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dolphin/Externals/SDL/include/SDL_audio.h
John Peterson 088440ab12 Externals > renamed SDL dir to match their SVN names 
git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@4537 8ced0084-cf51-0410-be5f-012b33b47a6e
2009-11-12 19:07:17 +00:00

506 lines
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C
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/*
SDL - Simple DirectMedia Layer
Copyright (C) 1997-2009 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
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
Lesser General Public License for more details.
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
slouken@libsdl.org
*/
/**
* \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
/* *INDENT-OFF* */
extern "C" {
/* *INDENT-ON* */
#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 */
#define AUDIO_S32 AUDIO_S32LSB
/*@}*/
/**
* \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 */
#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_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE)
/*@}*/
/*@}*//*Audio flags*/
/**
* 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) */
/**
* This function is called when the audio device needs more data.
*
* \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.
*/
void (SDLCALL * callback) (void *userdata, Uint8 * stream, int len);
void *userdata;
} SDL_AudioSpec;
struct SDL_AudioCVT;
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.
*/
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.
* - \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.
*/
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*/
/**
* 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()
*/
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.
*
* \return -1 if the format conversion is not supported, 0 if there's
* no conversion needed, or 1 if the audio filter is set up.
*/
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.
*/
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
#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.
*/
/*@{*/
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);
/**
* \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
/* *INDENT-OFF* */
}
/* *INDENT-ON* */
#endif
#include "close_code.h"
#endif /* _SDL_audio_h */
/* vi: set ts=4 sw=4 expandtab: */
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