--- /dev/null
+/*
+ Simple DirectMedia Layer
+ Copyright (C) 1997-2022 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.
+ */
+/* @{ */
+
+/**
+ * Use this function to get the number of built-in audio drivers.
+ *
+ * This function returns a hardcoded number. This never returns a negative
+ * value; if there are no drivers compiled into this build of SDL, this
+ * function returns zero. The presence of a driver in this list does not mean
+ * it will function, it just means SDL is capable of interacting with that
+ * interface. For example, a build of SDL might have esound support, but if
+ * there's no esound server available, SDL's esound driver would fail if used.
+ *
+ * By default, SDL tries all drivers, in its preferred order, until one is
+ * found to be usable.
+ *
+ * \returns the number of built-in audio drivers.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_GetAudioDriver
+ */
+extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
+
+/**
+ * Use this function to get the name of a built in audio driver.
+ *
+ * The list of audio drivers is given in the order that they are normally
+ * initialized by default; the drivers that seem more reasonable to choose
+ * first (as far as the SDL developers believe) are earlier in the list.
+ *
+ * The names of drivers are all simple, low-ASCII identifiers, like "alsa",
+ * "coreaudio" or "xaudio2". These never have Unicode characters, and are not
+ * meant to be proper names.
+ *
+ * \param index the index of the audio driver; the value ranges from 0 to
+ * SDL_GetNumAudioDrivers() - 1
+ * \returns the name of the audio driver at the requested index, or NULL if an
+ * invalid index was specified.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_GetNumAudioDrivers
+ */
+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().
+ */
+/* @{ */
+
+/**
+ * Use this function to initialize a particular audio driver.
+ *
+ * This function is used internally, and should not be used unless you have a
+ * specific need to designate the audio driver you want to use. You should
+ * normally use SDL_Init() or SDL_InitSubSystem().
+ *
+ * \param driver_name the name of the desired audio driver
+ * \returns 0 on success or a negative error code on failure; call
+ * SDL_GetError() for more information.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_AudioQuit
+ */
+extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
+
+/**
+ * Use this function to shut down audio if you initialized it with
+ * SDL_AudioInit().
+ *
+ * This function is 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_Quit() or SDL_QuitSubSystem().
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_AudioInit
+ */
+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.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \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.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \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.
+ *
+ * \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
+ *
+ * \since This function is available since SDL 2.0.16.
+ *
+ * \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.
+ *
+ * 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.
+ *
+ * Managing the audio spec via 'desired' and 'obtained':
+ *
+ * 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.
+ *
+ * The resulting audio specs, varying depending on hardware and on what
+ * changes were allowed, will then be written back to `obtained`.
+ *
+ * If your application can only handle one specific data format, pass a zero
+ * for `allowed_changes` and let SDL transparently handle any differences.
+ *
+ * \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;
+
+/**
+ * This function is a legacy means of querying the audio device.
+ *
+ * New programs might want to use SDL_GetAudioDeviceStatus() instead. This
+ * function is equivalent to calling...
+ *
+ * ```c
+ * SDL_GetAudioDeviceStatus(1);
+ * ```
+ *
+ * ...and is only useful if you used the legacy SDL_OpenAudio() function.
+ *
+ * \returns the SDL_AudioStatus of the audio device opened by SDL_OpenAudio().
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_GetAudioDeviceStatus
+ */
+extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
+
+/**
+ * Use this function to get the current audio state of an audio device.
+ *
+ * \param dev the ID of an audio device previously opened with
+ * SDL_OpenAudioDevice()
+ * \returns the SDL_AudioStatus of the specified audio device.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_PauseAudioDevice
+ */
+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.
+ */
+/* @{ */
+
+/**
+ * This function is a legacy means of pausing the audio device.
+ *
+ * New programs might want to use SDL_PauseAudioDevice() instead. This
+ * function is equivalent to calling...
+ *
+ * ```c
+ * SDL_PauseAudioDevice(1, pause_on);
+ * ```
+ *
+ * ...and is only useful if you used the legacy SDL_OpenAudio() function.
+ *
+ * \param pause_on non-zero to pause, 0 to unpause
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_GetAudioStatus
+ * \sa SDL_PauseAudioDevice
+ */
+extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
+
+/**
+ * Use this function to pause and unpause audio playback on a specified
+ * device.
+ *
+ * This function pauses and unpauses the audio callback processing for a given
+ * device. Newly-opened audio devices start in the paused state, so you must
+ * call this function with **pause_on**=0 after opening the specified audio
+ * device to start playing sound. This allows you to safely initialize data
+ * for your callback function after opening the audio device. Silence will be
+ * written to the audio device while paused, and the audio callback is
+ * guaranteed to not be called. Pausing one device does not prevent other
+ * unpaused devices from running their callbacks.
+ *
+ * Pausing state does not stack; even if you pause a device several times, a
+ * single unpause will start the device playing again, and vice versa. This is
+ * different from how SDL_LockAudioDevice() works.
+ *
+ * If you just need to protect a few variables from race conditions vs your
+ * callback, you shouldn't pause the audio device, as it will lead to dropouts
+ * in the audio playback. Instead, you should use SDL_LockAudioDevice().
+ *
+ * \param dev a device opened by SDL_OpenAudioDevice()
+ * \param pause_on non-zero to pause, 0 to unpause
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_LockAudioDevice
+ */
+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.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \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()
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \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.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \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.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \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.
+ *
+ * \since This function is available since SDL 2.0.7.
+ *
+ * \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.
+ *
+ * \since This function is available since SDL 2.0.7.
+ *
+ * \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
+ *
+ * \since This function is available since SDL 2.0.7.
+ *
+ * \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.
+ *
+ * \since This function is available since SDL 2.0.7.
+ *
+ * \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.
+ *
+ * \since This function is available since SDL 2.0.7.
+ *
+ * \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
+ *
+ * \since This function is available since SDL 2.0.7.
+ *
+ * \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
+ *
+ * \since This function is available since SDL 2.0.7.
+ *
+ * \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
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \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
+ *
+ * \since This function is available since SDL 2.0.0.
+ */
+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 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.
+ */
+/* @{ */
+
+/**
+ * This function is a legacy means of locking the audio device.
+ *
+ * New programs might want to use SDL_LockAudioDevice() instead. This function
+ * is equivalent to calling...
+ *
+ * ```c
+ * SDL_LockAudioDevice(1);
+ * ```
+ *
+ * ...and is only useful if you used the legacy SDL_OpenAudio() function.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_LockAudioDevice
+ * \sa SDL_UnlockAudio
+ * \sa SDL_UnlockAudioDevice
+ */
+extern DECLSPEC void SDLCALL SDL_LockAudio(void);
+
+/**
+ * Use this function to lock out the audio callback function for a specified
+ * device.
+ *
+ * The lock manipulated by these functions protects the audio callback
+ * function specified in SDL_OpenAudioDevice(). During a
+ * SDL_LockAudioDevice()/SDL_UnlockAudioDevice() pair, you can be guaranteed
+ * that the callback function for that device is not running, even if the
+ * device is not paused. While a device is locked, any other unpaused,
+ * unlocked devices may still run their callbacks.
+ *
+ * Calling this function from inside your audio callback is unnecessary. SDL
+ * obtains this lock before calling your function, and releases it when the
+ * function returns.
+ *
+ * You should not hold the lock longer than absolutely necessary. If you hold
+ * it too long, you'll experience dropouts in your audio playback. Ideally,
+ * your application locks the device, sets a few variables and unlocks again.
+ * Do not do heavy work while holding the lock for a device.
+ *
+ * It is safe to lock the audio device multiple times, as long as you unlock
+ * it an equivalent number of times. The callback will not run until the
+ * device has been unlocked completely in this way. If your application fails
+ * to unlock the device appropriately, your callback will never run, you might
+ * hear repeating bursts of audio, and SDL_CloseAudioDevice() will probably
+ * deadlock.
+ *
+ * Internally, the audio device lock is a mutex; if you lock from two threads
+ * at once, not only will you block the audio callback, you'll block the other
+ * thread.
+ *
+ * \param dev the ID of the device to be locked
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_UnlockAudioDevice
+ */
+extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
+
+/**
+ * This function is a legacy means of unlocking the audio device.
+ *
+ * New programs might want to use SDL_UnlockAudioDevice() instead. This
+ * function is equivalent to calling...
+ *
+ * ```c
+ * SDL_UnlockAudioDevice(1);
+ * ```
+ *
+ * ...and is only useful if you used the legacy SDL_OpenAudio() function.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_LockAudio
+ * \sa SDL_UnlockAudioDevice
+ */
+extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
+
+/**
+ * Use this function to unlock the audio callback function for a specified
+ * device.
+ *
+ * This function should be paired with a previous SDL_LockAudioDevice() call.
+ *
+ * \param dev the ID of the device to be unlocked
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \sa SDL_LockAudioDevice
+ */
+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.
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \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()
+ *
+ * \since This function is available since SDL 2.0.0.
+ *
+ * \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: */
projects / xonotic / darkplaces.git / blobdiff