include/SDL_audio.h
author Sam Lantinga <slouken@libsdl.org>
Sat, 19 Oct 2019 01:54:02 -0700
changeset 13150 29acbbbb41b2
parent 12806 b06fa7da012b
permissions -rw-r--r--
Don't try to use the Xbox HID protocol with the NVIDIA Shield controllers
     1 /*
     2   Simple DirectMedia Layer
     3   Copyright (C) 1997-2019 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_SAMPLES_CHANGE      0x00000008
   144 #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)
   145 /* @} */
   146 
   147 /* @} *//* Audio flags */
   148 
   149 /**
   150  *  This function is called when the audio device needs more data.
   151  *
   152  *  \param userdata An application-specific parameter saved in
   153  *                  the SDL_AudioSpec structure
   154  *  \param stream A pointer to the audio data buffer.
   155  *  \param len    The length of that buffer in bytes.
   156  *
   157  *  Once the callback returns, the buffer will no longer be valid.
   158  *  Stereo samples are stored in a LRLRLR ordering.
   159  *
   160  *  You can choose to avoid callbacks and use SDL_QueueAudio() instead, if
   161  *  you like. Just open your audio device with a NULL callback.
   162  */
   163 typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
   164                                             int len);
   165 
   166 /**
   167  *  The calculated values in this structure are calculated by SDL_OpenAudio().
   168  *
   169  *  For multi-channel audio, the default SDL channel mapping is:
   170  *  2:  FL FR                       (stereo)
   171  *  3:  FL FR LFE                   (2.1 surround)
   172  *  4:  FL FR BL BR                 (quad)
   173  *  5:  FL FR FC BL BR              (quad + center)
   174  *  6:  FL FR FC LFE SL SR          (5.1 surround - last two can also be BL BR)
   175  *  7:  FL FR FC LFE BC SL SR       (6.1 surround)
   176  *  8:  FL FR FC LFE BL BR SL SR    (7.1 surround)
   177  */
   178 typedef struct SDL_AudioSpec
   179 {
   180     int freq;                   /**< DSP frequency -- samples per second */
   181     SDL_AudioFormat format;     /**< Audio data format */
   182     Uint8 channels;             /**< Number of channels: 1 mono, 2 stereo */
   183     Uint8 silence;              /**< Audio buffer silence value (calculated) */
   184     Uint16 samples;             /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */
   185     Uint16 padding;             /**< Necessary for some compile environments */
   186     Uint32 size;                /**< Audio buffer size in bytes (calculated) */
   187     SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
   188     void *userdata;             /**< Userdata passed to callback (ignored for NULL callbacks). */
   189 } SDL_AudioSpec;
   190 
   191 
   192 struct SDL_AudioCVT;
   193 typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
   194                                           SDL_AudioFormat format);
   195 
   196 /**
   197  *  \brief Upper limit of filters in SDL_AudioCVT
   198  *
   199  *  The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is
   200  *  currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers,
   201  *  one of which is the terminating NULL pointer.
   202  */
   203 #define SDL_AUDIOCVT_MAX_FILTERS 9
   204 
   205 /**
   206  *  \struct SDL_AudioCVT
   207  *  \brief A structure to hold a set of audio conversion filters and buffers.
   208  *
   209  *  Note that various parts of the conversion pipeline can take advantage
   210  *  of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
   211  *  you to pass it aligned data, but can possibly run much faster if you
   212  *  set both its (buf) field to a pointer that is aligned to 16 bytes, and its
   213  *  (len) field to something that's a multiple of 16, if possible.
   214  */
   215 #ifdef __GNUC__
   216 /* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
   217    pad it out to 88 bytes to guarantee ABI compatibility between compilers.
   218    vvv
   219    The next time we rev the ABI, make sure to size the ints and add padding.
   220 */
   221 #define SDL_AUDIOCVT_PACKED __attribute__((packed))
   222 #else
   223 #define SDL_AUDIOCVT_PACKED
   224 #endif
   225 /* */
   226 typedef struct SDL_AudioCVT
   227 {
   228     int needed;                 /**< Set to 1 if conversion possible */
   229     SDL_AudioFormat src_format; /**< Source audio format */
   230     SDL_AudioFormat dst_format; /**< Target audio format */
   231     double rate_incr;           /**< Rate conversion increment */
   232     Uint8 *buf;                 /**< Buffer to hold entire audio data */
   233     int len;                    /**< Length of original audio buffer */
   234     int len_cvt;                /**< Length of converted audio buffer */
   235     int len_mult;               /**< buffer must be len*len_mult big */
   236     double len_ratio;           /**< Given len, final size is len*len_ratio */
   237     SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
   238     int filter_index;           /**< Current audio conversion function */
   239 } SDL_AUDIOCVT_PACKED SDL_AudioCVT;
   240 
   241 
   242 /* Function prototypes */
   243 
   244 /**
   245  *  \name Driver discovery functions
   246  *
   247  *  These functions return the list of built in audio drivers, in the
   248  *  order that they are normally initialized by default.
   249  */
   250 /* @{ */
   251 extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
   252 extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
   253 /* @} */
   254 
   255 /**
   256  *  \name Initialization and cleanup
   257  *
   258  *  \internal These functions are used internally, and should not be used unless
   259  *            you have a specific need to specify the audio driver you want to
   260  *            use.  You should normally use SDL_Init() or SDL_InitSubSystem().
   261  */
   262 /* @{ */
   263 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
   264 extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
   265 /* @} */
   266 
   267 /**
   268  *  This function returns the name of the current audio driver, or NULL
   269  *  if no driver has been initialized.
   270  */
   271 extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
   272 
   273 /**
   274  *  This function opens the audio device with the desired parameters, and
   275  *  returns 0 if successful, placing the actual hardware parameters in the
   276  *  structure pointed to by \c obtained.  If \c obtained is NULL, the audio
   277  *  data passed to the callback function will be guaranteed to be in the
   278  *  requested format, and will be automatically converted to the hardware
   279  *  audio format if necessary.  This function returns -1 if it failed
   280  *  to open the audio device, or couldn't set up the audio thread.
   281  *
   282  *  When filling in the desired audio spec structure,
   283  *    - \c desired->freq should be the desired audio frequency in samples-per-
   284  *      second.
   285  *    - \c desired->format should be the desired audio format.
   286  *    - \c desired->samples is the desired size of the audio buffer, in
   287  *      samples.  This number should be a power of two, and may be adjusted by
   288  *      the audio driver to a value more suitable for the hardware.  Good values
   289  *      seem to range between 512 and 8096 inclusive, depending on the
   290  *      application and CPU speed.  Smaller values yield faster response time,
   291  *      but can lead to underflow if the application is doing heavy processing
   292  *      and cannot fill the audio buffer in time.  A stereo sample consists of
   293  *      both right and left channels in LR ordering.
   294  *      Note that the number of samples is directly related to time by the
   295  *      following formula:  \code ms = (samples*1000)/freq \endcode
   296  *    - \c desired->size is the size in bytes of the audio buffer, and is
   297  *      calculated by SDL_OpenAudio().
   298  *    - \c desired->silence is the value used to set the buffer to silence,
   299  *      and is calculated by SDL_OpenAudio().
   300  *    - \c desired->callback should be set to a function that will be called
   301  *      when the audio device is ready for more data.  It is passed a pointer
   302  *      to the audio buffer, and the length in bytes of the audio buffer.
   303  *      This function usually runs in a separate thread, and so you should
   304  *      protect data structures that it accesses by calling SDL_LockAudio()
   305  *      and SDL_UnlockAudio() in your code. Alternately, you may pass a NULL
   306  *      pointer here, and call SDL_QueueAudio() with some frequency, to queue
   307  *      more audio samples to be played (or for capture devices, call
   308  *      SDL_DequeueAudio() with some frequency, to obtain audio samples).
   309  *    - \c desired->userdata is passed as the first parameter to your callback
   310  *      function. If you passed a NULL callback, this value is ignored.
   311  *
   312  *  The audio device starts out playing silence when it's opened, and should
   313  *  be enabled for playing by calling \c SDL_PauseAudio(0) when you are ready
   314  *  for your audio callback function to be called.  Since the audio driver
   315  *  may modify the requested size of the audio buffer, you should allocate
   316  *  any local mixing buffers after you open the audio device.
   317  */
   318 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,
   319                                           SDL_AudioSpec * obtained);
   320 
   321 /**
   322  *  SDL Audio Device IDs.
   323  *
   324  *  A successful call to SDL_OpenAudio() is always device id 1, and legacy
   325  *  SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls
   326  *  always returns devices >= 2 on success. The legacy calls are good both
   327  *  for backwards compatibility and when you don't care about multiple,
   328  *  specific, or capture devices.
   329  */
   330 typedef Uint32 SDL_AudioDeviceID;
   331 
   332 /**
   333  *  Get the number of available devices exposed by the current driver.
   334  *  Only valid after a successfully initializing the audio subsystem.
   335  *  Returns -1 if an explicit list of devices can't be determined; this is
   336  *  not an error. For example, if SDL is set up to talk to a remote audio
   337  *  server, it can't list every one available on the Internet, but it will
   338  *  still allow a specific host to be specified to SDL_OpenAudioDevice().
   339  *
   340  *  In many common cases, when this function returns a value <= 0, it can still
   341  *  successfully open the default device (NULL for first argument of
   342  *  SDL_OpenAudioDevice()).
   343  */
   344 extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
   345 
   346 /**
   347  *  Get the human-readable name of a specific audio device.
   348  *  Must be a value between 0 and (number of audio devices-1).
   349  *  Only valid after a successfully initializing the audio subsystem.
   350  *  The values returned by this function reflect the latest call to
   351  *  SDL_GetNumAudioDevices(); recall that function to redetect available
   352  *  hardware.
   353  *
   354  *  The string returned by this function is UTF-8 encoded, read-only, and
   355  *  managed internally. You are not to free it. If you need to keep the
   356  *  string for any length of time, you should make your own copy of it, as it
   357  *  will be invalid next time any of several other SDL functions is called.
   358  */
   359 extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
   360                                                            int iscapture);
   361 
   362 
   363 /**
   364  *  Open a specific audio device. Passing in a device name of NULL requests
   365  *  the most reasonable default (and is equivalent to calling SDL_OpenAudio()).
   366  *
   367  *  The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but
   368  *  some drivers allow arbitrary and driver-specific strings, such as a
   369  *  hostname/IP address for a remote audio server, or a filename in the
   370  *  diskaudio driver.
   371  *
   372  *  \return 0 on error, a valid device ID that is >= 2 on success.
   373  *
   374  *  SDL_OpenAudio(), unlike this function, always acts on device ID 1.
   375  */
   376 extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char
   377                                                               *device,
   378                                                               int iscapture,
   379                                                               const
   380                                                               SDL_AudioSpec *
   381                                                               desired,
   382                                                               SDL_AudioSpec *
   383                                                               obtained,
   384                                                               int
   385                                                               allowed_changes);
   386 
   387 
   388 
   389 /**
   390  *  \name Audio state
   391  *
   392  *  Get the current audio state.
   393  */
   394 /* @{ */
   395 typedef enum
   396 {
   397     SDL_AUDIO_STOPPED = 0,
   398     SDL_AUDIO_PLAYING,
   399     SDL_AUDIO_PAUSED
   400 } SDL_AudioStatus;
   401 extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
   402 
   403 extern DECLSPEC SDL_AudioStatus SDLCALL
   404 SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
   405 /* @} *//* Audio State */
   406 
   407 /**
   408  *  \name Pause audio functions
   409  *
   410  *  These functions pause and unpause the audio callback processing.
   411  *  They should be called with a parameter of 0 after opening the audio
   412  *  device to start playing sound.  This is so you can safely initialize
   413  *  data for your callback function after opening the audio device.
   414  *  Silence will be written to the audio device during the pause.
   415  */
   416 /* @{ */
   417 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
   418 extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
   419                                                   int pause_on);
   420 /* @} *//* Pause audio functions */
   421 
   422 /**
   423  *  \brief Load the audio data of a WAVE file into memory
   424  *
   425  *  Loading a WAVE file requires \c src, \c spec, \c audio_buf and \c audio_len
   426  *  to be valid pointers. The entire data portion of the file is then loaded
   427  *  into memory and decoded if necessary.
   428  *
   429  *  If \c freesrc is non-zero, the data source gets automatically closed and
   430  *  freed before the function returns.
   431  *
   432  *  Supported are RIFF WAVE files with the formats PCM (8, 16, 24, and 32 bits),
   433  *  IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and A-law and
   434  *  ยต-law (8 bits). Other formats are currently unsupported and cause an error.
   435  *
   436  *  If this function succeeds, the pointer returned by it is equal to \c spec
   437  *  and the pointer to the audio data allocated by the function is written to
   438  *  \c audio_buf and its length in bytes to \c audio_len. The \ref SDL_AudioSpec
   439  *  members \c freq, \c channels, and \c format are set to the values of the
   440  *  audio data in the buffer. The \c samples member is set to a sane default and
   441  *  all others are set to zero.
   442  *
   443  *  It's necessary to use SDL_FreeWAV() to free the audio data returned in
   444  *  \c audio_buf when it is no longer used.
   445  *
   446  *  Because of the underspecification of the Waveform format, there are many
   447  *  problematic files in the wild that cause issues with strict decoders. To
   448  *  provide compatibility with these files, this decoder is lenient in regards
   449  *  to the truncation of the file, the fact chunk, and the size of the RIFF
   450  *  chunk. The hints SDL_HINT_WAVE_RIFF_CHUNK_SIZE, SDL_HINT_WAVE_TRUNCATION,
   451  *  and SDL_HINT_WAVE_FACT_CHUNK can be used to tune the behavior of the
   452  *  loading process.
   453  *
   454  *  Any file that is invalid (due to truncation, corruption, or wrong values in
   455  *  the headers), too big, or unsupported causes an error. Additionally, any
   456  *  critical I/O error from the data source will terminate the loading process
   457  *  with an error. The function returns NULL on error and in all cases (with the
   458  *  exception of \c src being NULL), an appropriate error message will be set.
   459  *
   460  *  It is required that the data source supports seeking.
   461  *
   462  *  Example:
   463  *  \code
   464  *      SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
   465  *  \endcode
   466  *
   467  *  \param src The data source with the WAVE data
   468  *  \param freesrc A integer value that makes the function close the data source if non-zero
   469  *  \param spec A pointer filled with the audio format of the audio data
   470  *  \param audio_buf A pointer filled with the audio data allocated by the function
   471  *  \param audio_len A pointer filled with the length of the audio data buffer in bytes
   472  *  \return NULL on error, or non-NULL on success.
   473  */
   474 extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
   475                                                       int freesrc,
   476                                                       SDL_AudioSpec * spec,
   477                                                       Uint8 ** audio_buf,
   478                                                       Uint32 * audio_len);
   479 
   480 /**
   481  *  Loads a WAV from a file.
   482  *  Compatibility convenience function.
   483  */
   484 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
   485     SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
   486 
   487 /**
   488  *  This function frees data previously allocated with SDL_LoadWAV_RW()
   489  */
   490 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
   491 
   492 /**
   493  *  This function takes a source format and rate and a destination format
   494  *  and rate, and initializes the \c cvt structure with information needed
   495  *  by SDL_ConvertAudio() to convert a buffer of audio data from one format
   496  *  to the other. An unsupported format causes an error and -1 will be returned.
   497  *
   498  *  \return 0 if no conversion is needed, 1 if the audio filter is set up,
   499  *  or -1 on error.
   500  */
   501 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
   502                                               SDL_AudioFormat src_format,
   503                                               Uint8 src_channels,
   504                                               int src_rate,
   505                                               SDL_AudioFormat dst_format,
   506                                               Uint8 dst_channels,
   507                                               int dst_rate);
   508 
   509 /**
   510  *  Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(),
   511  *  created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of
   512  *  audio data in the source format, this function will convert it in-place
   513  *  to the desired format.
   514  *
   515  *  The data conversion may expand the size of the audio data, so the buffer
   516  *  \c cvt->buf should be allocated after the \c cvt structure is initialized by
   517  *  SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long.
   518  *
   519  *  \return 0 on success or -1 if \c cvt->buf is NULL.
   520  */
   521 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
   522 
   523 /* SDL_AudioStream is a new audio conversion interface.
   524    The benefits vs SDL_AudioCVT:
   525     - it can handle resampling data in chunks without generating
   526       artifacts, when it doesn't have the complete buffer available.
   527     - it can handle incoming data in any variable size.
   528     - You push data as you have it, and pull it when you need it
   529  */
   530 /* this is opaque to the outside world. */
   531 struct _SDL_AudioStream;
   532 typedef struct _SDL_AudioStream SDL_AudioStream;
   533 
   534 /**
   535  *  Create a new audio stream
   536  *
   537  *  \param src_format The format of the source audio
   538  *  \param src_channels The number of channels of the source audio
   539  *  \param src_rate The sampling rate of the source audio
   540  *  \param dst_format The format of the desired audio output
   541  *  \param dst_channels The number of channels of the desired audio output
   542  *  \param dst_rate The sampling rate of the desired audio output
   543  *  \return 0 on success, or -1 on error.
   544  *
   545  *  \sa SDL_AudioStreamPut
   546  *  \sa SDL_AudioStreamGet
   547  *  \sa SDL_AudioStreamAvailable
   548  *  \sa SDL_AudioStreamFlush
   549  *  \sa SDL_AudioStreamClear
   550  *  \sa SDL_FreeAudioStream
   551  */
   552 extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
   553                                            const Uint8 src_channels,
   554                                            const int src_rate,
   555                                            const SDL_AudioFormat dst_format,
   556                                            const Uint8 dst_channels,
   557                                            const int dst_rate);
   558 
   559 /**
   560  *  Add data to be converted/resampled to the stream
   561  *
   562  *  \param stream The stream the audio data is being added to
   563  *  \param buf A pointer to the audio data to add
   564  *  \param len The number of bytes to write to the stream
   565  *  \return 0 on success, or -1 on error.
   566  *
   567  *  \sa SDL_NewAudioStream
   568  *  \sa SDL_AudioStreamGet
   569  *  \sa SDL_AudioStreamAvailable
   570  *  \sa SDL_AudioStreamFlush
   571  *  \sa SDL_AudioStreamClear
   572  *  \sa SDL_FreeAudioStream
   573  */
   574 extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
   575 
   576 /**
   577  *  Get converted/resampled data from the stream
   578  *
   579  *  \param stream The stream the audio is being requested from
   580  *  \param buf A buffer to fill with audio data
   581  *  \param len The maximum number of bytes to fill
   582  *  \return The number of bytes read from the stream, or -1 on error
   583  *
   584  *  \sa SDL_NewAudioStream
   585  *  \sa SDL_AudioStreamPut
   586  *  \sa SDL_AudioStreamAvailable
   587  *  \sa SDL_AudioStreamFlush
   588  *  \sa SDL_AudioStreamClear
   589  *  \sa SDL_FreeAudioStream
   590  */
   591 extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
   592 
   593 /**
   594  * Get the number of converted/resampled bytes available. The stream may be
   595  *  buffering data behind the scenes until it has enough to resample
   596  *  correctly, so this number might be lower than what you expect, or even
   597  *  be zero. Add more data or flush the stream if you need the data now.
   598  *
   599  *  \sa SDL_NewAudioStream
   600  *  \sa SDL_AudioStreamPut
   601  *  \sa SDL_AudioStreamGet
   602  *  \sa SDL_AudioStreamFlush
   603  *  \sa SDL_AudioStreamClear
   604  *  \sa SDL_FreeAudioStream
   605  */
   606 extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
   607 
   608 /**
   609  * Tell the stream that you're done sending data, and anything being buffered
   610  *  should be converted/resampled and made available immediately.
   611  *
   612  * It is legal to add more data to a stream after flushing, but there will
   613  *  be audio gaps in the output. Generally this is intended to signal the
   614  *  end of input, so the complete output becomes available.
   615  *
   616  *  \sa SDL_NewAudioStream
   617  *  \sa SDL_AudioStreamPut
   618  *  \sa SDL_AudioStreamGet
   619  *  \sa SDL_AudioStreamAvailable
   620  *  \sa SDL_AudioStreamClear
   621  *  \sa SDL_FreeAudioStream
   622  */
   623 extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);
   624 
   625 /**
   626  *  Clear any pending data in the stream without converting it
   627  *
   628  *  \sa SDL_NewAudioStream
   629  *  \sa SDL_AudioStreamPut
   630  *  \sa SDL_AudioStreamGet
   631  *  \sa SDL_AudioStreamAvailable
   632  *  \sa SDL_AudioStreamFlush
   633  *  \sa SDL_FreeAudioStream
   634  */
   635 extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
   636 
   637 /**
   638  * Free an audio stream
   639  *
   640  *  \sa SDL_NewAudioStream
   641  *  \sa SDL_AudioStreamPut
   642  *  \sa SDL_AudioStreamGet
   643  *  \sa SDL_AudioStreamAvailable
   644  *  \sa SDL_AudioStreamFlush
   645  *  \sa SDL_AudioStreamClear
   646  */
   647 extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
   648 
   649 #define SDL_MIX_MAXVOLUME 128
   650 /**
   651  *  This takes two audio buffers of the playing audio format and mixes
   652  *  them, performing addition, volume adjustment, and overflow clipping.
   653  *  The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME
   654  *  for full audio volume.  Note this does not change hardware volume.
   655  *  This is provided for convenience -- you can mix your own audio data.
   656  */
   657 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
   658                                           Uint32 len, int volume);
   659 
   660 /**
   661  *  This works like SDL_MixAudio(), but you specify the audio format instead of
   662  *  using the format of audio device 1. Thus it can be used when no audio
   663  *  device is open at all.
   664  */
   665 extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
   666                                                 const Uint8 * src,
   667                                                 SDL_AudioFormat format,
   668                                                 Uint32 len, int volume);
   669 
   670 /**
   671  *  Queue more audio on non-callback devices.
   672  *
   673  *  (If you are looking to retrieve queued audio from a non-callback capture
   674  *  device, you want SDL_DequeueAudio() instead. This will return -1 to
   675  *  signify an error if you use it with capture devices.)
   676  *
   677  *  SDL offers two ways to feed audio to the device: you can either supply a
   678  *  callback that SDL triggers with some frequency to obtain more audio
   679  *  (pull method), or you can supply no callback, and then SDL will expect
   680  *  you to supply data at regular intervals (push method) with this function.
   681  *
   682  *  There are no limits on the amount of data you can queue, short of
   683  *  exhaustion of address space. Queued data will drain to the device as
   684  *  necessary without further intervention from you. If the device needs
   685  *  audio but there is not enough queued, it will play silence to make up
   686  *  the difference. This means you will have skips in your audio playback
   687  *  if you aren't routinely queueing sufficient data.
   688  *
   689  *  This function copies the supplied data, so you are safe to free it when
   690  *  the function returns. This function is thread-safe, but queueing to the
   691  *  same device from two threads at once does not promise which buffer will
   692  *  be queued first.
   693  *
   694  *  You may not queue audio on a device that is using an application-supplied
   695  *  callback; doing so returns an error. You have to use the audio callback
   696  *  or queue audio with this function, but not both.
   697  *
   698  *  You should not call SDL_LockAudio() on the device before queueing; SDL
   699  *  handles locking internally for this function.
   700  *
   701  *  \param dev The device ID to which we will queue audio.
   702  *  \param data The data to queue to the device for later playback.
   703  *  \param len The number of bytes (not samples!) to which (data) points.
   704  *  \return 0 on success, or -1 on error.
   705  *
   706  *  \sa SDL_GetQueuedAudioSize
   707  *  \sa SDL_ClearQueuedAudio
   708  */
   709 extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
   710 
   711 /**
   712  *  Dequeue more audio on non-callback devices.
   713  *
   714  *  (If you are looking to queue audio for output on a non-callback playback
   715  *  device, you want SDL_QueueAudio() instead. This will always return 0
   716  *  if you use it with playback devices.)
   717  *
   718  *  SDL offers two ways to retrieve audio from a capture device: you can
   719  *  either supply a callback that SDL triggers with some frequency as the
   720  *  device records more audio data, (push method), or you can supply no
   721  *  callback, and then SDL will expect you to retrieve data at regular
   722  *  intervals (pull method) with this function.
   723  *
   724  *  There are no limits on the amount of data you can queue, short of
   725  *  exhaustion of address space. Data from the device will keep queuing as
   726  *  necessary without further intervention from you. This means you will
   727  *  eventually run out of memory if you aren't routinely dequeueing data.
   728  *
   729  *  Capture devices will not queue data when paused; if you are expecting
   730  *  to not need captured audio for some length of time, use
   731  *  SDL_PauseAudioDevice() to stop the capture device from queueing more
   732  *  data. This can be useful during, say, level loading times. When
   733  *  unpaused, capture devices will start queueing data from that point,
   734  *  having flushed any capturable data available while paused.
   735  *
   736  *  This function is thread-safe, but dequeueing from the same device from
   737  *  two threads at once does not promise which thread will dequeued data
   738  *  first.
   739  *
   740  *  You may not dequeue audio from a device that is using an
   741  *  application-supplied callback; doing so returns an error. You have to use
   742  *  the audio callback, or dequeue audio with this function, but not both.
   743  *
   744  *  You should not call SDL_LockAudio() on the device before queueing; SDL
   745  *  handles locking internally for this function.
   746  *
   747  *  \param dev The device ID from which we will dequeue audio.
   748  *  \param data A pointer into where audio data should be copied.
   749  *  \param len The number of bytes (not samples!) to which (data) points.
   750  *  \return number of bytes dequeued, which could be less than requested.
   751  *
   752  *  \sa SDL_GetQueuedAudioSize
   753  *  \sa SDL_ClearQueuedAudio
   754  */
   755 extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
   756 
   757 /**
   758  *  Get the number of bytes of still-queued audio.
   759  *
   760  *  For playback device:
   761  *
   762  *    This is the number of bytes that have been queued for playback with
   763  *    SDL_QueueAudio(), but have not yet been sent to the hardware. This
   764  *    number may shrink at any time, so this only informs of pending data.
   765  *
   766  *    Once we've sent it to the hardware, this function can not decide the
   767  *    exact byte boundary of what has been played. It's possible that we just
   768  *    gave the hardware several kilobytes right before you called this
   769  *    function, but it hasn't played any of it yet, or maybe half of it, etc.
   770  *
   771  *  For capture devices:
   772  *
   773  *    This is the number of bytes that have been captured by the device and
   774  *    are waiting for you to dequeue. This number may grow at any time, so
   775  *    this only informs of the lower-bound of available data.
   776  *
   777  *  You may not queue audio on a device that is using an application-supplied
   778  *  callback; calling this function on such a device always returns 0.
   779  *  You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
   780  *  the audio callback, but not both.
   781  *
   782  *  You should not call SDL_LockAudio() on the device before querying; SDL
   783  *  handles locking internally for this function.
   784  *
   785  *  \param dev The device ID of which we will query queued audio size.
   786  *  \return Number of bytes (not samples!) of queued audio.
   787  *
   788  *  \sa SDL_QueueAudio
   789  *  \sa SDL_ClearQueuedAudio
   790  */
   791 extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
   792 
   793 /**
   794  *  Drop any queued audio data. For playback devices, this is any queued data
   795  *  still waiting to be submitted to the hardware. For capture devices, this
   796  *  is any data that was queued by the device that hasn't yet been dequeued by
   797  *  the application.
   798  *
   799  *  Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
   800  *  playback devices, the hardware will start playing silence if more audio
   801  *  isn't queued. Unpaused capture devices will start filling the queue again
   802  *  as soon as they have more data available (which, depending on the state
   803  *  of the hardware and the thread, could be before this function call
   804  *  returns!).
   805  *
   806  *  This will not prevent playback of queued audio that's already been sent
   807  *  to the hardware, as we can not undo that, so expect there to be some
   808  *  fraction of a second of audio that might still be heard. This can be
   809  *  useful if you want to, say, drop any pending music during a level change
   810  *  in your game.
   811  *
   812  *  You may not queue audio on a device that is using an application-supplied
   813  *  callback; calling this function on such a device is always a no-op.
   814  *  You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
   815  *  the audio callback, but not both.
   816  *
   817  *  You should not call SDL_LockAudio() on the device before clearing the
   818  *  queue; SDL handles locking internally for this function.
   819  *
   820  *  This function always succeeds and thus returns void.
   821  *
   822  *  \param dev The device ID of which to clear the audio queue.
   823  *
   824  *  \sa SDL_QueueAudio
   825  *  \sa SDL_GetQueuedAudioSize
   826  */
   827 extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);
   828 
   829 
   830 /**
   831  *  \name Audio lock functions
   832  *
   833  *  The lock manipulated by these functions protects the callback function.
   834  *  During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that
   835  *  the callback function is not running.  Do not call these from the callback
   836  *  function or you will cause deadlock.
   837  */
   838 /* @{ */
   839 extern DECLSPEC void SDLCALL SDL_LockAudio(void);
   840 extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
   841 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
   842 extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
   843 /* @} *//* Audio lock functions */
   844 
   845 /**
   846  *  This function shuts down audio processing and closes the audio device.
   847  */
   848 extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
   849 extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
   850 
   851 /* Ends C function definitions when using C++ */
   852 #ifdef __cplusplus
   853 }
   854 #endif
   855 #include "close_code.h"
   856 
   857 #endif /* SDL_audio_h_ */
   858 
   859 /* vi: set ts=4 sw=4 expandtab: */