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  1. /*
  2.   Simple DirectMedia Layer
  3.   Copyright (C) 1997-2017 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 int 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: */
  826.