424 lines
18 KiB
C
424 lines
18 KiB
C
/*
|
|
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
|
|
|
|
typedef Uint16 SDL_AudioFormat;
|
|
|
|
/* 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.
|
|
'stream' is a pointer to the audio data buffer
|
|
'len' is 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;
|
|
|
|
|
|
/*
|
|
These are what the 16 bits in SDL_AudioFormat currently mean...
|
|
(Unspecified bits are always zero.)
|
|
|
|
++-----------------------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
|
|
|
|
There are macros in SDL 1.3 and later to query these bits.
|
|
*/
|
|
|
|
#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))
|
|
|
|
/* 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
|
|
|
|
/* 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
|
|
|
|
/* 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
|
|
|
|
/* 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
|
|
|
|
/* 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)
|
|
|
|
/* A structure to hold a set of audio conversion filters and buffers */
|
|
struct SDL_AudioCVT;
|
|
typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
|
|
SDL_AudioFormat format);
|
|
|
|
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 */
|
|
|
|
/* 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);
|
|
|
|
/* 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 '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 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,
|
|
* 'desired->freq' should be the desired audio frequency in samples-per-second.
|
|
* 'desired->format' should be the desired audio format.
|
|
* '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: ms = (samples*1000)/freq
|
|
* 'desired->size' is the size in bytes of the audio buffer, and is
|
|
* calculated by SDL_OpenAudio().
|
|
* 'desired->silence' is the value used to set the buffer to silence,
|
|
* and is calculated by SDL_OpenAudio().
|
|
* '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.
|
|
* '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 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_GetAudioDevice(), 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.
|
|
* Returns 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);
|
|
|
|
|
|
|
|
/*
|
|
* 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);
|
|
|
|
/*
|
|
* This function pauses and unpauses the audio callback processing.
|
|
* It 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);
|
|
|
|
/*
|
|
* This function loads a WAVE from the data source, automatically freeing
|
|
* that source if 'freesrc' is non-zero. For example, to load a WAVE file,
|
|
* you could do:
|
|
* SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
|
|
*
|
|
* If this function succeeds, it returns the given SDL_AudioSpec,
|
|
* filled with the audio data format of the wave data, and sets
|
|
* 'audio_buf' to a malloc()'d buffer containing the audio data,
|
|
* and sets '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);
|
|
|
|
/* Compatibility convenience function -- loads a WAV from a file */
|
|
#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 'cvt' structure with information needed
|
|
* by SDL_ConvertAudio() to convert a buffer of audio data from one format
|
|
* to the other.
|
|
* Returns -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 'cvt' structure using SDL_BuildAudioCVT(),
|
|
* created an audio buffer cvt->buf, and filled it with 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
|
|
* cvt->buf should be allocated after the cvt structure is initialized by
|
|
* SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.
|
|
*/
|
|
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
|
|
|
|
/*
|
|
* 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.
|
|
*/
|
|
#define SDL_MIX_MAXVOLUME 128
|
|
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);
|
|
|
|
/*
|
|
* The lock manipulated by these functions protects the callback function.
|
|
* During a LockAudio/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);
|
|
|
|
/*
|
|
* 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);
|
|
|
|
/*
|
|
* Returns 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: */
|