/*

SDL - Simple DirectMedia Layer

This library is free software; you can redistribute it and/or

This library is distributed in the hope that it will be useful,

but WITHOUT ANY WARRANTY; without even the implied warranty of

MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

You should have received a copy of the GNU Lesser General Public

License along with this library; if not, write to the Free Software

Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA

Sam Lantinga

*/

* \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_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

#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 1.3 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

*

* New to SDL 1.3.

*/

/*@{*/

#define AUDIO_S32LSB 0x8020 /**< 32-bit integer samples */

#define AUDIO_S32MSB 0x9020 /**< As above, but big-endian byte order */

/**

* \name float32 support

*

* New to SDL 1.3.

*/

/*@{*/

#define AUDIO_F32LSB 0x8120 /**< 32-bit floating point samples */

#define AUDIO_F32MSB 0x9120 /**< As above, but big-endian byte order */

/**

* \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

/**

* \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_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_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.

*/

typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,

int len);

/**

* The calculated values in this structure are calculated by SDL_OpenAudio().

*/

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 samples (power of 2) */

Uint16 padding; /**< Necessary for some compile environments */

Uint32 size; /**< Audio buffer size in bytes (calculated) */

void *userdata;

} SDL_AudioSpec;

typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,

SDL_AudioFormat format);

/**

* A structure to hold a set of audio conversion filters and buffers.

*/

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[10]; /**< Filter list */

int filter_index; /**< Current audio conversion function */

} 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.

/**

* 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.

* - \c desired->userdata is passed as the first parameter to your callback

* function.

*

* 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.

/**

* 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.

* 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 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.

*

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);

/**

* \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_PauseAudioDevice(SDL_AudioDeviceID dev,

int pause_on);

/**

* This function loads a WAVE from the data source, automatically freeing

* that source if \c freesrc is non-zero. For example, to load a WAVE file,

* you could do:

* \code

* SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);

* \endcode

* If this function succeeds, it returns the given SDL_AudioSpec,

* filled with the audio data format of the wave data, and sets

* \c *audio_buf to a malloc()'d buffer containing the audio data,

* and sets \c *audio_len to the length of that audio buffer, in bytes.

* You need to free the audio buffer with SDL_FreeWAV() when you are

* done with it.

* This function returns NULL and sets the SDL error message if the

* wave file cannot be opened, uses an unknown data format, or is

* corrupt. Currently raw and MS-ADPCM WAVE files are supported.

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()

/**

* 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.

*

* \return -1 if the format conversion is not supported, 0 if there's

Uint8 src_channels,

int src_rate,

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.

#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);

/**

* \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 shuts down audio processing and closes the audio device.

extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);

/**

* \return 1 if audio device is still functioning, zero if not, -1 on error.

*/

extern DECLSPEC int SDLCALL SDL_AudioDeviceConnected(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: */