, the default SDL channel mapping is:
+ *  2:  FL FR                       (stereo)
+ *  3:  FL FR LFE                   (2.1 surround)
+ *  4:  FL FR BL BR                 (quad)
+ *  5:  FL FR FC BL BR              (quad + center)
+ *  6:  FL FR FC LFE SL SR          (5.1 surround - last two can also be BL BR)
+ *  7:  FL FR FC LFE BC SL SR       (6.1 surround)
+ *  8:  FL FR FC LFE BL BR SL SR    (7.1 surround)
+ */
+typedef struct SDL_AudioSpec
+{
+    int freq;                   /**< DSP frequency -- samples per second */
+    SDL_AudioFormat format;     /**< Audio data format */
+    Uint8 channels;             /**< Number of channels: 1 mono, 2 stereo */
+    Uint8 silence;              /**< Audio buffer silence value (calculated) */
+    Uint16 samples;             /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */
+    Uint16 padding;             /**< Necessary for some compile environments */
+    Uint32 size;                /**< Audio buffer size in bytes (calculated) */
+    SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
+    void *userdata;             /**< Userdata passed to callback (ignored for NULL callbacks). */
+} SDL_AudioSpec;
+
+
+struct SDL_AudioCVT;
+typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
+                                          SDL_AudioFormat format);
+
+/**
+ *  \brief Upper limit of filters in SDL_AudioCVT
+ *
+ *  The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is
+ *  currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers,
+ *  one of which is the terminating NULL pointer.
+ */
+#define SDL_AUDIOCVT_MAX_FILTERS 9
+
+/**
+ *  \struct SDL_AudioCVT
+ *  \brief A structure to hold a set of audio conversion filters and buffers.
+ *
+ *  Note that various parts of the conversion pipeline can take advantage
+ *  of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
+ *  you to pass it aligned data, but can possibly run much faster if you
+ *  set both its (buf) field to a pointer that is aligned to 16 bytes, and its
+ *  (len) field to something that's a multiple of 16, if possible.
+ */
+#ifdef __GNUC__
+/* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
+   pad it out to 88 bytes to guarantee ABI compatibility between compilers.
+   vvv
+   The next time we rev the ABI, make sure to size the ints and add padding.
+*/
+#define SDL_AUDIOCVT_PACKED __attribute__((packed))
+#else
+#define SDL_AUDIOCVT_PACKED
+#endif
+/* */
+typedef struct SDL_AudioCVT
+{
+    int needed;                 /**< Set to 1 if conversion possible */
+    SDL_AudioFormat src_format; /**< Source audio format */
+    SDL_AudioFormat dst_format; /**< Target audio format */
+    double rate_incr;           /**< Rate conversion increment */
+    Uint8 *buf;                 /**< Buffer to hold entire audio data */
+    int len;                    /**< Length of original audio buffer */
+    int len_cvt;                /**< Length of converted audio buffer */
+    int len_mult;               /**< buffer must be len*len_mult big */
+    double len_ratio;           /**< Given len, final size is len*len_ratio */
+    SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
+    int filter_index;           /**< Current audio conversion function */
+} SDL_AUDIOCVT_PACKED SDL_AudioCVT;
+
+
+/* Function prototypes */
+
+/**
+ *  \name Driver discovery functions
+ *
+ *  These functions return the list of built in audio drivers, in the
+ *  order that they are normally initialized by default.
+ */
+/* @{ */
+extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
+extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
+/* @} */
+
+/**
+ *  \name Initialization and cleanup
+ *
+ *  \internal These functions are used internally, and should not be used unless
+ *            you have a specific need to specify the audio driver you want to
+ *            use.  You should normally use SDL_Init() or SDL_InitSubSystem().
+ */
+/* @{ */
+extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
+extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
+/* @} */
+
+/**
+ *  This function returns the name of the current audio driver, or NULL
+ *  if no driver has been initialized.
+ */
+extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
+
+/**
+ *  This function opens the audio device with the desired parameters, and
+ *  returns 0 if successful, placing the actual hardware parameters in the
+ *  structure pointed to by \c obtained.  If \c obtained is NULL, the audio
+ *  data passed to the callback function will be guaranteed to be in the
+ *  requested format, and will be automatically converted to the hardware
+ *  audio format if necessary.  This function returns -1 if it failed
+ *  to open the audio device, or couldn't set up the audio thread.
+ *
+ *  When filling in the desired audio spec structure,
+ *    - \c desired->freq should be the desired audio frequency in samples-per-
+ *      second.
+ *    - \c desired->format should be the desired audio format.
+ *    - \c desired->samples is the desired size of the audio buffer, in
+ *      samples.  This number should be a power of two, and may be adjusted by
+ *      the audio driver to a value more suitable for the hardware.  Good values
+ *      seem to range between 512 and 8096 inclusive, depending on the
+ *      application and CPU speed.  Smaller values yield faster response time,
+ *      but can lead to underflow if the application is doing heavy processing
+ *      and cannot fill the audio buffer in time.  A stereo sample consists of
+ *      both right and left channels in LR ordering.
+ *      Note that the number of samples is directly related to time by the
+ *      following formula:  \code ms = (samples*1000)/freq \endcode
+ *    - \c desired->size is the size in bytes of the audio buffer, and is
+ *      calculated by SDL_OpenAudio().
+ *    - \c desired->silence is the value used to set the buffer to silence,
+ *      and is calculated by SDL_OpenAudio().
+ *    - \c desired->callback should be set to a function that will be called
+ *      when the audio device is ready for more data.  It is passed a pointer
+ *      to the audio buffer, and the length in bytes of the audio buffer.
+ *      This function usually runs in a separate thread, and so you should
+ *      protect data structures that it accesses by calling SDL_LockAudio()
+ *      and SDL_UnlockAudio() in your code. Alternately, you may pass a NULL
+ *      pointer here, and call SDL_QueueAudio() with some frequency, to queue
+ *      more audio samples to be played (or for capture devices, call
+ *      SDL_DequeueAudio() with some frequency, to obtain audio samples).
+ *    - \c desired->userdata is passed as the first parameter to your callback
+ *      function. If you passed a NULL callback, this value is ignored.
+ *
+ *  The audio device starts out playing silence when it's opened, and should
+ *  be enabled for playing by calling \c SDL_PauseAudio(0) when you are ready
+ *  for your audio callback function to be called.  Since the audio driver
+ *  may modify the requested size of the audio buffer, you should allocate
+ *  any local mixing buffers after you open the audio device.
+ */
+extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,
+                                          SDL_AudioSpec * obtained);
+
+/**
+ *  SDL Audio Device IDs.
+ *
+ *  A successful call to SDL_OpenAudio() is always device id 1, and legacy
+ *  SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls
+ *  always returns devices >= 2 on success. The legacy calls are good both
+ *  for backwards compatibility and when you don't care about multiple,
+ *  specific, or capture devices.
+ */
+typedef Uint32 SDL_AudioDeviceID;
+
+/**
+ *  Get the number of available devices exposed by the current driver.
+ *  Only valid after a successfully initializing the audio subsystem.
+ *  Returns -1 if an explicit list of devices can't be determined; this is
+ *  not an error. For example, if SDL is set up to talk to a remote audio
+ *  server, it can't list every one available on the Internet, but it will
+ *  still allow a specific host to be specified to SDL_OpenAudioDevice().
+ *
+ *  In many common cases, when this function returns a value <= 0, it can still
+ *  successfully open the default device (NULL for first argument of
+ *  SDL_OpenAudioDevice()).
+ */
+extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
+
+/**
+ *  Get the human-readable name of a specific audio device.
+ *  Must be a value between 0 and (number of audio devices-1).
+ *  Only valid after a successfully initializing the audio subsystem.
+ *  The values returned by this function reflect the latest call to
+ *  SDL_GetNumAudioDevices(); recall that function to redetect available
+ *  hardware.
+ *
+ *  The string returned by this function is UTF-8 encoded, read-only, and
+ *  managed internally. You are not to free it. If you need to keep the
+ *  string for any length of time, you should make your own copy of it, as it
+ *  will be invalid next time any of several other SDL functions is called.
+ */
+extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
+                                                           int iscapture);
+
+
+/**
+ *  Open a specific audio device. Passing in a device name of NULL requests
+ *  the most reasonable default (and is equivalent to calling SDL_OpenAudio()).
+ *
+ *  The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but
+ *  some drivers allow arbitrary and driver-specific strings, such as a
+ *  hostname/IP address for a remote audio server, or a filename in the
+ *  diskaudio driver.
+ *
+ *  \return 0 on error, a valid device ID that is >= 2 on success.
+ *
+ *  SDL_OpenAudio(), unlike this function, always acts on device ID 1.
+ */
+extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char
+                                                              *device,
+                                                              int iscapture,
+                                                              const
+                                                              SDL_AudioSpec *
+                                                              desired,
+                                                              SDL_AudioSpec *
+                                                              obtained,
+                                                              int
+                                                              allowed_changes);
+
+
+
+/**
+ *  \name Audio state
+ *
+ *  Get the current audio state.
+ */
+/* @{ */
+typedef enum
+{
+    SDL_AUDIO_STOPPED = 0,
+    SDL_AUDIO_PLAYING,
+    SDL_AUDIO_PAUSED
+} SDL_AudioStatus;
+extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
+
+extern DECLSPEC SDL_AudioStatus SDLCALL
+SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
+/* @} *//* Audio State */
+
+/**
+ *  \name Pause audio functions
+ *
+ *  These functions pause and unpause the audio callback processing.
+ *  They should be called with a parameter of 0 after opening the audio
+ *  device to start playing sound.  This is so you can safely initialize
+ *  data for your callback function after opening the audio device.
+ *  Silence will be written to the audio device during the pause.
+ */
+/* @{ */
+extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
+extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
+                                                  int pause_on);
+/* @} *//* Pause audio functions */
+
+/**
+ *  \brief Load the audio data of a WAVE file into memory
+ *
+ *  Loading a WAVE file requires \c src, \c spec, \c audio_buf and \c audio_len
+ *  to be valid pointers. The entire data portion of the file is then loaded
+ *  into memory and decoded if necessary.
+ *
+ *  If \c freesrc is non-zero, the data source gets automatically closed and
+ *  freed before the function returns.
+ *
+ *  Supported are RIFF WAVE files with the formats PCM (8, 16, 24, and 32 bits),
+ *  IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and A-law and
+ *  µ-law (8 bits). Other formats are currently unsupported and cause an error.
+ *
+ *  If this function succeeds, the pointer returned by it is equal to \c spec
+ *  and the pointer to the audio data allocated by the function is written to
+ *  \c audio_buf and its length in bytes to \c audio_len. The \ref SDL_AudioSpec
+ *  members \c freq, \c channels, and \c format are set to the values of the
+ *  audio data in the buffer. The \c samples member is set to a sane default and
+ *  all others are set to zero.
+ *
+ *  It's necessary to use SDL_FreeWAV() to free the audio data returned in
+ *  \c audio_buf when it is no longer used.
+ *
+ *  Because of the underspecification of the Waveform format, there are many
+ *  problematic files in the wild that cause issues with strict decoders. To
+ *  provide compatibility with these files, this decoder is lenient in regards
+ *  to the truncation of the file, the fact chunk, and the size of the RIFF
+ *  chunk. The hints SDL_HINT_WAVE_RIFF_CHUNK_SIZE, SDL_HINT_WAVE_TRUNCATION,
+ *  and SDL_HINT_WAVE_FACT_CHUNK can be used to tune the behavior of the
+ *  loading process.
+ *
+ *  Any file that is invalid (due to truncation, corruption, or wrong values in
+ *  the headers), too big, or unsupported causes an error. Additionally, any
+ *  critical I/O error from the data source will terminate the loading process
+ *  with an error. The function returns NULL on error and in all cases (with the
+ *  exception of \c src being NULL), an appropriate error message will be set.
+ *
+ *  It is required that the data source supports seeking.
+ *
+ *  Example:
+ *  \code
+ *      SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
+ *  \endcode
+ *
+ *  \param src The data source with the WAVE data
+ *  \param freesrc A integer value that makes the function close the data source if non-zero
+ *  \param spec A pointer filled with the audio format of the audio data
+ *  \param audio_buf A pointer filled with the audio data allocated by the function
+ *  \param audio_len A pointer filled with the length of the audio data buffer in bytes
+ *  \return NULL on error, or non-NULL on success.
+ */
+extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
+                                                      int freesrc,
+                                                      SDL_AudioSpec * spec,
+                                                      Uint8 ** audio_buf,
+                                                      Uint32 * audio_len);
+
+/**
+ *  Loads a WAV from a file.
+ *  Compatibility convenience function.
+ */
+#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
+    SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
+
+/**
+ *  This function frees data previously allocated with SDL_LoadWAV_RW()
+ */
+extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
+
+/**
+ *  This function takes a source format and rate and a destination format
+ *  and rate, and initializes the \c cvt structure with information needed
+ *  by SDL_ConvertAudio() to convert a buffer of audio data from one format
+ *  to the other. An unsupported format causes an error and -1 will be returned.
+ *
+ *  \return 0 if no conversion is needed, 1 if the audio filter is set up,
+ *  or -1 on error.
+ */
+extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
+                                              SDL_AudioFormat src_format,
+                                              Uint8 src_channels,
+                                              int src_rate,
+                                              SDL_AudioFormat dst_format,
+                                              Uint8 dst_channels,
+                                              int dst_rate);
+
+/**
+ *  Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(),
+ *  created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of
+ *  audio data in the source format, this function will convert it in-place
+ *  to the desired format.
+ *
+ *  The data conversion may expand the size of the audio data, so the buffer
+ *  \c cvt->buf should be allocated after the \c cvt structure is initialized by
+ *  SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long.
+ *
+ *  \return 0 on success or -1 if \c cvt->buf is NULL.
+ */
+extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
+
+/* SDL_AudioStream is a new audio conversion interface.
+   The benefits vs SDL_AudioCVT:
+    - it can handle resampling data in chunks without generating
+      artifacts, when it doesn't have the complete buffer available.
+    - it can handle incoming data in any variable size.
+    - You push data as you have it, and pull it when you need it
+ */
+/* this is opaque to the outside world. */
+struct _SDL_AudioStream;
+typedef struct _SDL_AudioStream SDL_AudioStream;
+
+/**
+ *  Create a new audio stream
+ *
+ *  \param src_format The format of the source audio
+ *  \param src_channels The number of channels of the source audio
+ *  \param src_rate The sampling rate of the source audio
+ *  \param dst_format The format of the desired audio output
+ *  \param dst_channels The number of channels of the desired audio output
+ *  \param dst_rate The sampling rate of the desired audio output
+ *  \return 0 on success, or -1 on error.
+ *
+ *  \sa SDL_AudioStreamPut
+ *  \sa SDL_AudioStreamGet
+ *  \sa SDL_AudioStreamAvailable
+ *  \sa SDL_AudioStreamFlush
+ *  \sa SDL_AudioStreamClear
+ *  \sa SDL_FreeAudioStream
+ */
+extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
+                                           const Uint8 src_channels,
+                                           const int src_rate,
+                                           const SDL_AudioFormat dst_format,
+                                           const Uint8 dst_channels,
+                                           const int dst_rate);
+
+/**
+ *  Add data to be converted/resampled to the stream
+ *
+ *  \param stream The stream the audio data is being added to
+ *  \param buf A pointer to the audio data to add
+ *  \param len The number of bytes to write to the stream
+ *  \return 0 on success, or -1 on error.
+ *
+ *  \sa SDL_NewAudioStream
+ *  \sa SDL_AudioStreamGet
+ *  \sa SDL_AudioStreamAvailable
+ *  \sa SDL_AudioStreamFlush
+ *  \sa SDL_AudioStreamClear
+ *  \sa SDL_FreeAudioStream
+ */
+extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
+
+/**
+ *  Get converted/resampled data from the stream
+ *
+ *  \param stream The stream the audio is being requested from
+ *  \param buf A buffer to fill with audio data
+ *  \param len The maximum number of bytes to fill
+ *  \return The number of bytes read from the stream, or -1 on error
+ *
+ *  \sa SDL_NewAudioStream
+ *  \sa SDL_AudioStreamPut
+ *  \sa SDL_AudioStreamAvailable
+ *  \sa SDL_AudioStreamFlush
+ *  \sa SDL_AudioStreamClear
+ *  \sa SDL_FreeAudioStream
+ */
+extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
+
+/**
+ * Get the number of converted/resampled bytes available. The stream may be
+ *  buffering data behind the scenes until it has enough to resample
+ *  correctly, so this number might be lower than what you expect, or even
+ *  be zero. Add more data or flush the stream if you need the data now.
+ *
+ *  \sa SDL_NewAudioStream
+ *  \sa SDL_AudioStreamPut
+ *  \sa SDL_AudioStreamGet
+ *  \sa SDL_AudioStreamFlush
+ *  \sa SDL_AudioStreamClear
+ *  \sa SDL_FreeAudioStream
+ */
+extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
+
+/**
+ * Tell the stream that you're done sending data, and anything being buffered
+ *  should be converted/resampled and made available immediately.
+ *
+ * It is legal to add more data to a stream after flushing, but there will
+ *  be audio gaps in the output. Generally this is intended to signal the
+ *  end of input, so the complete output becomes available.
+ *
+ *  \sa SDL_NewAudioStream
+ *  \sa SDL_AudioStreamPut
+ *  \sa SDL_AudioStreamGet
+ *  \sa SDL_AudioStreamAvailable
+ *  \sa SDL_AudioStreamClear
+ *  \sa SDL_FreeAudioStream
+ */
+extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);
+
+/**
+ *  Clear any pending data in the stream without converting it
+ *
+ *  \sa SDL_NewAudioStream
+ *  \sa SDL_AudioStreamPut
+ *  \sa SDL_AudioStreamGet
+ *  \sa SDL_AudioStreamAvailable
+ *  \sa SDL_AudioStreamFlush
+ *  \sa SDL_FreeAudioStream
+ */
+extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
+
+/**
+ * Free an audio stream
+ *
+ *  \sa SDL_NewAudioStream
+ *  \sa SDL_AudioStreamPut
+ *  \sa SDL_AudioStreamGet
+ *  \sa SDL_AudioStreamAvailable
+ *  \sa SDL_AudioStreamFlush
+ *  \sa SDL_AudioStreamClear
+ */
+extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
+
+#define SDL_MIX_MAXVOLUME 128
+/**
+ *  This takes two audio buffers of the playing audio format and mixes
+ *  them, performing addition, volume adjustment, and overflow clipping.
+ *  The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME
+ *  for full audio volume.  Note this does not change hardware volume.
+ *  This is provided for convenience -- you can mix your own audio data.
+ */
+extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
+                                          Uint32 len, int volume);
+
+/**
+ *  This works like SDL_MixAudio(), but you specify the audio format instead of
+ *  using the format of audio device 1. Thus it can be used when no audio
+ *  device is open at all.
+ */
+extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
+                                                const Uint8 * src,
+                                                SDL_AudioFormat format,
+                                                Uint32 len, int volume);
+
+/**
+ *  Queue more audio on non-callback devices.
+ *
+ *  (If you are looking to retrieve queued audio from a non-callback capture
+ *  device, you want SDL_DequeueAudio() instead. This will return -1 to
+ *  signify an error if you use it with capture devices.)
+ *
+ *  SDL offers two ways to feed audio to the device: you can either supply a
+ *  callback that SDL triggers with some frequency to obtain more audio
+ *  (pull method), or you can supply no callback, and then SDL will expect
+ *  you to supply data at regular intervals (push method) with this function.
+ *
+ *  There are no limits on the amount of data you can queue, short of
+ *  exhaustion of address space. Queued data will drain to the device as
+ *  necessary without further intervention from you. If the device needs
+ *  audio but there is not enough queued, it will play silence to make up
+ *  the difference. This means you will have skips in your audio playback
+ *  if you aren't routinely queueing sufficient data.
+ *
+ *  This function copies the supplied data, so you are safe to free it when
+ *  the function returns. This function is thread-safe, but queueing to the
+ *  same device from two threads at once does not promise which buffer will
+ *  be queued first.
+ *
+ *  You may not queue audio on a device that is using an application-supplied
+ *  callback; doing so returns an error. You have to use the audio callback
+ *  or queue audio with this function, but not both.
+ *
+ *  You should not call SDL_LockAudio() on the device before queueing; SDL
+ *  handles locking internally for this function.
+ *
+ *  \param dev The device ID to which we will queue audio.
+ *  \param data The data to queue to the device for later playback.
+ *  \param len The number of bytes (not samples!) to which (data) points.
+ *  \return 0 on success, or -1 on error.
+ *
+ *  \sa SDL_GetQueuedAudioSize
+ *  \sa SDL_ClearQueuedAudio
+ */
+extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
+
+/**
+ *  Dequeue more audio on non-callback devices.
+ *
+ *  (If you are looking to queue audio for output on a non-callback playback
+ *  device, you want SDL_QueueAudio() instead. This will always return 0
+ *  if you use it with playback devices.)
+ *
+ *  SDL offers two ways to retrieve audio from a capture device: you can
+ *  either supply a callback that SDL triggers with some frequency as the
+ *  device records more audio data, (push method), or you can supply no
+ *  callback, and then SDL will expect you to retrieve data at regular
+ *  intervals (pull method) with this function.
+ *
+ *  There are no limits on the amount of data you can queue, short of
+ *  exhaustion of address space. Data from the device will keep queuing as
+ *  necessary without further intervention from you. This means you will
+ *  eventually run out of memory if you aren't routinely dequeueing data.
+ *
+ *  Capture devices will not queue data when paused; if you are expecting
+ *  to not need captured audio for some length of time, use
+ *  SDL_PauseAudioDevice() to stop the capture device from queueing more
+ *  data. This can be useful during, say, level loading times. When
+ *  unpaused, capture devices will start queueing data from that point,
+ *  having flushed any capturable data available while paused.
+ *
+ *  This function is thread-safe, but dequeueing from the same device from
+ *  two threads at once does not promise which thread will dequeued data
+ *  first.
+ *
+ *  You may not dequeue audio from a device that is using an
+ *  application-supplied callback; doing so returns an error. You have to use
+ *  the audio callback, or dequeue audio with this function, but not both.
+ *
+ *  You should not call SDL_LockAudio() on the device before queueing; SDL
+ *  handles locking internally for this function.
+ *
+ *  \param dev The device ID from which we will dequeue audio.
+ *  \param data A pointer into where audio data should be copied.
+ *  \param len The number of bytes (not samples!) to which (data) points.
+ *  \return number of bytes dequeued, which could be less than requested.
+ *
+ *  \sa SDL_GetQueuedAudioSize
+ *  \sa SDL_ClearQueuedAudio
+ */
+extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
+
+/**
+ *  Get the number of bytes of still-queued audio.
+ *
+ *  For playback device:
+ *
+ *    This is the number of bytes that have been queued for playback with
+ *    SDL_QueueAudio(), but have not yet been sent to the hardware. This
+ *    number may shrink at any time, so this only informs of pending data.
+ *
+ *    Once we've sent it to the hardware, this function can not decide the
+ *    exact byte boundary of what has been played. It's possible that we just
+ *    gave the hardware several kilobytes right before you called this
+ *    function, but it hasn't played any of it yet, or maybe half of it, etc.
+ *
+ *  For capture devices:
+ *
+ *    This is the number of bytes that have been captured by the device and
+ *    are waiting for you to dequeue. This number may grow at any time, so
+ *    this only informs of the lower-bound of available data.
+ *
+ *  You may not queue audio on a device that is using an application-supplied
+ *  callback; calling this function on such a device always returns 0.
+ *  You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
+ *  the audio callback, but not both.
+ *
+ *  You should not call SDL_LockAudio() on the device before querying; SDL
+ *  handles locking internally for this function.
+ *
+ *  \param dev The device ID of which we will query queued audio size.
+ *  \return Number of bytes (not samples!) of queued audio.
+ *
+ *  \sa SDL_QueueAudio
+ *  \sa SDL_ClearQueuedAudio
+ */
+extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
+
+/**
+ *  Drop any queued audio data. For playback devices, this is any queued data
+ *  still waiting to be submitted to the hardware. For capture devices, this
+ *  is any data that was queued by the device that hasn't yet been dequeued by
+ *  the application.
+ *
+ *  Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
+ *  playback devices, the hardware will start playing silence if more audio
+ *  isn't queued. Unpaused capture devices will start filling the queue again
+ *  as soon as they have more data available (which, depending on the state
+ *  of the hardware and the thread, could be before this function call
+ *  returns!).
+ *
+ *  This will not prevent playback of queued audio that's already been sent
+ *  to the hardware, as we can not undo that, so expect there to be some
+ *  fraction of a second of audio that might still be heard. This can be
+ *  useful if you want to, say, drop any pending music during a level change
+ *  in your game.
+ *
+ *  You may not queue audio on a device that is using an application-supplied
+ *  callback; calling this function on such a device is always a no-op.
+ *  You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
+ *  the audio callback, but not both.
+ *
+ *  You should not call SDL_LockAudio() on the device before clearing the
+ *  queue; SDL handles locking internally for this function.
+ *
+ *  This function always succeeds and thus returns void.
+ *
+ *  \param dev The device ID of which to clear the audio queue.
+ *
+ *  \sa SDL_QueueAudio
+ *  \sa SDL_GetQueuedAudioSize
+ */
+extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);
+
+
+/**
+ *  \name Audio lock functions
+ *
+ *  The lock manipulated by these functions protects the callback function.
+ *  During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that
+ *  the callback function is not running.  Do not call these from the callback
+ *  function or you will cause deadlock.
+ */
+/* @{ */
+extern DECLSPEC void SDLCALL SDL_LockAudio(void);
+extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
+extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
+extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
+/* @} *//* Audio lock functions */
+
+/**
+ *  This function shuts down audio processing and closes the audio device.
+ */
+extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
+extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
+
+/* Ends C function definitions when using C++ */
+#ifdef __cplusplus
+}
+#endif
+#include "close_code.h"
+
+#endif /* SDL_audio_h_ */
+
+/* vi: set ts=4 sw=4 expandtab: */