include/SDL_audio.h
author Ozkan Sezer <sezeroz@gmail.com>
Thu, 16 Aug 2018 11:01:02 +0300
changeset 12120 0131b11bd03f
parent 11811 5d94cb6b24d3
permissions -rw-r--r--
SDL_hidapi_ps4.c: define NTDDI_VISTA / _WIN32_WINNT_VISTA if not defined

it still needs a Vista or newer Platform SDK to build, though.
     1 /*
     2   Simple DirectMedia Layer
     3   Copyright (C) 1997-2018 Sam Lantinga <slouken@libsdl.org>
     4 
     5   This software is provided 'as-is', without any express or implied
     6   warranty.  In no event will the authors be held liable for any damages
     7   arising from the use of this software.
     8 
     9   Permission is granted to anyone to use this software for any purpose,
    10   including commercial applications, and to alter it and redistribute it
    11   freely, subject to the following restrictions:
    12 
    13   1. The origin of this software must not be misrepresented; you must not
    14      claim that you wrote the original software. If you use this software
    15      in a product, an acknowledgment in the product documentation would be
    16      appreciated but is not required.
    17   2. Altered source versions must be plainly marked as such, and must not be
    18      misrepresented as being the original software.
    19   3. This notice may not be removed or altered from any source distribution.
    20 */
    21 
    22 /**
    23  *  \file SDL_audio.h
    24  *
    25  *  Access to the raw audio mixing buffer for the SDL library.
    26  */
    27 
    28 #ifndef SDL_audio_h_
    29 #define SDL_audio_h_
    30 
    31 #include "SDL_stdinc.h"
    32 #include "SDL_error.h"
    33 #include "SDL_endian.h"
    34 #include "SDL_mutex.h"
    35 #include "SDL_thread.h"
    36 #include "SDL_rwops.h"
    37 
    38 #include "begin_code.h"
    39 /* Set up for C function definitions, even when using C++ */
    40 #ifdef __cplusplus
    41 extern "C" {
    42 #endif
    43 
    44 /**
    45  *  \brief Audio format flags.
    46  *
    47  *  These are what the 16 bits in SDL_AudioFormat currently mean...
    48  *  (Unspecified bits are always zero).
    49  *
    50  *  \verbatim
    51     ++-----------------------sample is signed if set
    52     ||
    53     ||       ++-----------sample is bigendian if set
    54     ||       ||
    55     ||       ||          ++---sample is float if set
    56     ||       ||          ||
    57     ||       ||          || +---sample bit size---+
    58     ||       ||          || |                     |
    59     15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
    60     \endverbatim
    61  *
    62  *  There are macros in SDL 2.0 and later to query these bits.
    63  */
    64 typedef Uint16 SDL_AudioFormat;
    65 
    66 /**
    67  *  \name Audio flags
    68  */
    69 /* @{ */
    70 
    71 #define SDL_AUDIO_MASK_BITSIZE       (0xFF)
    72 #define SDL_AUDIO_MASK_DATATYPE      (1<<8)
    73 #define SDL_AUDIO_MASK_ENDIAN        (1<<12)
    74 #define SDL_AUDIO_MASK_SIGNED        (1<<15)
    75 #define SDL_AUDIO_BITSIZE(x)         (x & SDL_AUDIO_MASK_BITSIZE)
    76 #define SDL_AUDIO_ISFLOAT(x)         (x & SDL_AUDIO_MASK_DATATYPE)
    77 #define SDL_AUDIO_ISBIGENDIAN(x)     (x & SDL_AUDIO_MASK_ENDIAN)
    78 #define SDL_AUDIO_ISSIGNED(x)        (x & SDL_AUDIO_MASK_SIGNED)
    79 #define SDL_AUDIO_ISINT(x)           (!SDL_AUDIO_ISFLOAT(x))
    80 #define SDL_AUDIO_ISLITTLEENDIAN(x)  (!SDL_AUDIO_ISBIGENDIAN(x))
    81 #define SDL_AUDIO_ISUNSIGNED(x)      (!SDL_AUDIO_ISSIGNED(x))
    82 
    83 /**
    84  *  \name Audio format flags
    85  *
    86  *  Defaults to LSB byte order.
    87  */
    88 /* @{ */
    89 #define AUDIO_U8        0x0008  /**< Unsigned 8-bit samples */
    90 #define AUDIO_S8        0x8008  /**< Signed 8-bit samples */
    91 #define AUDIO_U16LSB    0x0010  /**< Unsigned 16-bit samples */
    92 #define AUDIO_S16LSB    0x8010  /**< Signed 16-bit samples */
    93 #define AUDIO_U16MSB    0x1010  /**< As above, but big-endian byte order */
    94 #define AUDIO_S16MSB    0x9010  /**< As above, but big-endian byte order */
    95 #define AUDIO_U16       AUDIO_U16LSB
    96 #define AUDIO_S16       AUDIO_S16LSB
    97 /* @} */
    98 
    99 /**
   100  *  \name int32 support
   101  */
   102 /* @{ */
   103 #define AUDIO_S32LSB    0x8020  /**< 32-bit integer samples */
   104 #define AUDIO_S32MSB    0x9020  /**< As above, but big-endian byte order */
   105 #define AUDIO_S32       AUDIO_S32LSB
   106 /* @} */
   107 
   108 /**
   109  *  \name float32 support
   110  */
   111 /* @{ */
   112 #define AUDIO_F32LSB    0x8120  /**< 32-bit floating point samples */
   113 #define AUDIO_F32MSB    0x9120  /**< As above, but big-endian byte order */
   114 #define AUDIO_F32       AUDIO_F32LSB
   115 /* @} */
   116 
   117 /**
   118  *  \name Native audio byte ordering
   119  */
   120 /* @{ */
   121 #if SDL_BYTEORDER == SDL_LIL_ENDIAN
   122 #define AUDIO_U16SYS    AUDIO_U16LSB
   123 #define AUDIO_S16SYS    AUDIO_S16LSB
   124 #define AUDIO_S32SYS    AUDIO_S32LSB
   125 #define AUDIO_F32SYS    AUDIO_F32LSB
   126 #else
   127 #define AUDIO_U16SYS    AUDIO_U16MSB
   128 #define AUDIO_S16SYS    AUDIO_S16MSB
   129 #define AUDIO_S32SYS    AUDIO_S32MSB
   130 #define AUDIO_F32SYS    AUDIO_F32MSB
   131 #endif
   132 /* @} */
   133 
   134 /**
   135  *  \name Allow change flags
   136  *
   137  *  Which audio format changes are allowed when opening a device.
   138  */
   139 /* @{ */
   140 #define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE    0x00000001
   141 #define SDL_AUDIO_ALLOW_FORMAT_CHANGE       0x00000002
   142 #define SDL_AUDIO_ALLOW_CHANNELS_CHANGE     0x00000004
   143 #define SDL_AUDIO_ALLOW_ANY_CHANGE          (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE)
   144 /* @} */
   145 
   146 /* @} *//* Audio flags */
   147 
   148 /**
   149  *  This function is called when the audio device needs more data.
   150  *
   151  *  \param userdata An application-specific parameter saved in
   152  *                  the SDL_AudioSpec structure
   153  *  \param stream A pointer to the audio data buffer.
   154  *  \param len    The length of that buffer in bytes.
   155  *
   156  *  Once the callback returns, the buffer will no longer be valid.
   157  *  Stereo samples are stored in a LRLRLR ordering.
   158  *
   159  *  You can choose to avoid callbacks and use SDL_QueueAudio() instead, if
   160  *  you like. Just open your audio device with a NULL callback.
   161  */
   162 typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
   163                                             int len);
   164 
   165 /**
   166  *  The calculated values in this structure are calculated by SDL_OpenAudio().
   167  *
   168  *  For multi-channel audio, the default SDL channel mapping is:
   169  *  2:  FL FR                       (stereo)
   170  *  3:  FL FR LFE                   (2.1 surround)
   171  *  4:  FL FR BL BR                 (quad)
   172  *  5:  FL FR FC BL BR              (quad + center)
   173  *  6:  FL FR FC LFE SL SR          (5.1 surround - last two can also be BL BR)
   174  *  7:  FL FR FC LFE BC SL SR       (6.1 surround)
   175  *  8:  FL FR FC LFE BL BR SL SR    (7.1 surround)
   176  */
   177 typedef struct SDL_AudioSpec
   178 {
   179     int freq;                   /**< DSP frequency -- samples per second */
   180     SDL_AudioFormat format;     /**< Audio data format */
   181     Uint8 channels;             /**< Number of channels: 1 mono, 2 stereo */
   182     Uint8 silence;              /**< Audio buffer silence value (calculated) */
   183     Uint16 samples;             /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */
   184     Uint16 padding;             /**< Necessary for some compile environments */
   185     Uint32 size;                /**< Audio buffer size in bytes (calculated) */
   186     SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
   187     void *userdata;             /**< Userdata passed to callback (ignored for NULL callbacks). */
   188 } SDL_AudioSpec;
   189 
   190 
   191 struct SDL_AudioCVT;
   192 typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
   193                                           SDL_AudioFormat format);
   194 
   195 /**
   196  *  \brief Upper limit of filters in SDL_AudioCVT
   197  *
   198  *  The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is
   199  *  currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers,
   200  *  one of which is the terminating NULL pointer.
   201  */
   202 #define SDL_AUDIOCVT_MAX_FILTERS 9
   203 
   204 /**
   205  *  \struct SDL_AudioCVT
   206  *  \brief A structure to hold a set of audio conversion filters and buffers.
   207  *
   208  *  Note that various parts of the conversion pipeline can take advantage
   209  *  of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
   210  *  you to pass it aligned data, but can possibly run much faster if you
   211  *  set both its (buf) field to a pointer that is aligned to 16 bytes, and its
   212  *  (len) field to something that's a multiple of 16, if possible.
   213  */
   214 #ifdef __GNUC__
   215 /* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
   216    pad it out to 88 bytes to guarantee ABI compatibility between compilers.
   217    vvv
   218    The next time we rev the ABI, make sure to size the ints and add padding.
   219 */
   220 #define SDL_AUDIOCVT_PACKED __attribute__((packed))
   221 #else
   222 #define SDL_AUDIOCVT_PACKED
   223 #endif
   224 /* */
   225 typedef struct SDL_AudioCVT
   226 {
   227     int needed;                 /**< Set to 1 if conversion possible */
   228     SDL_AudioFormat src_format; /**< Source audio format */
   229     SDL_AudioFormat dst_format; /**< Target audio format */
   230     double rate_incr;           /**< Rate conversion increment */
   231     Uint8 *buf;                 /**< Buffer to hold entire audio data */
   232     int len;                    /**< Length of original audio buffer */
   233     int len_cvt;                /**< Length of converted audio buffer */
   234     int len_mult;               /**< buffer must be len*len_mult big */
   235     double len_ratio;           /**< Given len, final size is len*len_ratio */
   236     SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
   237     int filter_index;           /**< Current audio conversion function */
   238 } SDL_AUDIOCVT_PACKED SDL_AudioCVT;
   239 
   240 
   241 /* Function prototypes */
   242 
   243 /**
   244  *  \name Driver discovery functions
   245  *
   246  *  These functions return the list of built in audio drivers, in the
   247  *  order that they are normally initialized by default.
   248  */
   249 /* @{ */
   250 extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
   251 extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
   252 /* @} */
   253 
   254 /**
   255  *  \name Initialization and cleanup
   256  *
   257  *  \internal These functions are used internally, and should not be used unless
   258  *            you have a specific need to specify the audio driver you want to
   259  *            use.  You should normally use SDL_Init() or SDL_InitSubSystem().
   260  */
   261 /* @{ */
   262 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
   263 extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
   264 /* @} */
   265 
   266 /**
   267  *  This function returns the name of the current audio driver, or NULL
   268  *  if no driver has been initialized.
   269  */
   270 extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
   271 
   272 /**
   273  *  This function opens the audio device with the desired parameters, and
   274  *  returns 0 if successful, placing the actual hardware parameters in the
   275  *  structure pointed to by \c obtained.  If \c obtained is NULL, the audio
   276  *  data passed to the callback function will be guaranteed to be in the
   277  *  requested format, and will be automatically converted to the hardware
   278  *  audio format if necessary.  This function returns -1 if it failed
   279  *  to open the audio device, or couldn't set up the audio thread.
   280  *
   281  *  When filling in the desired audio spec structure,
   282  *    - \c desired->freq should be the desired audio frequency in samples-per-
   283  *      second.
   284  *    - \c desired->format should be the desired audio format.
   285  *    - \c desired->samples is the desired size of the audio buffer, in
   286  *      samples.  This number should be a power of two, and may be adjusted by
   287  *      the audio driver to a value more suitable for the hardware.  Good values
   288  *      seem to range between 512 and 8096 inclusive, depending on the
   289  *      application and CPU speed.  Smaller values yield faster response time,
   290  *      but can lead to underflow if the application is doing heavy processing
   291  *      and cannot fill the audio buffer in time.  A stereo sample consists of
   292  *      both right and left channels in LR ordering.
   293  *      Note that the number of samples is directly related to time by the
   294  *      following formula:  \code ms = (samples*1000)/freq \endcode
   295  *    - \c desired->size is the size in bytes of the audio buffer, and is
   296  *      calculated by SDL_OpenAudio().
   297  *    - \c desired->silence is the value used to set the buffer to silence,
   298  *      and is calculated by SDL_OpenAudio().
   299  *    - \c desired->callback should be set to a function that will be called
   300  *      when the audio device is ready for more data.  It is passed a pointer
   301  *      to the audio buffer, and the length in bytes of the audio buffer.
   302  *      This function usually runs in a separate thread, and so you should
   303  *      protect data structures that it accesses by calling SDL_LockAudio()
   304  *      and SDL_UnlockAudio() in your code. Alternately, you may pass a NULL
   305  *      pointer here, and call SDL_QueueAudio() with some frequency, to queue
   306  *      more audio samples to be played (or for capture devices, call
   307  *      SDL_DequeueAudio() with some frequency, to obtain audio samples).
   308  *    - \c desired->userdata is passed as the first parameter to your callback
   309  *      function. If you passed a NULL callback, this value is ignored.
   310  *
   311  *  The audio device starts out playing silence when it's opened, and should
   312  *  be enabled for playing by calling \c SDL_PauseAudio(0) when you are ready
   313  *  for your audio callback function to be called.  Since the audio driver
   314  *  may modify the requested size of the audio buffer, you should allocate
   315  *  any local mixing buffers after you open the audio device.
   316  */
   317 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,
   318                                           SDL_AudioSpec * obtained);
   319 
   320 /**
   321  *  SDL Audio Device IDs.
   322  *
   323  *  A successful call to SDL_OpenAudio() is always device id 1, and legacy
   324  *  SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls
   325  *  always returns devices >= 2 on success. The legacy calls are good both
   326  *  for backwards compatibility and when you don't care about multiple,
   327  *  specific, or capture devices.
   328  */
   329 typedef Uint32 SDL_AudioDeviceID;
   330 
   331 /**
   332  *  Get the number of available devices exposed by the current driver.
   333  *  Only valid after a successfully initializing the audio subsystem.
   334  *  Returns -1 if an explicit list of devices can't be determined; this is
   335  *  not an error. For example, if SDL is set up to talk to a remote audio
   336  *  server, it can't list every one available on the Internet, but it will
   337  *  still allow a specific host to be specified to SDL_OpenAudioDevice().
   338  *
   339  *  In many common cases, when this function returns a value <= 0, it can still
   340  *  successfully open the default device (NULL for first argument of
   341  *  SDL_OpenAudioDevice()).
   342  */
   343 extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
   344 
   345 /**
   346  *  Get the human-readable name of a specific audio device.
   347  *  Must be a value between 0 and (number of audio devices-1).
   348  *  Only valid after a successfully initializing the audio subsystem.
   349  *  The values returned by this function reflect the latest call to
   350  *  SDL_GetNumAudioDevices(); recall that function to redetect available
   351  *  hardware.
   352  *
   353  *  The string returned by this function is UTF-8 encoded, read-only, and
   354  *  managed internally. You are not to free it. If you need to keep the
   355  *  string for any length of time, you should make your own copy of it, as it
   356  *  will be invalid next time any of several other SDL functions is called.
   357  */
   358 extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
   359                                                            int iscapture);
   360 
   361 
   362 /**
   363  *  Open a specific audio device. Passing in a device name of NULL requests
   364  *  the most reasonable default (and is equivalent to calling SDL_OpenAudio()).
   365  *
   366  *  The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but
   367  *  some drivers allow arbitrary and driver-specific strings, such as a
   368  *  hostname/IP address for a remote audio server, or a filename in the
   369  *  diskaudio driver.
   370  *
   371  *  \return 0 on error, a valid device ID that is >= 2 on success.
   372  *
   373  *  SDL_OpenAudio(), unlike this function, always acts on device ID 1.
   374  */
   375 extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char
   376                                                               *device,
   377                                                               int iscapture,
   378                                                               const
   379                                                               SDL_AudioSpec *
   380                                                               desired,
   381                                                               SDL_AudioSpec *
   382                                                               obtained,
   383                                                               int
   384                                                               allowed_changes);
   385 
   386 
   387 
   388 /**
   389  *  \name Audio state
   390  *
   391  *  Get the current audio state.
   392  */
   393 /* @{ */
   394 typedef enum
   395 {
   396     SDL_AUDIO_STOPPED = 0,
   397     SDL_AUDIO_PLAYING,
   398     SDL_AUDIO_PAUSED
   399 } SDL_AudioStatus;
   400 extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
   401 
   402 extern DECLSPEC SDL_AudioStatus SDLCALL
   403 SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
   404 /* @} *//* Audio State */
   405 
   406 /**
   407  *  \name Pause audio functions
   408  *
   409  *  These functions pause and unpause the audio callback processing.
   410  *  They should be called with a parameter of 0 after opening the audio
   411  *  device to start playing sound.  This is so you can safely initialize
   412  *  data for your callback function after opening the audio device.
   413  *  Silence will be written to the audio device during the pause.
   414  */
   415 /* @{ */
   416 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
   417 extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
   418                                                   int pause_on);
   419 /* @} *//* Pause audio functions */
   420 
   421 /**
   422  *  This function loads a WAVE from the data source, automatically freeing
   423  *  that source if \c freesrc is non-zero.  For example, to load a WAVE file,
   424  *  you could do:
   425  *  \code
   426  *      SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
   427  *  \endcode
   428  *
   429  *  If this function succeeds, it returns the given SDL_AudioSpec,
   430  *  filled with the audio data format of the wave data, and sets
   431  *  \c *audio_buf to a malloc()'d buffer containing the audio data,
   432  *  and sets \c *audio_len to the length of that audio buffer, in bytes.
   433  *  You need to free the audio buffer with SDL_FreeWAV() when you are
   434  *  done with it.
   435  *
   436  *  This function returns NULL and sets the SDL error message if the
   437  *  wave file cannot be opened, uses an unknown data format, or is
   438  *  corrupt.  Currently raw and MS-ADPCM WAVE files are supported.
   439  */
   440 extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
   441                                                       int freesrc,
   442                                                       SDL_AudioSpec * spec,
   443                                                       Uint8 ** audio_buf,
   444                                                       Uint32 * audio_len);
   445 
   446 /**
   447  *  Loads a WAV from a file.
   448  *  Compatibility convenience function.
   449  */
   450 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
   451     SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
   452 
   453 /**
   454  *  This function frees data previously allocated with SDL_LoadWAV_RW()
   455  */
   456 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
   457 
   458 /**
   459  *  This function takes a source format and rate and a destination format
   460  *  and rate, and initializes the \c cvt structure with information needed
   461  *  by SDL_ConvertAudio() to convert a buffer of audio data from one format
   462  *  to the other. An unsupported format causes an error and -1 will be returned.
   463  *
   464  *  \return 0 if no conversion is needed, 1 if the audio filter is set up,
   465  *  or -1 on error.
   466  */
   467 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
   468                                               SDL_AudioFormat src_format,
   469                                               Uint8 src_channels,
   470                                               int src_rate,
   471                                               SDL_AudioFormat dst_format,
   472                                               Uint8 dst_channels,
   473                                               int dst_rate);
   474 
   475 /**
   476  *  Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(),
   477  *  created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of
   478  *  audio data in the source format, this function will convert it in-place
   479  *  to the desired format.
   480  *
   481  *  The data conversion may expand the size of the audio data, so the buffer
   482  *  \c cvt->buf should be allocated after the \c cvt structure is initialized by
   483  *  SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long.
   484  *
   485  *  \return 0 on success or -1 if \c cvt->buf is NULL.
   486  */
   487 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
   488 
   489 /* SDL_AudioStream is a new audio conversion interface.
   490    The benefits vs SDL_AudioCVT:
   491     - it can handle resampling data in chunks without generating
   492       artifacts, when it doesn't have the complete buffer available.
   493     - it can handle incoming data in any variable size.
   494     - You push data as you have it, and pull it when you need it
   495  */
   496 /* this is opaque to the outside world. */
   497 struct _SDL_AudioStream;
   498 typedef struct _SDL_AudioStream SDL_AudioStream;
   499 
   500 /**
   501  *  Create a new audio stream
   502  *
   503  *  \param src_format The format of the source audio
   504  *  \param src_channels The number of channels of the source audio
   505  *  \param src_rate The sampling rate of the source audio
   506  *  \param dst_format The format of the desired audio output
   507  *  \param dst_channels The number of channels of the desired audio output
   508  *  \param dst_rate The sampling rate of the desired audio output
   509  *  \return 0 on success, or -1 on error.
   510  *
   511  *  \sa SDL_AudioStreamPut
   512  *  \sa SDL_AudioStreamGet
   513  *  \sa SDL_AudioStreamAvailable
   514  *  \sa SDL_AudioStreamFlush
   515  *  \sa SDL_AudioStreamClear
   516  *  \sa SDL_FreeAudioStream
   517  */
   518 extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
   519                                            const Uint8 src_channels,
   520                                            const int src_rate,
   521                                            const SDL_AudioFormat dst_format,
   522                                            const Uint8 dst_channels,
   523                                            const int dst_rate);
   524 
   525 /**
   526  *  Add data to be converted/resampled to the stream
   527  *
   528  *  \param stream The stream the audio data is being added to
   529  *  \param buf A pointer to the audio data to add
   530  *  \param len The number of bytes to write to the stream
   531  *  \return 0 on success, or -1 on error.
   532  *
   533  *  \sa SDL_NewAudioStream
   534  *  \sa SDL_AudioStreamGet
   535  *  \sa SDL_AudioStreamAvailable
   536  *  \sa SDL_AudioStreamFlush
   537  *  \sa SDL_AudioStreamClear
   538  *  \sa SDL_FreeAudioStream
   539  */
   540 extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
   541 
   542 /**
   543  *  Get converted/resampled data from the stream
   544  *
   545  *  \param stream The stream the audio is being requested from
   546  *  \param buf A buffer to fill with audio data
   547  *  \param len The maximum number of bytes to fill
   548  *  \return The number of bytes read from the stream, or -1 on error
   549  *
   550  *  \sa SDL_NewAudioStream
   551  *  \sa SDL_AudioStreamPut
   552  *  \sa SDL_AudioStreamAvailable
   553  *  \sa SDL_AudioStreamFlush
   554  *  \sa SDL_AudioStreamClear
   555  *  \sa SDL_FreeAudioStream
   556  */
   557 extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
   558 
   559 /**
   560  * Get the number of converted/resampled bytes available. The stream may be
   561  *  buffering data behind the scenes until it has enough to resample
   562  *  correctly, so this number might be lower than what you expect, or even
   563  *  be zero. Add more data or flush the stream if you need the data now.
   564  *
   565  *  \sa SDL_NewAudioStream
   566  *  \sa SDL_AudioStreamPut
   567  *  \sa SDL_AudioStreamGet
   568  *  \sa SDL_AudioStreamFlush
   569  *  \sa SDL_AudioStreamClear
   570  *  \sa SDL_FreeAudioStream
   571  */
   572 extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
   573 
   574 /**
   575  * Tell the stream that you're done sending data, and anything being buffered
   576  *  should be converted/resampled and made available immediately.
   577  *
   578  * It is legal to add more data to a stream after flushing, but there will
   579  *  be audio gaps in the output. Generally this is intended to signal the
   580  *  end of input, so the complete output becomes available.
   581  *
   582  *  \sa SDL_NewAudioStream
   583  *  \sa SDL_AudioStreamPut
   584  *  \sa SDL_AudioStreamGet
   585  *  \sa SDL_AudioStreamAvailable
   586  *  \sa SDL_AudioStreamClear
   587  *  \sa SDL_FreeAudioStream
   588  */
   589 extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);
   590 
   591 /**
   592  *  Clear any pending data in the stream without converting it
   593  *
   594  *  \sa SDL_NewAudioStream
   595  *  \sa SDL_AudioStreamPut
   596  *  \sa SDL_AudioStreamGet
   597  *  \sa SDL_AudioStreamAvailable
   598  *  \sa SDL_AudioStreamFlush
   599  *  \sa SDL_FreeAudioStream
   600  */
   601 extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
   602 
   603 /**
   604  * Free an audio stream
   605  *
   606  *  \sa SDL_NewAudioStream
   607  *  \sa SDL_AudioStreamPut
   608  *  \sa SDL_AudioStreamGet
   609  *  \sa SDL_AudioStreamAvailable
   610  *  \sa SDL_AudioStreamFlush
   611  *  \sa SDL_AudioStreamClear
   612  */
   613 extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
   614 
   615 #define SDL_MIX_MAXVOLUME 128
   616 /**
   617  *  This takes two audio buffers of the playing audio format and mixes
   618  *  them, performing addition, volume adjustment, and overflow clipping.
   619  *  The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME
   620  *  for full audio volume.  Note this does not change hardware volume.
   621  *  This is provided for convenience -- you can mix your own audio data.
   622  */
   623 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
   624                                           Uint32 len, int volume);
   625 
   626 /**
   627  *  This works like SDL_MixAudio(), but you specify the audio format instead of
   628  *  using the format of audio device 1. Thus it can be used when no audio
   629  *  device is open at all.
   630  */
   631 extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
   632                                                 const Uint8 * src,
   633                                                 SDL_AudioFormat format,
   634                                                 Uint32 len, int volume);
   635 
   636 /**
   637  *  Queue more audio on non-callback devices.
   638  *
   639  *  (If you are looking to retrieve queued audio from a non-callback capture
   640  *  device, you want SDL_DequeueAudio() instead. This will return -1 to
   641  *  signify an error if you use it with capture devices.)
   642  *
   643  *  SDL offers two ways to feed audio to the device: you can either supply a
   644  *  callback that SDL triggers with some frequency to obtain more audio
   645  *  (pull method), or you can supply no callback, and then SDL will expect
   646  *  you to supply data at regular intervals (push method) with this function.
   647  *
   648  *  There are no limits on the amount of data you can queue, short of
   649  *  exhaustion of address space. Queued data will drain to the device as
   650  *  necessary without further intervention from you. If the device needs
   651  *  audio but there is not enough queued, it will play silence to make up
   652  *  the difference. This means you will have skips in your audio playback
   653  *  if you aren't routinely queueing sufficient data.
   654  *
   655  *  This function copies the supplied data, so you are safe to free it when
   656  *  the function returns. This function is thread-safe, but queueing to the
   657  *  same device from two threads at once does not promise which buffer will
   658  *  be queued first.
   659  *
   660  *  You may not queue audio on a device that is using an application-supplied
   661  *  callback; doing so returns an error. You have to use the audio callback
   662  *  or queue audio with this function, but not both.
   663  *
   664  *  You should not call SDL_LockAudio() on the device before queueing; SDL
   665  *  handles locking internally for this function.
   666  *
   667  *  \param dev The device ID to which we will queue audio.
   668  *  \param data The data to queue to the device for later playback.
   669  *  \param len The number of bytes (not samples!) to which (data) points.
   670  *  \return 0 on success, or -1 on error.
   671  *
   672  *  \sa SDL_GetQueuedAudioSize
   673  *  \sa SDL_ClearQueuedAudio
   674  */
   675 extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
   676 
   677 /**
   678  *  Dequeue more audio on non-callback devices.
   679  *
   680  *  (If you are looking to queue audio for output on a non-callback playback
   681  *  device, you want SDL_QueueAudio() instead. This will always return 0
   682  *  if you use it with playback devices.)
   683  *
   684  *  SDL offers two ways to retrieve audio from a capture device: you can
   685  *  either supply a callback that SDL triggers with some frequency as the
   686  *  device records more audio data, (push method), or you can supply no
   687  *  callback, and then SDL will expect you to retrieve data at regular
   688  *  intervals (pull method) with this function.
   689  *
   690  *  There are no limits on the amount of data you can queue, short of
   691  *  exhaustion of address space. Data from the device will keep queuing as
   692  *  necessary without further intervention from you. This means you will
   693  *  eventually run out of memory if you aren't routinely dequeueing data.
   694  *
   695  *  Capture devices will not queue data when paused; if you are expecting
   696  *  to not need captured audio for some length of time, use
   697  *  SDL_PauseAudioDevice() to stop the capture device from queueing more
   698  *  data. This can be useful during, say, level loading times. When
   699  *  unpaused, capture devices will start queueing data from that point,
   700  *  having flushed any capturable data available while paused.
   701  *
   702  *  This function is thread-safe, but dequeueing from the same device from
   703  *  two threads at once does not promise which thread will dequeued data
   704  *  first.
   705  *
   706  *  You may not dequeue audio from a device that is using an
   707  *  application-supplied callback; doing so returns an error. You have to use
   708  *  the audio callback, or dequeue audio with this function, but not both.
   709  *
   710  *  You should not call SDL_LockAudio() on the device before queueing; SDL
   711  *  handles locking internally for this function.
   712  *
   713  *  \param dev The device ID from which we will dequeue audio.
   714  *  \param data A pointer into where audio data should be copied.
   715  *  \param len The number of bytes (not samples!) to which (data) points.
   716  *  \return number of bytes dequeued, which could be less than requested.
   717  *
   718  *  \sa SDL_GetQueuedAudioSize
   719  *  \sa SDL_ClearQueuedAudio
   720  */
   721 extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
   722 
   723 /**
   724  *  Get the number of bytes of still-queued audio.
   725  *
   726  *  For playback device:
   727  *
   728  *    This is the number of bytes that have been queued for playback with
   729  *    SDL_QueueAudio(), but have not yet been sent to the hardware. This
   730  *    number may shrink at any time, so this only informs of pending data.
   731  *
   732  *    Once we've sent it to the hardware, this function can not decide the
   733  *    exact byte boundary of what has been played. It's possible that we just
   734  *    gave the hardware several kilobytes right before you called this
   735  *    function, but it hasn't played any of it yet, or maybe half of it, etc.
   736  *
   737  *  For capture devices:
   738  *
   739  *    This is the number of bytes that have been captured by the device and
   740  *    are waiting for you to dequeue. This number may grow at any time, so
   741  *    this only informs of the lower-bound of available data.
   742  *
   743  *  You may not queue audio on a device that is using an application-supplied
   744  *  callback; calling this function on such a device always returns 0.
   745  *  You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
   746  *  the audio callback, but not both.
   747  *
   748  *  You should not call SDL_LockAudio() on the device before querying; SDL
   749  *  handles locking internally for this function.
   750  *
   751  *  \param dev The device ID of which we will query queued audio size.
   752  *  \return Number of bytes (not samples!) of queued audio.
   753  *
   754  *  \sa SDL_QueueAudio
   755  *  \sa SDL_ClearQueuedAudio
   756  */
   757 extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
   758 
   759 /**
   760  *  Drop any queued audio data. For playback devices, this is any queued data
   761  *  still waiting to be submitted to the hardware. For capture devices, this
   762  *  is any data that was queued by the device that hasn't yet been dequeued by
   763  *  the application.
   764  *
   765  *  Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
   766  *  playback devices, the hardware will start playing silence if more audio
   767  *  isn't queued. Unpaused capture devices will start filling the queue again
   768  *  as soon as they have more data available (which, depending on the state
   769  *  of the hardware and the thread, could be before this function call
   770  *  returns!).
   771  *
   772  *  This will not prevent playback of queued audio that's already been sent
   773  *  to the hardware, as we can not undo that, so expect there to be some
   774  *  fraction of a second of audio that might still be heard. This can be
   775  *  useful if you want to, say, drop any pending music during a level change
   776  *  in your game.
   777  *
   778  *  You may not queue audio on a device that is using an application-supplied
   779  *  callback; calling this function on such a device is always a no-op.
   780  *  You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
   781  *  the audio callback, but not both.
   782  *
   783  *  You should not call SDL_LockAudio() on the device before clearing the
   784  *  queue; SDL handles locking internally for this function.
   785  *
   786  *  This function always succeeds and thus returns void.
   787  *
   788  *  \param dev The device ID of which to clear the audio queue.
   789  *
   790  *  \sa SDL_QueueAudio
   791  *  \sa SDL_GetQueuedAudioSize
   792  */
   793 extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);
   794 
   795 
   796 /**
   797  *  \name Audio lock functions
   798  *
   799  *  The lock manipulated by these functions protects the callback function.
   800  *  During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that
   801  *  the callback function is not running.  Do not call these from the callback
   802  *  function or you will cause deadlock.
   803  */
   804 /* @{ */
   805 extern DECLSPEC void SDLCALL SDL_LockAudio(void);
   806 extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
   807 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
   808 extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
   809 /* @} *//* Audio lock functions */
   810 
   811 /**
   812  *  This function shuts down audio processing and closes the audio device.
   813  */
   814 extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
   815 extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
   816 
   817 /* Ends C function definitions when using C++ */
   818 #ifdef __cplusplus
   819 }
   820 #endif
   821 #include "close_code.h"
   822 
   823 #endif /* SDL_audio_h_ */
   824 
   825 /* vi: set ts=4 sw=4 expandtab: */