libsdl-org-mirror/include/SDL_audio.h

/*
  Simple DirectMedia Layer
  Copyright (C) 1997-2021 Sam Lantinga <slouken@libsdl.org>

  This software is provided 'as-is', without any express or implied
  warranty.  In no event will the authors be held liable for any damages
  arising from the use of this software.

  Permission is granted to anyone to use this software for any purpose,
  including commercial applications, and to alter it and redistribute it
  freely, subject to the following restrictions:

  1. The origin of this software must not be misrepresented; you must not
     claim that you wrote the original software. If you use this software
     in a product, an acknowledgment in the product documentation would be
     appreciated but is not required.
  2. Altered source versions must be plainly marked as such, and must not be
     misrepresented as being the original software.
  3. This notice may not be removed or altered from any source distribution.
*/

/* !!! FIXME: several functions in here need Doxygen comments. */

/**
 *  \file SDL_audio.h
 *
 *  Access to the raw audio mixing buffer for the SDL library.
 */

#ifndef SDL_audio_h_
#define SDL_audio_h_

#include "SDL_stdinc.h"
#include "SDL_error.h"
#include "SDL_endian.h"
#include "SDL_mutex.h"
#include "SDL_thread.h"
#include "SDL_rwops.h"

#include "begin_code.h"
/* Set up for C function definitions, even when using C++ */
#ifdef __cplusplus
extern "C" {
#endif

/**
 *  \brief Audio format flags.
 *
 *  These are what the 16 bits in SDL_AudioFormat currently mean...
 *  (Unspecified bits are always zero).
 *
 *  \verbatim
    ++-----------------------sample is signed if set
    ||
    ||       ++-----------sample is bigendian if set
    ||       ||
    ||       ||          ++---sample is float if set
    ||       ||          ||
    ||       ||          || +---sample bit size---+
    ||       ||          || |                     |
    15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
    \endverbatim
 *
 *  There are macros in SDL 2.0 and later to query these bits.
 */
typedef Uint16 SDL_AudioFormat;

/**
 *  \name Audio flags
 */
/* @{ */

#define SDL_AUDIO_MASK_BITSIZE       (0xFF)
#define SDL_AUDIO_MASK_DATATYPE      (1<<8)
#define SDL_AUDIO_MASK_ENDIAN        (1<<12)
#define SDL_AUDIO_MASK_SIGNED        (1<<15)
#define SDL_AUDIO_BITSIZE(x)         (x & SDL_AUDIO_MASK_BITSIZE)
#define SDL_AUDIO_ISFLOAT(x)         (x & SDL_AUDIO_MASK_DATATYPE)
#define SDL_AUDIO_ISBIGENDIAN(x)     (x & SDL_AUDIO_MASK_ENDIAN)
#define SDL_AUDIO_ISSIGNED(x)        (x & SDL_AUDIO_MASK_SIGNED)
#define SDL_AUDIO_ISINT(x)           (!SDL_AUDIO_ISFLOAT(x))
#define SDL_AUDIO_ISLITTLEENDIAN(x)  (!SDL_AUDIO_ISBIGENDIAN(x))
#define SDL_AUDIO_ISUNSIGNED(x)      (!SDL_AUDIO_ISSIGNED(x))

/**
 *  \name Audio format flags
 *
 *  Defaults to LSB byte order.
 */
/* @{ */
#define AUDIO_U8        0x0008  /**< Unsigned 8-bit samples */
#define AUDIO_S8        0x8008  /**< Signed 8-bit samples */
#define AUDIO_U16LSB    0x0010  /**< Unsigned 16-bit samples */
#define AUDIO_S16LSB    0x8010  /**< Signed 16-bit samples */
#define AUDIO_U16MSB    0x1010  /**< As above, but big-endian byte order */
#define AUDIO_S16MSB    0x9010  /**< As above, but big-endian byte order */
#define AUDIO_U16       AUDIO_U16LSB
#define AUDIO_S16       AUDIO_S16LSB
/* @} */

/**
 *  \name int32 support
 */
/* @{ */
#define AUDIO_S32LSB    0x8020  /**< 32-bit integer samples */
#define AUDIO_S32MSB    0x9020  /**< As above, but big-endian byte order */
#define AUDIO_S32       AUDIO_S32LSB
/* @} */

/**
 *  \name float32 support
 */
/* @{ */
#define AUDIO_F32LSB    0x8120  /**< 32-bit floating point samples */
#define AUDIO_F32MSB    0x9120  /**< As above, but big-endian byte order */
#define AUDIO_F32       AUDIO_F32LSB
/* @} */

/**
 *  \name Native audio byte ordering
 */
/* @{ */
#if SDL_BYTEORDER == SDL_LIL_ENDIAN
#define AUDIO_U16SYS    AUDIO_U16LSB
#define AUDIO_S16SYS    AUDIO_S16LSB
#define AUDIO_S32SYS    AUDIO_S32LSB
#define AUDIO_F32SYS    AUDIO_F32LSB
#else
#define AUDIO_U16SYS    AUDIO_U16MSB
#define AUDIO_S16SYS    AUDIO_S16MSB
#define AUDIO_S32SYS    AUDIO_S32MSB
#define AUDIO_F32SYS    AUDIO_F32MSB
#endif
/* @} */

/**
 *  \name Allow change flags
 *
 *  Which audio format changes are allowed when opening a device.
 */
/* @{ */
#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE    0x00000001
#define SDL_AUDIO_ALLOW_FORMAT_CHANGE       0x00000002
#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE     0x00000004
#define SDL_AUDIO_ALLOW_SAMPLES_CHANGE      0x00000008
#define SDL_AUDIO_ALLOW_ANY_CHANGE          (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE|SDL_AUDIO_ALLOW_SAMPLES_CHANGE)
/* @} */

/* @} *//* Audio flags */

/**
 *  This function is called when the audio device needs more data.
 *
 *  \param userdata An application-specific parameter saved in
 *                  the SDL_AudioSpec structure
 *  \param stream A pointer to the audio data buffer.
 *  \param len    The length of that buffer in bytes.
 *
 *  Once the callback returns, the buffer will no longer be valid.
 *  Stereo samples are stored in a LRLRLR ordering.
 *
 *  You can choose to avoid callbacks and use SDL_QueueAudio() instead, if
 *  you like. Just open your audio device with a NULL callback.
 */
typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
                                            int len);

/**
 *  The calculated values in this structure are calculated by SDL_OpenAudio().
 *
 *  For multi-channel audio, the default SDL channel mapping is:
 *  2:  FL FR                       (stereo)
 *  3:  FL FR LFE                   (2.1 surround)
 *  4:  FL FR BL BR                 (quad)
 *  5:  FL FR FC BL BR              (quad + center)
 *  6:  FL FR FC LFE SL SR          (5.1 surround - last two can also be BL BR)
 *  7:  FL FR FC LFE BC SL SR       (6.1 surround)
 *  8:  FL FR FC LFE BL BR SL SR    (7.1 surround)
 */
typedef struct SDL_AudioSpec
{
    int freq;                   /**< DSP frequency -- samples per second */
    SDL_AudioFormat format;     /**< Audio data format */
    Uint8 channels;             /**< Number of channels: 1 mono, 2 stereo */
    Uint8 silence;              /**< Audio buffer silence value (calculated) */
    Uint16 samples;             /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */
    Uint16 padding;             /**< Necessary for some compile environments */
    Uint32 size;                /**< Audio buffer size in bytes (calculated) */
    SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
    void *userdata;             /**< Userdata passed to callback (ignored for NULL callbacks). */
} SDL_AudioSpec;


struct SDL_AudioCVT;
typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
                                          SDL_AudioFormat format);

/**
 *  \brief Upper limit of filters in SDL_AudioCVT
 *
 *  The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is
 *  currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers,
 *  one of which is the terminating NULL pointer.
 */
#define SDL_AUDIOCVT_MAX_FILTERS 9

/**
 *  \struct SDL_AudioCVT
 *  \brief A structure to hold a set of audio conversion filters and buffers.
 *
 *  Note that various parts of the conversion pipeline can take advantage
 *  of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
 *  you to pass it aligned data, but can possibly run much faster if you
 *  set both its (buf) field to a pointer that is aligned to 16 bytes, and its
 *  (len) field to something that's a multiple of 16, if possible.
 */
#if defined(__GNUC__) && !defined(__CHERI_PURE_CAPABILITY__)
/* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
   pad it out to 88 bytes to guarantee ABI compatibility between compilers.
   This is not a concern on CHERI architectures, where pointers must be stored
   at aligned locations otherwise they will become invalid, and thus structs
   containing pointers cannot be packed without giving a warning or error.
   vvv
   The next time we rev the ABI, make sure to size the ints and add padding.
*/
#define SDL_AUDIOCVT_PACKED __attribute__((packed))
#else
#define SDL_AUDIOCVT_PACKED
#endif
/* */
typedef struct SDL_AudioCVT
{
    int needed;                 /**< Set to 1 if conversion possible */
    SDL_AudioFormat src_format; /**< Source audio format */
    SDL_AudioFormat dst_format; /**< Target audio format */
    double rate_incr;           /**< Rate conversion increment */
    Uint8 *buf;                 /**< Buffer to hold entire audio data */
    int len;                    /**< Length of original audio buffer */
    int len_cvt;                /**< Length of converted audio buffer */
    int len_mult;               /**< buffer must be len*len_mult big */
    double len_ratio;           /**< Given len, final size is len*len_ratio */
    SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
    int filter_index;           /**< Current audio conversion function */
} SDL_AUDIOCVT_PACKED SDL_AudioCVT;


/* Function prototypes */

/**
 *  \name Driver discovery functions
 *
 *  These functions return the list of built in audio drivers, in the
 *  order that they are normally initialized by default.
 */
/* @{ */
extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
/* @} */

/**
 *  \name Initialization and cleanup
 *
 *  \internal These functions are used internally, and should not be used unless
 *            you have a specific need to specify the audio driver you want to
 *            use.  You should normally use SDL_Init() or SDL_InitSubSystem().
 */
/* @{ */
extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
/* @} */

/**
 * Get the name of the current audio driver.
 *
 * The returned string points to internal static memory and thus never becomes
 * invalid, even if you quit the audio subsystem and initialize a new driver
 * (although such a case would return a different static string from another
 * call to this function, of course). As such, you should not modify or free
 * the returned string.
 *
 * \returns the name of the current audio driver or NULL if no driver has been
 *          initialized.
 *
 * \since This function is available since SDL 2.0.0.
 *
 * \sa SDL_AudioInit
 */
extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);

/**
 * This function is a legacy means of opening the audio device.
 *
 * This function remains for compatibility with SDL 1.2, but also because it's
 * slightly easier to use than the new functions in SDL 2.0. The new, more
 * powerful, and preferred way to do this is SDL_OpenAudioDevice().
 *
 * This function is roughly equivalent to:
 *
 * ```c++
 * SDL_OpenAudioDevice(NULL, 0, desired, obtained, SDL_AUDIO_ALLOW_ANY_CHANGE);
 * ```
 *
 * With two notable exceptions:
 *
 * - If `obtained` is NULL, we use `desired` (and allow no changes), which
 *   means desired will be modified to have the correct values for silence,
 *   etc, and SDL will convert any differences between your app's specific
 *   request and the hardware behind the scenes.
 * - The return value is always success or failure, and not a device ID, which
 *   means you can only have one device open at a time with this function.
 *
 * \param desired an SDL_AudioSpec structure representing the desired output
 *                format. Please refer to the SDL_OpenAudioDevice
 *                documentation for details on how to prepare this structure.
 * \param obtained an SDL_AudioSpec structure filled in with the actual
 *                 parameters, or NULL.
 * \returns 0 if successful, placing the actual hardware parameters in the
 *          structure pointed to by `obtained`.
 *
 *          If `obtained` is NULL, the audio data passed to the callback
 *          function will be guaranteed to be in the requested format, and
 *          will be automatically converted to the actual hardware audio
 *          format if necessary. If `obtained` is NULL, `desired` will have
 *          fields modified.
 *
 *          This function returns a negative error code on failure to open the
 *          audio device or failure to set up the audio thread; call
 *          SDL_GetError() for more information.
 *
 * \sa SDL_CloseAudio
 * \sa SDL_LockAudio
 * \sa SDL_PauseAudio
 * \sa SDL_UnlockAudio
 */
extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,
                                          SDL_AudioSpec * obtained);

/**
 *  SDL Audio Device IDs.
 *
 *  A successful call to SDL_OpenAudio() is always device id 1, and legacy
 *  SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls
 *  always returns devices >= 2 on success. The legacy calls are good both
 *  for backwards compatibility and when you don't care about multiple,
 *  specific, or capture devices.
 */
typedef Uint32 SDL_AudioDeviceID;

/**
 * Get the number of built-in audio devices.
 *
 * This function is only valid after successfully initializing the audio
 * subsystem.
 *
 * Note that audio capture support is not implemented as of SDL 2.0.4, so the
 * `iscapture` parameter is for future expansion and should always be zero for
 * now.
 *
 * This function will return -1 if an explicit list of devices can't be
 * determined. Returning -1 is not an error. For example, if SDL is set up to
 * talk to a remote audio server, it can't list every one available on the
 * Internet, but it will still allow a specific host to be specified in
 * SDL_OpenAudioDevice().
 *
 * In many common cases, when this function returns a value <= 0, it can still
 * successfully open the default device (NULL for first argument of
 * SDL_OpenAudioDevice()).
 *
 * This function may trigger a complete redetect of available hardware. It
 * should not be called for each iteration of a loop, but rather once at the
 * start of a loop:
 *
 * ```c++
 * // Don't do this:
 * for (int i = 0; i < SDL_GetNumAudioDevices(0); i++)
 *
 * // do this instead:
 * const int count = SDL_GetNumAudioDevices(0);
 * for (int i = 0; i < count; ++i) { do_something_here(); }
 * ```
 *
 * \param iscapture zero to request playback devices, non-zero to request
 *                  recording devices
 * \returns the number of available devices exposed by the current driver or
 *          -1 if an explicit list of devices can't be determined. A return
 *          value of -1 does not necessarily mean an error condition.
 *
 * \since This function is available since SDL 2.0.0.
 *
 * \sa SDL_GetAudioDeviceName
 * \sa SDL_OpenAudioDevice
 */
extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);

/**
 * Get the human-readable name of a specific audio device.
 *
 * This function is only valid after successfully initializing the audio
 * subsystem. The values returned by this function reflect the latest call to
 * SDL_GetNumAudioDevices(); re-call that function to redetect available
 * hardware.
 *
 * The string returned by this function is UTF-8 encoded, read-only, and
 * managed internally. You are not to free it. If you need to keep the string
 * for any length of time, you should make your own copy of it, as it will be
 * invalid next time any of several other SDL functions are called.
 *
 * \param index the index of the audio device; valid values range from 0 to
 *              SDL_GetNumAudioDevices() - 1
 * \param iscapture non-zero to query the list of recording devices, zero to
 *                  query the list of output devices.
 * \returns the name of the audio device at the requested index, or NULL on
 *          error.
 *
 * \sa SDL_GetNumAudioDevices
 */
extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
                                                           int iscapture);

/**
 * Get the preferred audio format of a specific audio device.
 *
 * This function is only valid after a successfully initializing the audio
 * subsystem. The values returned by this function reflect the latest call to
 * SDL_GetNumAudioDevices(); re-call that function to redetect available
 * hardware.
 *
 * `spec` will be filled with the sample rate, sample format, and channel
 * count. All other values in the structure are filled with 0. When the
 * supported struct members are 0, SDL was unable to get the property from the
 * backend.
 *
 * \param index the index of the audio device; valid values range from 0 to
 *              SDL_GetNumAudioDevices() - 1
 * \param iscapture non-zero to query the list of recording devices, zero to
 *                  query the list of output devices.
 * \param spec The SDL_AudioSpec to be initialized by this function.
 * \returns 0 on success, nonzero on error
 *
 * \sa SDL_GetNumAudioDevices
 */
extern DECLSPEC int SDLCALL SDL_GetAudioDeviceSpec(int index,
                                                   int iscapture,
                                                   SDL_AudioSpec *spec);


/**
 * Open a specific audio device.
 *
 * SDL_OpenAudio(), unlike this function, always acts on device ID 1. As such,
 * this function will never return a 1 so as not to conflict with the legacy
 * function.
 *
 * Please note that SDL 2.0 before 2.0.5 did not support recording; as such,
 * this function would fail if `iscapture` was not zero. Starting with SDL
 * 2.0.5, recording is implemented and this value can be non-zero.
 *
 * Passing in a `device` name of NULL requests the most reasonable default
 * (and is equivalent to what SDL_OpenAudio() does to choose a device). The
 * `device` name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but
 * some drivers allow arbitrary and driver-specific strings, such as a
 * hostname/IP address for a remote audio server, or a filename in the
 * diskaudio driver.
 *
 * When filling in the desired audio spec structure:
 *
 * - `desired->freq` should be the frequency in sample-frames-per-second (Hz).
 * - `desired->format` should be the audio format (`AUDIO_S16SYS`, etc).
 * - `desired->samples` is the desired size of the audio buffer, in _sample
 *   frames_ (with stereo output, two samples--left and right--would make a
 *   single sample frame). This number should be a power of two, and may be
 *   adjusted by the audio driver to a value more suitable for the hardware.
 *   Good values seem to range between 512 and 8096 inclusive, depending on
 *   the application and CPU speed. Smaller values reduce latency, but can
 *   lead to underflow if the application is doing heavy processing and cannot
 *   fill the audio buffer in time. Note that the number of sample frames is
 *   directly related to time by the following formula: `ms =
 *   (sampleframes*1000)/freq`
 * - `desired->size` is the size in _bytes_ of the audio buffer, and is
 *   calculated by SDL_OpenAudioDevice(). You don't initialize this.
 * - `desired->silence` is the value used to set the buffer to silence, and is
 *   calculated by SDL_OpenAudioDevice(). You don't initialize this.
 * - `desired->callback` should be set to a function that will be called when
 *   the audio device is ready for more data. It is passed a pointer to the
 *   audio buffer, and the length in bytes of the audio buffer. This function
 *   usually runs in a separate thread, and so you should protect data
 *   structures that it accesses by calling SDL_LockAudioDevice() and
 *   SDL_UnlockAudioDevice() in your code. Alternately, you may pass a NULL
 *   pointer here, and call SDL_QueueAudio() with some frequency, to queue
 *   more audio samples to be played (or for capture devices, call
 *   SDL_DequeueAudio() with some frequency, to obtain audio samples).
 * - `desired->userdata` is passed as the first parameter to your callback
 *   function. If you passed a NULL callback, this value is ignored.
 *
 * `allowed_changes` can have the following flags OR'd together:
 *
 * - `SDL_AUDIO_ALLOW_FREQUENCY_CHANGE`
 * - `SDL_AUDIO_ALLOW_FORMAT_CHANGE`
 * - `SDL_AUDIO_ALLOW_CHANNELS_CHANGE`
 * - `SDL_AUDIO_ALLOW_ANY_CHANGE`
 *
 * These flags specify how SDL should behave when a device cannot offer a
 * specific feature. If the application requests a feature that the hardware
 * doesn't offer, SDL will always try to get the closest equivalent.
 *
 * For example, if you ask for float32 audio format, but the sound card only
 * supports int16, SDL will set the hardware to int16. If you had set
 * SDL_AUDIO_ALLOW_FORMAT_CHANGE, SDL will change the format in the `obtained`
 * structure. If that flag was *not* set, SDL will prepare to convert your
 * callback's float32 audio to int16 before feeding it to the hardware and
 * will keep the originally requested format in the `obtained` structure.
 *
 * If your application can only handle one specific data format, pass a zero
 * for `allowed_changes` and let SDL transparently handle any differences.
 *
 * An opened audio device starts out paused, and should be enabled for playing
 * by calling SDL_PauseAudioDevice(devid, 0) when you are ready for your audio
 * callback function to be called. Since the audio driver may modify the
 * requested size of the audio buffer, you should allocate any local mixing
 * buffers after you open the audio device.
 *
 * The audio callback runs in a separate thread in most cases; you can prevent
 * race conditions between your callback and other threads without fully
 * pausing playback with SDL_LockAudioDevice(). For more information about the
 * callback, see SDL_AudioSpec.
 *
 * \param device a UTF-8 string reported by SDL_GetAudioDeviceName() or a
 *               driver-specific name as appropriate. NULL requests the most
 *               reasonable default device.
 * \param iscapture non-zero to specify a device should be opened for
 *                  recording, not playback
 * \param desired an SDL_AudioSpec structure representing the desired output
 *                format; see SDL_OpenAudio() for more information
 * \param obtained an SDL_AudioSpec structure filled in with the actual output
 *                 format; see SDL_OpenAudio() for more information
 * \param allowed_changes 0, or one or more flags OR'd together
 * \returns a valid device ID that is > 0 on success or 0 on failure; call
 *          SDL_GetError() for more information.
 *
 *          For compatibility with SDL 1.2, this will never return 1, since
 *          SDL reserves that ID for the legacy SDL_OpenAudio() function.
 *
 * \since This function is available since SDL 2.0.0.
 *
 * \sa SDL_CloseAudioDevice
 * \sa SDL_GetAudioDeviceName
 * \sa SDL_LockAudioDevice
 * \sa SDL_OpenAudio
 * \sa SDL_PauseAudioDevice
 * \sa SDL_UnlockAudioDevice
 */
extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(
                                                  const char *device,
                                                  int iscapture,
                                                  const SDL_AudioSpec *desired,
                                                  SDL_AudioSpec *obtained,
                                                  int allowed_changes);



/**
 *  \name Audio state
 *
 *  Get the current audio state.
 */
/* @{ */
typedef enum
{
    SDL_AUDIO_STOPPED = 0,
    SDL_AUDIO_PLAYING,
    SDL_AUDIO_PAUSED
} SDL_AudioStatus;
extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
/* @} *//* Audio State */

/**
 *  \name Pause audio functions
 *
 *  These functions pause and unpause the audio callback processing.
 *  They should be called with a parameter of 0 after opening the audio
 *  device to start playing sound.  This is so you can safely initialize
 *  data for your callback function after opening the audio device.
 *  Silence will be written to the audio device during the pause.
 */
/* @{ */
extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
                                                  int pause_on);
/* @} *//* Pause audio functions */

/**
 * Load the audio data of a WAVE file into memory.
 *
 * Loading a WAVE file requires `src`, `spec`, `audio_buf` and `audio_len` to
 * be valid pointers. The entire data portion of the file is then loaded into
 * memory and decoded if necessary.
 *
 * If `freesrc` is non-zero, the data source gets automatically closed and
 * freed before the function returns.
 *
 * Supported formats are RIFF WAVE files with the formats PCM (8, 16, 24, and
 * 32 bits), IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and
 * A-law and mu-law (8 bits). Other formats are currently unsupported and
 * cause an error.
 *
 * If this function succeeds, the pointer returned by it is equal to `spec`
 * and the pointer to the audio data allocated by the function is written to
 * `audio_buf` and its length in bytes to `audio_len`. The SDL_AudioSpec
 * members `freq`, `channels`, and `format` are set to the values of the audio
 * data in the buffer. The `samples` member is set to a sane default and all
 * others are set to zero.
 *
 * It's necessary to use SDL_FreeWAV() to free the audio data returned in
 * `audio_buf` when it is no longer used.
 *
 * Because of the underspecification of the .WAV format, there are many
 * problematic files in the wild that cause issues with strict decoders. To
 * provide compatibility with these files, this decoder is lenient in regards
 * to the truncation of the file, the fact chunk, and the size of the RIFF
 * chunk. The hints `SDL_HINT_WAVE_RIFF_CHUNK_SIZE`,
 * `SDL_HINT_WAVE_TRUNCATION`, and `SDL_HINT_WAVE_FACT_CHUNK` can be used to
 * tune the behavior of the loading process.
 *
 * Any file that is invalid (due to truncation, corruption, or wrong values in
 * the headers), too big, or unsupported causes an error. Additionally, any
 * critical I/O error from the data source will terminate the loading process
 * with an error. The function returns NULL on error and in all cases (with
 * the exception of `src` being NULL), an appropriate error message will be
 * set.
 *
 * It is required that the data source supports seeking.
 *
 * Example:
 *
 * ```c++
 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, &spec, &buf, &len);
 * ```
 *
 * Note that the SDL_LoadWAV macro does this same thing for you, but in a less
 * messy way:
 *
 * ```c++
 * SDL_LoadWAV("sample.wav", &spec, &buf, &len);
 * ```
 *
 * \param src The data source for the WAVE data
 * \param freesrc If non-zero, SDL will _always_ free the data source
 * \param spec An SDL_AudioSpec that will be filled in with the wave file's
 *             format details
 * \param audio_buf A pointer filled with the audio data, allocated by the
 *                  function.
 * \param audio_len A pointer filled with the length of the audio data buffer
 *                  in bytes
 * \returns This function, if successfully called, returns `spec`, which will
 *          be filled with the audio data format of the wave source data.
 *          `audio_buf` will be filled with a pointer to an allocated buffer
 *          containing the audio data, and `audio_len` is filled with the
 *          length of that audio buffer in bytes.
 *
 *          This function returns NULL if the .WAV file cannot be opened, uses
 *          an unknown data format, or is corrupt; call SDL_GetError() for
 *          more information.
 *
 *          When the application is done with the data returned in
 *          `audio_buf`, it should call SDL_FreeWAV() to dispose of it.
 *
 * \sa SDL_FreeWAV
 * \sa SDL_LoadWAV
 */
extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
                                                      int freesrc,
                                                      SDL_AudioSpec * spec,
                                                      Uint8 ** audio_buf,
                                                      Uint32 * audio_len);

/**
 *  Loads a WAV from a file.
 *  Compatibility convenience function.
 */
#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
    SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)

/**
 * Free data previously allocated with SDL_LoadWAV() or SDL_LoadWAV_RW().
 *
 * After a WAVE file has been opened with SDL_LoadWAV() or SDL_LoadWAV_RW()
 * its data can eventually be freed with SDL_FreeWAV(). It is safe to call
 * this function with a NULL pointer.
 *
 * \param audio_buf a pointer to the buffer created by SDL_LoadWAV() or
 *                  SDL_LoadWAV_RW()
 *
 * \sa SDL_LoadWAV
 * \sa SDL_LoadWAV_RW
 */
extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);

/**
 * Initialize an SDL_AudioCVT structure for conversion.
 *
 * Before an SDL_AudioCVT structure can be used to convert audio data it must
 * be initialized with source and destination information.
 *
 * This function will zero out every field of the SDL_AudioCVT, so it must be
 * called before the application fills in the final buffer information.
 *
 * Once this function has returned successfully, and reported that a
 * conversion is necessary, the application fills in the rest of the fields in
 * SDL_AudioCVT, now that it knows how large a buffer it needs to allocate,
 * and then can call SDL_ConvertAudio() to complete the conversion.
 *
 * \param cvt an SDL_AudioCVT structure filled in with audio conversion
 *            information
 * \param src_format the source format of the audio data; for more info see
 *                   SDL_AudioFormat
 * \param src_channels the number of channels in the source
 * \param src_rate the frequency (sample-frames-per-second) of the source
 * \param dst_format the destination format of the audio data; for more info
 *                   see SDL_AudioFormat
 * \param dst_channels the number of channels in the destination
 * \param dst_rate the frequency (sample-frames-per-second) of the destination
 * \returns 1 if the audio filter is prepared, 0 if no conversion is needed,
 *          or a negative error code on failure; call SDL_GetError() for more
 *          information.
 *
 * \sa SDL_ConvertAudio
 */
extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
                                              SDL_AudioFormat src_format,
                                              Uint8 src_channels,
                                              int src_rate,
                                              SDL_AudioFormat dst_format,
                                              Uint8 dst_channels,
                                              int dst_rate);

/**
 * Convert audio data to a desired audio format.
 *
 * This function does the actual audio data conversion, after the application
 * has called SDL_BuildAudioCVT() to prepare the conversion information and
 * then filled in the buffer details.
 *
 * Once the application has initialized the `cvt` structure using
 * SDL_BuildAudioCVT(), allocated an audio buffer and filled it with audio
 * data in the source format, this function will convert the buffer, in-place,
 * to the desired format.
 *
 * The data conversion may go through several passes; any given pass may
 * possibly temporarily increase the size of the data. For example, SDL might
 * expand 16-bit data to 32 bits before resampling to a lower frequency,
 * shrinking the data size after having grown it briefly. Since the supplied
 * buffer will be both the source and destination, converting as necessary
 * in-place, the application must allocate a buffer that will fully contain
 * the data during its largest conversion pass. After SDL_BuildAudioCVT()
 * returns, the application should set the `cvt->len` field to the size, in
 * bytes, of the source data, and allocate a buffer that is `cvt->len *
 * cvt->len_mult` bytes long for the `buf` field.
 *
 * The source data should be copied into this buffer before the call to
 * SDL_ConvertAudio(). Upon successful return, this buffer will contain the
 * converted audio, and `cvt->len_cvt` will be the size of the converted data,
 * in bytes. Any bytes in the buffer past `cvt->len_cvt` are undefined once
 * this function returns.
 *
 * \param cvt an SDL_AudioCVT structure that was previously set up by
 *            SDL_BuildAudioCVT().
 * \returns 0 if the conversion was completed successfully or a negative error
 *          code on failure; call SDL_GetError() for more information.
 *
 * \sa SDL_BuildAudioCVT
 */
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);

/* SDL_AudioStream is a new audio conversion interface.
   The benefits vs SDL_AudioCVT:
    - it can handle resampling data in chunks without generating
      artifacts, when it doesn't have the complete buffer available.
    - it can handle incoming data in any variable size.
    - You push data as you have it, and pull it when you need it
 */
/* this is opaque to the outside world. */
struct _SDL_AudioStream;
typedef struct _SDL_AudioStream SDL_AudioStream;

/**
 * Create a new audio stream.
 *
 * \param src_format The format of the source audio
 * \param src_channels The number of channels of the source audio
 * \param src_rate The sampling rate of the source audio
 * \param dst_format The format of the desired audio output
 * \param dst_channels The number of channels of the desired audio output
 * \param dst_rate The sampling rate of the desired audio output
 * \returns 0 on success, or -1 on error.
 *
 * \sa SDL_AudioStreamPut
 * \sa SDL_AudioStreamGet
 * \sa SDL_AudioStreamAvailable
 * \sa SDL_AudioStreamFlush
 * \sa SDL_AudioStreamClear
 * \sa SDL_FreeAudioStream
 */
extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
                                           const Uint8 src_channels,
                                           const int src_rate,
                                           const SDL_AudioFormat dst_format,
                                           const Uint8 dst_channels,
                                           const int dst_rate);

/**
 * Add data to be converted/resampled to the stream.
 *
 * \param stream The stream the audio data is being added to
 * \param buf A pointer to the audio data to add
 * \param len The number of bytes to write to the stream
 * \returns 0 on success, or -1 on error.
 *
 * \sa SDL_NewAudioStream
 * \sa SDL_AudioStreamGet
 * \sa SDL_AudioStreamAvailable
 * \sa SDL_AudioStreamFlush
 * \sa SDL_AudioStreamClear
 * \sa SDL_FreeAudioStream
 */
extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);

/**
 * Get converted/resampled data from the stream
 *
 * \param stream The stream the audio is being requested from
 * \param buf A buffer to fill with audio data
 * \param len The maximum number of bytes to fill
 * \returns the number of bytes read from the stream, or -1 on error
 *
 * \sa SDL_NewAudioStream
 * \sa SDL_AudioStreamPut
 * \sa SDL_AudioStreamAvailable
 * \sa SDL_AudioStreamFlush
 * \sa SDL_AudioStreamClear
 * \sa SDL_FreeAudioStream
 */
extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);

/**
 * Get the number of converted/resampled bytes available.
 *
 * The stream may be buffering data behind the scenes until it has enough to
 * resample correctly, so this number might be lower than what you expect, or
 * even be zero. Add more data or flush the stream if you need the data now.
 *
 * \sa SDL_NewAudioStream
 * \sa SDL_AudioStreamPut
 * \sa SDL_AudioStreamGet
 * \sa SDL_AudioStreamFlush
 * \sa SDL_AudioStreamClear
 * \sa SDL_FreeAudioStream
 */
extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);

/**
 * Tell the stream that you're done sending data, and anything being buffered
 * should be converted/resampled and made available immediately.
 *
 * It is legal to add more data to a stream after flushing, but there will be
 * audio gaps in the output. Generally this is intended to signal the end of
 * input, so the complete output becomes available.
 *
 * \sa SDL_NewAudioStream
 * \sa SDL_AudioStreamPut
 * \sa SDL_AudioStreamGet
 * \sa SDL_AudioStreamAvailable
 * \sa SDL_AudioStreamClear
 * \sa SDL_FreeAudioStream
 */
extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);

/**
 * Clear any pending data in the stream without converting it
 *
 * \sa SDL_NewAudioStream
 * \sa SDL_AudioStreamPut
 * \sa SDL_AudioStreamGet
 * \sa SDL_AudioStreamAvailable
 * \sa SDL_AudioStreamFlush
 * \sa SDL_FreeAudioStream
 */
extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);

/**
 * Free an audio stream
 *
 * \sa SDL_NewAudioStream
 * \sa SDL_AudioStreamPut
 * \sa SDL_AudioStreamGet
 * \sa SDL_AudioStreamAvailable
 * \sa SDL_AudioStreamFlush
 * \sa SDL_AudioStreamClear
 */
extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);

#define SDL_MIX_MAXVOLUME 128
/**
 * This function is a legacy means of mixing audio.
 *
 * This function is equivalent to calling
 *
 * ```c++
 * SDL_MixAudioFormat(dst, src, format, len, volume);
 * ```
 *
 * where `format` is the obtained format of the audio device from the legacy
 * SDL_OpenAudio() function.
 *
 * \param dst the destination for the mixed audio
 * \param src the source audio buffer to be mixed
 * \param len the length of the audio buffer in bytes
 * \param volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
 *               for full audio volume
 *
 * \sa SDL_MixAudioFormat
 */
extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
                                          Uint32 len, int volume);

/**
 * Mix audio data in a specified format.
 *
 * This takes an audio buffer `src` of `len` bytes of `format` data and mixes
 * it into `dst`, performing addition, volume adjustment, and overflow
 * clipping. The buffer pointed to by `dst` must also be `len` bytes of
 * `format` data.
 *
 * This is provided for convenience -- you can mix your own audio data.
 *
 * Do not use this function for mixing together more than two streams of
 * sample data. The output from repeated application of this function may be
 * distorted by clipping, because there is no accumulator with greater range
 * than the input (not to mention this being an inefficient way of doing it).
 *
 * It is a common misconception that this function is required to write audio
 * data to an output stream in an audio callback. While you can do that,
 * SDL_MixAudioFormat() is really only needed when you're mixing a single
 * audio stream with a volume adjustment.
 *
 * \param dst the destination for the mixed audio
 * \param src the source audio buffer to be mixed
 * \param format the SDL_AudioFormat structure representing the desired audio
 *               format
 * \param len the length of the audio buffer in bytes
 * \param volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
 *               for full audio volume
 */
extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
                                                const Uint8 * src,
                                                SDL_AudioFormat format,
                                                Uint32 len, int volume);

/**
 * Queue more audio on non-callback devices.
 *
 * If you are looking to retrieve queued audio from a non-callback capture
 * device, you want SDL_DequeueAudio() instead. SDL_QueueAudio() will return
 * -1 to signify an error if you use it with capture devices.
 *
 * SDL offers two ways to feed audio to the device: you can either supply a
 * callback that SDL triggers with some frequency to obtain more audio (pull
 * method), or you can supply no callback, and then SDL will expect you to
 * supply data at regular intervals (push method) with this function.
 *
 * There are no limits on the amount of data you can queue, short of
 * exhaustion of address space. Queued data will drain to the device as
 * necessary without further intervention from you. If the device needs audio
 * but there is not enough queued, it will play silence to make up the
 * difference. This means you will have skips in your audio playback if you
 * aren't routinely queueing sufficient data.
 *
 * This function copies the supplied data, so you are safe to free it when the
 * function returns. This function is thread-safe, but queueing to the same
 * device from two threads at once does not promise which buffer will be
 * queued first.
 *
 * You may not queue audio on a device that is using an application-supplied
 * callback; doing so returns an error. You have to use the audio callback or
 * queue audio with this function, but not both.
 *
 * You should not call SDL_LockAudio() on the device before queueing; SDL
 * handles locking internally for this function.
 *
 * Note that SDL2
 * [https://discourse.libsdl.org/t/sdl2-support-for-planar-audio/31263/3 does
 * not support planar audio]. You will need to resample from planar audio
 * formats into a non-planar one (see SDL_AudioFormat) before queuing audio.
 *
 * \param dev the device ID to which we will queue audio
 * \param data the data to queue to the device for later playback
 * \param len the number of bytes (not samples!) to which `data` points
 * \returns 0 on success or a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \since This function is available since SDL 2.0.4.
 *
 * \sa SDL_ClearQueuedAudio
 * \sa SDL_GetQueuedAudioSize
 */
extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);

/**
 * Dequeue more audio on non-callback devices.
 *
 * If you are looking to queue audio for output on a non-callback playback
 * device, you want SDL_QueueAudio() instead. SDL_DequeueAudio() will always
 * return 0 if you use it with playback devices.
 *
 * SDL offers two ways to retrieve audio from a capture device: you can either
 * supply a callback that SDL triggers with some frequency as the device
 * records more audio data, (push method), or you can supply no callback, and
 * then SDL will expect you to retrieve data at regular intervals (pull
 * method) with this function.
 *
 * There are no limits on the amount of data you can queue, short of
 * exhaustion of address space. Data from the device will keep queuing as
 * necessary without further intervention from you. This means you will
 * eventually run out of memory if you aren't routinely dequeueing data.
 *
 * Capture devices will not queue data when paused; if you are expecting to
 * not need captured audio for some length of time, use SDL_PauseAudioDevice()
 * to stop the capture device from queueing more data. This can be useful
 * during, say, level loading times. When unpaused, capture devices will start
 * queueing data from that point, having flushed any capturable data available
 * while paused.
 *
 * This function is thread-safe, but dequeueing from the same device from two
 * threads at once does not promise which thread will dequeue data first.
 *
 * You may not dequeue audio from a device that is using an
 * application-supplied callback; doing so returns an error. You have to use
 * the audio callback, or dequeue audio with this function, but not both.
 *
 * You should not call SDL_LockAudio() on the device before dequeueing; SDL
 * handles locking internally for this function.
 *
 * \param dev the device ID from which we will dequeue audio
 * \param data a pointer into where audio data should be copied
 * \param len the number of bytes (not samples!) to which (data) points
 * \returns the number of bytes dequeued, which could be less than requested;
 *          call SDL_GetError() for more information.
 *
 * \since This function is available since SDL 2.0.5.
 *
 * \sa SDL_ClearQueuedAudio
 * \sa SDL_GetQueuedAudioSize
 */
extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);

/**
 * Get the number of bytes of still-queued audio.
 *
 * For playback devices: this is the number of bytes that have been queued for
 * playback with SDL_QueueAudio(), but have not yet been sent to the hardware.
 *
 * Once we've sent it to the hardware, this function can not decide the exact
 * byte boundary of what has been played. It's possible that we just gave the
 * hardware several kilobytes right before you called this function, but it
 * hasn't played any of it yet, or maybe half of it, etc.
 *
 * For capture devices, this is the number of bytes that have been captured by
 * the device and are waiting for you to dequeue. This number may grow at any
 * time, so this only informs of the lower-bound of available data.
 *
 * You may not queue or dequeue audio on a device that is using an
 * application-supplied callback; calling this function on such a device
 * always returns 0. You have to use the audio callback or queue audio, but
 * not both.
 *
 * You should not call SDL_LockAudio() on the device before querying; SDL
 * handles locking internally for this function.
 *
 * \param dev the device ID of which we will query queued audio size
 * \returns the number of bytes (not samples!) of queued audio.
 *
 * \since This function is available since SDL 2.0.4.
 *
 * \sa SDL_ClearQueuedAudio
 * \sa SDL_QueueAudio
 * \sa SDL_DequeueAudio
 */
extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);

/**
 * Drop any queued audio data waiting to be sent to the hardware.
 *
 * Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
 * output devices, the hardware will start playing silence if more audio isn't
 * queued. For capture devices, the hardware will start filling the empty
 * queue with new data if the capture device isn't paused.
 *
 * This will not prevent playback of queued audio that's already been sent to
 * the hardware, as we can not undo that, so expect there to be some fraction
 * of a second of audio that might still be heard. This can be useful if you
 * want to, say, drop any pending music or any unprocessed microphone input
 * during a level change in your game.
 *
 * You may not queue or dequeue audio on a device that is using an
 * application-supplied callback; calling this function on such a device
 * always returns 0. You have to use the audio callback or queue audio, but
 * not both.
 *
 * You should not call SDL_LockAudio() on the device before clearing the
 * queue; SDL handles locking internally for this function.
 *
 * This function always succeeds and thus returns void.
 *
 * \param dev the device ID of which to clear the audio queue
 *
 * \since This function is available since SDL 2.0.4.
 *
 * \sa SDL_GetQueuedAudioSize
 * \sa SDL_QueueAudio
 * \sa SDL_DequeueAudio
 */
extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);


/**
 *  \name Audio lock functions
 *
 *  The lock manipulated by these functions protects the callback function.
 *  During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that
 *  the callback function is not running.  Do not call these from the callback
 *  function or you will cause deadlock.
 */
/* @{ */
extern DECLSPEC void SDLCALL SDL_LockAudio(void);
extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
/* @} *//* Audio lock functions */

/**
 * This function is a legacy means of closing the audio device.
 *
 * This function is equivalent to calling
 *
 * ```c++
 * SDL_CloseAudioDevice(1);
 * ```
 *
 * and is only useful if you used the legacy SDL_OpenAudio() function.
 *
 * \sa SDL_OpenAudio
 */
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);

/**
 * Use this function to shut down audio processing and close the audio device.
 *
 * The application should close open audio devices once they are no longer
 * needed. Calling this function will wait until the device's audio callback
 * is not running, release the audio hardware and then clean up internal
 * state. No further audio will play from this device once this function
 * returns.
 *
 * This function may block briefly while pending audio data is played by the
 * hardware, so that applications don't drop the last buffer of data they
 * supplied.
 *
 * The device ID is invalid as soon as the device is closed, and is eligible
 * for reuse in a new SDL_OpenAudioDevice() call immediately.
 *
 * \param dev an audio device previously opened with SDL_OpenAudioDevice()
 *
 * \sa SDL_OpenAudioDevice
 */
extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);

/* Ends C function definitions when using C++ */
#ifdef __cplusplus
}
#endif
#include "close_code.h"

#endif /* SDL_audio_h_ */

/* vi: set ts=4 sw=4 expandtab: */