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SRB2/libs/SDL2/x86_64-w64-mingw32/include/SDL2/SDL_audio.h
Steel Titanium 312b9f76b6 Revert "Merge branch 'revert-eda30fc9' into 'master'" 
This reverts merge request !435
2019-10-30 10:37:22 +00:00

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/*
Simple DirectMedia Layer
Copyright (C) 1997-2019 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.
*/
/**
* \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.
*/
#ifdef __GNUC__
/* 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.
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);
/* @} */
/**
* This function returns the name of the current audio driver, or NULL
* if no driver has been initialized.
*/
extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
/**
* This function opens the audio device with the desired parameters, and
* returns 0 if successful, placing the actual hardware parameters in the
* structure pointed to by \c obtained. If \c 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 hardware
* audio format if necessary. This function returns -1 if it failed
* to open the audio device, or couldn't set up the audio thread.
*
* When filling in the desired audio spec structure,
* - \c desired->freq should be the desired audio frequency in samples-per-
* second.
* - \c desired->format should be the desired audio format.
* - \c desired->samples is the desired size of the audio buffer, in
* samples. 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 yield faster response time,
* but can lead to underflow if the application is doing heavy processing
* and cannot fill the audio buffer in time. A stereo sample consists of
* both right and left channels in LR ordering.
* Note that the number of samples is directly related to time by the
* following formula: \code ms = (samples*1000)/freq \endcode
* - \c desired->size is the size in bytes of the audio buffer, and is
* calculated by SDL_OpenAudio().
* - \c desired->silence is the value used to set the buffer to silence,
* and is calculated by SDL_OpenAudio().
* - \c 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_LockAudio()
* and SDL_UnlockAudio() 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).
* - \c desired->userdata is passed as the first parameter to your callback
* function. If you passed a NULL callback, this value is ignored.
*
* The audio device starts out playing silence when it's opened, and should
* be enabled for playing by calling \c SDL_PauseAudio(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.
*/
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 available devices exposed by the current driver.
* Only valid after a successfully initializing the audio subsystem.
* Returns -1 if an explicit list of devices can't be determined; this 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 to 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()).
*/
extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
/**
* Get the human-readable name of a specific audio device.
* Must be a value between 0 and (number of audio devices-1).
* Only valid after a successfully initializing the audio subsystem.
* The values returned by this function reflect the latest call to
* SDL_GetNumAudioDevices(); recall 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 is called.
*/
extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
int iscapture);
/**
* Open a specific audio device. Passing in a device name of NULL requests
* the most reasonable default (and is equivalent to calling SDL_OpenAudio()).
*
* 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.
*
* \return 0 on error, a valid device ID that is >= 2 on success.
*
* SDL_OpenAudio(), unlike this function, always acts on device ID 1.
*/
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 */
/**
* \brief Load the audio data of a WAVE file into memory
*
* Loading a WAVE file requires \c src, \c spec, \c audio_buf and \c audio_len
* to be valid pointers. The entire data portion of the file is then loaded
* into memory and decoded if necessary.
*
* If \c freesrc is non-zero, the data source gets automatically closed and
* freed before the function returns.
*
* Supported 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
* µ-law (8 bits). Other formats are currently unsupported and cause an error.
*
* If this function succeeds, the pointer returned by it is equal to \c spec
* and the pointer to the audio data allocated by the function is written to
* \c audio_buf and its length in bytes to \c audio_len. The \ref SDL_AudioSpec
* members \c freq, \c channels, and \c format are set to the values of the
* audio data in the buffer. The \c 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
* \c audio_buf when it is no longer used.
*
* Because of the underspecification of the Waveform 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 \c src being NULL), an appropriate error message will be set.
*
* It is required that the data source supports seeking.
*
* Example:
* \code
* SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
* \endcode
*
* \param src The data source with the WAVE data
* \param freesrc A integer value that makes the function close the data source if non-zero
* \param spec A pointer filled with the audio format of the audio data
* \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
* \return NULL on error, or non-NULL on success.
*/
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)
/**
* This function frees data previously allocated with SDL_LoadWAV_RW()
*/
extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
/**
* This function takes a source format and rate and a destination format
* and rate, and initializes the \c cvt structure with information needed
* by SDL_ConvertAudio() to convert a buffer of audio data from one format
* to the other. An unsupported format causes an error and -1 will be returned.
*
* \return 0 if no conversion is needed, 1 if the audio filter is set up,
* or -1 on error.
*/
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);
/**
* Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(),
* created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of
* audio data in the source format, this function will convert it in-place
* to the desired format.
*
* The data conversion may expand the size of the audio data, so the buffer
* \c cvt->buf should be allocated after the \c cvt structure is initialized by
* SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long.
*
* \return 0 on success or -1 if \c cvt->buf is NULL.
*/
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
* \return 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
* \return 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
* \return 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 takes two audio buffers of the playing audio format and mixes
* them, performing addition, volume adjustment, and overflow clipping.
* The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME
* for full audio volume. Note this does not change hardware volume.
* This is provided for convenience -- you can mix your own audio data.
*/
extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
Uint32 len, int volume);
/**
* This works like SDL_MixAudio(), but you specify the audio format instead of
* using the format of audio device 1. Thus it can be used when no audio
* device is open at all.
*/
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. This 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.
*
* \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.
* \return 0 on success, or -1 on error.
*
* \sa SDL_GetQueuedAudioSize
* \sa SDL_ClearQueuedAudio
*/
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. This 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 dequeued 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 queueing; 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.
* \return number of bytes dequeued, which could be less than requested.
*
* \sa SDL_GetQueuedAudioSize
* \sa SDL_ClearQueuedAudio
*/
extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
/**
* Get the number of bytes of still-queued audio.
*
* For playback device:
*
* This is the number of bytes that have been queued for playback with
* SDL_QueueAudio(), but have not yet been sent to the hardware. This
* number may shrink at any time, so this only informs of pending data.
*
* 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 audio on a device that is using an application-supplied
* callback; calling this function on such a device always returns 0.
* You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
* the audio callback, 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.
* \return Number of bytes (not samples!) of queued audio.
*
* \sa SDL_QueueAudio
* \sa SDL_ClearQueuedAudio
*/
extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
/**
* Drop any queued audio data. For playback devices, this is any queued data
* still waiting to be submitted to the hardware. For capture devices, this
* is any data that was queued by the device that hasn't yet been dequeued by
* the application.
*
* Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
* playback devices, the hardware will start playing silence if more audio
* isn't queued. Unpaused capture devices will start filling the queue again
* as soon as they have more data available (which, depending on the state
* of the hardware and the thread, could be before this function call
* returns!).
*
* 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 during a level change
* in your game.
*
* You may not queue audio on a device that is using an application-supplied
* callback; calling this function on such a device is always a no-op.
* You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
* the audio callback, 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.
*
* \sa SDL_QueueAudio
* \sa SDL_GetQueuedAudioSize
*/
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 shuts down audio processing and closes the audio device.
*/
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
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: */
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