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00001 /* 00002 Simple DirectMedia Layer 00003 Copyright (C) 1997-2014 Sam Lantinga <slouken@libsdl.org> 00004 00005 This software is provided 'as-is', without any express or implied 00006 warranty. In no event will the authors be held liable for any damages 00007 arising from the use of this software. 00008 00009 Permission is granted to anyone to use this software for any purpose, 00010 including commercial applications, and to alter it and redistribute it 00011 freely, subject to the following restrictions: 00012 00013 1. The origin of this software must not be misrepresented; you must not 00014 claim that you wrote the original software. If you use this software 00015 in a product, an acknowledgment in the product documentation would be 00016 appreciated but is not required. 00017 2. Altered source versions must be plainly marked as such, and must not be 00018 misrepresented as being the original software. 00019 3. This notice may not be removed or altered from any source distribution. 00020 */ 00021 00022 /** 00023 * \file SDL_audio.h 00024 * 00025 * Access to the raw audio mixing buffer for the SDL library. 00026 */ 00027 00028 #ifndef _SDL_audio_h 00029 #define _SDL_audio_h 00030 00031 #include "SDL_stdinc.h" 00032 #include "SDL_error.h" 00033 #include "SDL_endian.h" 00034 #include "SDL_mutex.h" 00035 #include "SDL_thread.h" 00036 #include "SDL_rwops.h" 00037 00038 #include "begin_code.h" 00039 /* Set up for C function definitions, even when using C++ */ 00040 #ifdef __cplusplus 00041 extern "C" { 00042 #endif 00043 00044 /** 00045 * \brief Audio format flags. 00046 * 00047 * These are what the 16 bits in SDL_AudioFormat currently mean... 00048 * (Unspecified bits are always zero). 00049 * 00050 * \verbatim 00051 ++-----------------------sample is signed if set 00052 || 00053 || ++-----------sample is bigendian if set 00054 || || 00055 || || ++---sample is float if set 00056 || || || 00057 || || || +---sample bit size---+ 00058 || || || | | 00059 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00 00060 \endverbatim 00061 * 00062 * There are macros in SDL 2.0 and later to query these bits. 00063 */ 00064 typedef Uint16 SDL_AudioFormat; 00065 00066 /** 00067 * \name Audio flags 00068 */ 00069 /* @{ */ 00070 00071 #define SDL_AUDIO_MASK_BITSIZE (0xFF) 00072 #define SDL_AUDIO_MASK_DATATYPE (1<<8) 00073 #define SDL_AUDIO_MASK_ENDIAN (1<<12) 00074 #define SDL_AUDIO_MASK_SIGNED (1<<15) 00075 #define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE) 00076 #define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE) 00077 #define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN) 00078 #define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED) 00079 #define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x)) 00080 #define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x)) 00081 #define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x)) 00082 00083 /** 00084 * \name Audio format flags 00085 * 00086 * Defaults to LSB byte order. 00087 */ 00088 /* @{ */ 00089 #define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */ 00090 #define AUDIO_S8 0x8008 /**< Signed 8-bit samples */ 00091 #define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */ 00092 #define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */ 00093 #define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */ 00094 #define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */ 00095 #define AUDIO_U16 AUDIO_U16LSB 00096 #define AUDIO_S16 AUDIO_S16LSB 00097 /* @} */ 00098 00099 /** 00100 * \name int32 support 00101 */ 00102 /* @{ */ 00103 #define AUDIO_S32LSB 0x8020 /**< 32-bit integer samples */ 00104 #define AUDIO_S32MSB 0x9020 /**< As above, but big-endian byte order */ 00105 #define AUDIO_S32 AUDIO_S32LSB 00106 /* @} */ 00107 00108 /** 00109 * \name float32 support 00110 */ 00111 /* @{ */ 00112 #define AUDIO_F32LSB 0x8120 /**< 32-bit floating point samples */ 00113 #define AUDIO_F32MSB 0x9120 /**< As above, but big-endian byte order */ 00114 #define AUDIO_F32 AUDIO_F32LSB 00115 /* @} */ 00116 00117 /** 00118 * \name Native audio byte ordering 00119 */ 00120 /* @{ */ 00121 #if SDL_BYTEORDER == SDL_LIL_ENDIAN 00122 #define AUDIO_U16SYS AUDIO_U16LSB 00123 #define AUDIO_S16SYS AUDIO_S16LSB 00124 #define AUDIO_S32SYS AUDIO_S32LSB 00125 #define AUDIO_F32SYS AUDIO_F32LSB 00126 #else 00127 #define AUDIO_U16SYS AUDIO_U16MSB 00128 #define AUDIO_S16SYS AUDIO_S16MSB 00129 #define AUDIO_S32SYS AUDIO_S32MSB 00130 #define AUDIO_F32SYS AUDIO_F32MSB 00131 #endif 00132 /* @} */ 00133 00134 /** 00135 * \name Allow change flags 00136 * 00137 * Which audio format changes are allowed when opening a device. 00138 */ 00139 /* @{ */ 00140 #define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001 00141 #define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002 00142 #define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004 00143 #define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE) 00144 /* @} */ 00145 00146 /* @} *//* Audio flags */ 00147 00148 /** 00149 * This function is called when the audio device needs more data. 00150 * 00151 * \param userdata An application-specific parameter saved in 00152 * the SDL_AudioSpec structure 00153 * \param stream A pointer to the audio data buffer. 00154 * \param len The length of that buffer in bytes. 00155 * 00156 * Once the callback returns, the buffer will no longer be valid. 00157 * Stereo samples are stored in a LRLRLR ordering. 00158 */ 00159 typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream, 00160 int len); 00161 00162 /** 00163 * The calculated values in this structure are calculated by SDL_OpenAudio(). 00164 */ 00165 typedef struct SDL_AudioSpec 00166 { 00167 int freq; /**< DSP frequency -- samples per second */ 00168 SDL_AudioFormat format; /**< Audio data format */ 00169 Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */ 00170 Uint8 silence; /**< Audio buffer silence value (calculated) */ 00171 Uint16 samples; /**< Audio buffer size in samples (power of 2) */ 00172 Uint16 padding; /**< Necessary for some compile environments */ 00173 Uint32 size; /**< Audio buffer size in bytes (calculated) */ 00174 SDL_AudioCallback callback; 00175 void *userdata; 00176 } SDL_AudioSpec; 00177 00178 00179 struct SDL_AudioCVT; 00180 typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt, 00181 SDL_AudioFormat format); 00182 00183 /** 00184 * A structure to hold a set of audio conversion filters and buffers. 00185 */ 00186 #ifdef __GNUC__ 00187 /* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't 00188 pad it out to 88 bytes to guarantee ABI compatibility between compilers. 00189 vvv 00190 The next time we rev the ABI, make sure to size the ints and add padding. 00191 */ 00192 #define SDL_AUDIOCVT_PACKED __attribute__((packed)) 00193 #else 00194 #define SDL_AUDIOCVT_PACKED 00195 #endif 00196 /* */ 00197 typedef struct SDL_AudioCVT 00198 { 00199 int needed; /**< Set to 1 if conversion possible */ 00200 SDL_AudioFormat src_format; /**< Source audio format */ 00201 SDL_AudioFormat dst_format; /**< Target audio format */ 00202 double rate_incr; /**< Rate conversion increment */ 00203 Uint8 *buf; /**< Buffer to hold entire audio data */ 00204 int len; /**< Length of original audio buffer */ 00205 int len_cvt; /**< Length of converted audio buffer */ 00206 int len_mult; /**< buffer must be len*len_mult big */ 00207 double len_ratio; /**< Given len, final size is len*len_ratio */ 00208 SDL_AudioFilter filters[10]; /**< Filter list */ 00209 int filter_index; /**< Current audio conversion function */ 00210 } SDL_AUDIOCVT_PACKED SDL_AudioCVT; 00211 00212 00213 /* Function prototypes */ 00214 00215 /** 00216 * \name Driver discovery functions 00217 * 00218 * These functions return the list of built in audio drivers, in the 00219 * order that they are normally initialized by default. 00220 */ 00221 /* @{ */ 00222 extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void); 00223 extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index); 00224 /* @} */ 00225 00226 /** 00227 * \name Initialization and cleanup 00228 * 00229 * \internal These functions are used internally, and should not be used unless 00230 * you have a specific need to specify the audio driver you want to 00231 * use. You should normally use SDL_Init() or SDL_InitSubSystem(). 00232 */ 00233 /* @{ */ 00234 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); 00235 extern DECLSPEC void SDLCALL SDL_AudioQuit(void); 00236 /* @} */ 00237 00238 /** 00239 * This function returns the name of the current audio driver, or NULL 00240 * if no driver has been initialized. 00241 */ 00242 extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void); 00243 00244 /** 00245 * This function opens the audio device with the desired parameters, and 00246 * returns 0 if successful, placing the actual hardware parameters in the 00247 * structure pointed to by \c obtained. If \c obtained is NULL, the audio 00248 * data passed to the callback function will be guaranteed to be in the 00249 * requested format, and will be automatically converted to the hardware 00250 * audio format if necessary. This function returns -1 if it failed 00251 * to open the audio device, or couldn't set up the audio thread. 00252 * 00253 * When filling in the desired audio spec structure, 00254 * - \c desired->freq should be the desired audio frequency in samples-per- 00255 * second. 00256 * - \c desired->format should be the desired audio format. 00257 * - \c desired->samples is the desired size of the audio buffer, in 00258 * samples. This number should be a power of two, and may be adjusted by 00259 * the audio driver to a value more suitable for the hardware. Good values 00260 * seem to range between 512 and 8096 inclusive, depending on the 00261 * application and CPU speed. Smaller values yield faster response time, 00262 * but can lead to underflow if the application is doing heavy processing 00263 * and cannot fill the audio buffer in time. A stereo sample consists of 00264 * both right and left channels in LR ordering. 00265 * Note that the number of samples is directly related to time by the 00266 * following formula: \code ms = (samples*1000)/freq \endcode 00267 * - \c desired->size is the size in bytes of the audio buffer, and is 00268 * calculated by SDL_OpenAudio(). 00269 * - \c desired->silence is the value used to set the buffer to silence, 00270 * and is calculated by SDL_OpenAudio(). 00271 * - \c desired->callback should be set to a function that will be called 00272 * when the audio device is ready for more data. It is passed a pointer 00273 * to the audio buffer, and the length in bytes of the audio buffer. 00274 * This function usually runs in a separate thread, and so you should 00275 * protect data structures that it accesses by calling SDL_LockAudio() 00276 * and SDL_UnlockAudio() in your code. 00277 * - \c desired->userdata is passed as the first parameter to your callback 00278 * function. 00279 * 00280 * The audio device starts out playing silence when it's opened, and should 00281 * be enabled for playing by calling \c SDL_PauseAudio(0) when you are ready 00282 * for your audio callback function to be called. Since the audio driver 00283 * may modify the requested size of the audio buffer, you should allocate 00284 * any local mixing buffers after you open the audio device. 00285 */ 00286 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired, 00287 SDL_AudioSpec * obtained); 00288 00289 /** 00290 * SDL Audio Device IDs. 00291 * 00292 * A successful call to SDL_OpenAudio() is always device id 1, and legacy 00293 * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls 00294 * always returns devices >= 2 on success. The legacy calls are good both 00295 * for backwards compatibility and when you don't care about multiple, 00296 * specific, or capture devices. 00297 */ 00298 typedef Uint32 SDL_AudioDeviceID; 00299 00300 /** 00301 * Get the number of available devices exposed by the current driver. 00302 * Only valid after a successfully initializing the audio subsystem. 00303 * Returns -1 if an explicit list of devices can't be determined; this is 00304 * not an error. For example, if SDL is set up to talk to a remote audio 00305 * server, it can't list every one available on the Internet, but it will 00306 * still allow a specific host to be specified to SDL_OpenAudioDevice(). 00307 * 00308 * In many common cases, when this function returns a value <= 0, it can still 00309 * successfully open the default device (NULL for first argument of 00310 * SDL_OpenAudioDevice()). 00311 */ 00312 extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture); 00313 00314 /** 00315 * Get the human-readable name of a specific audio device. 00316 * Must be a value between 0 and (number of audio devices-1). 00317 * Only valid after a successfully initializing the audio subsystem. 00318 * The values returned by this function reflect the latest call to 00319 * SDL_GetNumAudioDevices(); recall that function to redetect available 00320 * hardware. 00321 * 00322 * The string returned by this function is UTF-8 encoded, read-only, and 00323 * managed internally. You are not to free it. If you need to keep the 00324 * string for any length of time, you should make your own copy of it, as it 00325 * will be invalid next time any of several other SDL functions is called. 00326 */ 00327 extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index, 00328 int iscapture); 00329 00330 00331 /** 00332 * Open a specific audio device. Passing in a device name of NULL requests 00333 * the most reasonable default (and is equivalent to calling SDL_OpenAudio()). 00334 * 00335 * The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but 00336 * some drivers allow arbitrary and driver-specific strings, such as a 00337 * hostname/IP address for a remote audio server, or a filename in the 00338 * diskaudio driver. 00339 * 00340 * \return 0 on error, a valid device ID that is >= 2 on success. 00341 * 00342 * SDL_OpenAudio(), unlike this function, always acts on device ID 1. 00343 */ 00344 extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char 00345 *device, 00346 int iscapture, 00347 const 00348 SDL_AudioSpec * 00349 desired, 00350 SDL_AudioSpec * 00351 obtained, 00352 int 00353 allowed_changes); 00354 00355 00356 00357 /** 00358 * \name Audio state 00359 * 00360 * Get the current audio state. 00361 */ 00362 /* @{ */ 00363 typedef enum 00364 { 00365 SDL_AUDIO_STOPPED = 0, 00366 SDL_AUDIO_PLAYING, 00367 SDL_AUDIO_PAUSED 00368 } SDL_AudioStatus; 00369 extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void); 00370 00371 extern DECLSPEC SDL_AudioStatus SDLCALL 00372 SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev); 00373 /* @} *//* Audio State */ 00374 00375 /** 00376 * \name Pause audio functions 00377 * 00378 * These functions pause and unpause the audio callback processing. 00379 * They should be called with a parameter of 0 after opening the audio 00380 * device to start playing sound. This is so you can safely initialize 00381 * data for your callback function after opening the audio device. 00382 * Silence will be written to the audio device during the pause. 00383 */ 00384 /* @{ */ 00385 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); 00386 extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev, 00387 int pause_on); 00388 /* @} *//* Pause audio functions */ 00389 00390 /** 00391 * This function loads a WAVE from the data source, automatically freeing 00392 * that source if \c freesrc is non-zero. For example, to load a WAVE file, 00393 * you could do: 00394 * \code 00395 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); 00396 * \endcode 00397 * 00398 * If this function succeeds, it returns the given SDL_AudioSpec, 00399 * filled with the audio data format of the wave data, and sets 00400 * \c *audio_buf to a malloc()'d buffer containing the audio data, 00401 * and sets \c *audio_len to the length of that audio buffer, in bytes. 00402 * You need to free the audio buffer with SDL_FreeWAV() when you are 00403 * done with it. 00404 * 00405 * This function returns NULL and sets the SDL error message if the 00406 * wave file cannot be opened, uses an unknown data format, or is 00407 * corrupt. Currently raw and MS-ADPCM WAVE files are supported. 00408 */ 00409 extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, 00410 int freesrc, 00411 SDL_AudioSpec * spec, 00412 Uint8 ** audio_buf, 00413 Uint32 * audio_len); 00414 00415 /** 00416 * Loads a WAV from a file. 00417 * Compatibility convenience function. 00418 */ 00419 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ 00420 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) 00421 00422 /** 00423 * This function frees data previously allocated with SDL_LoadWAV_RW() 00424 */ 00425 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf); 00426 00427 /** 00428 * This function takes a source format and rate and a destination format 00429 * and rate, and initializes the \c cvt structure with information needed 00430 * by SDL_ConvertAudio() to convert a buffer of audio data from one format 00431 * to the other. 00432 * 00433 * \return -1 if the format conversion is not supported, 0 if there's 00434 * no conversion needed, or 1 if the audio filter is set up. 00435 */ 00436 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt, 00437 SDL_AudioFormat src_format, 00438 Uint8 src_channels, 00439 int src_rate, 00440 SDL_AudioFormat dst_format, 00441 Uint8 dst_channels, 00442 int dst_rate); 00443 00444 /** 00445 * Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(), 00446 * created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of 00447 * audio data in the source format, this function will convert it in-place 00448 * to the desired format. 00449 * 00450 * The data conversion may expand the size of the audio data, so the buffer 00451 * \c cvt->buf should be allocated after the \c cvt structure is initialized by 00452 * SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long. 00453 */ 00454 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt); 00455 00456 #define SDL_MIX_MAXVOLUME 128 00457 /** 00458 * This takes two audio buffers of the playing audio format and mixes 00459 * them, performing addition, volume adjustment, and overflow clipping. 00460 * The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME 00461 * for full audio volume. Note this does not change hardware volume. 00462 * This is provided for convenience -- you can mix your own audio data. 00463 */ 00464 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src, 00465 Uint32 len, int volume); 00466 00467 /** 00468 * This works like SDL_MixAudio(), but you specify the audio format instead of 00469 * using the format of audio device 1. Thus it can be used when no audio 00470 * device is open at all. 00471 */ 00472 extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst, 00473 const Uint8 * src, 00474 SDL_AudioFormat format, 00475 Uint32 len, int volume); 00476 00477 /** 00478 * \name Audio lock functions 00479 * 00480 * The lock manipulated by these functions protects the callback function. 00481 * During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that 00482 * the callback function is not running. Do not call these from the callback 00483 * function or you will cause deadlock. 00484 */ 00485 /* @{ */ 00486 extern DECLSPEC void SDLCALL SDL_LockAudio(void); 00487 extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev); 00488 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); 00489 extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev); 00490 /* @} *//* Audio lock functions */ 00491 00492 /** 00493 * This function shuts down audio processing and closes the audio device. 00494 */ 00495 extern DECLSPEC void SDLCALL SDL_CloseAudio(void); 00496 extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev); 00497 00498 /* Ends C function definitions when using C++ */ 00499 #ifdef __cplusplus 00500 } 00501 #endif 00502 #include "close_code.h" 00503 00504 #endif /* _SDL_audio_h */ 00505 00506 /* vi: set ts=4 sw=4 expandtab: */
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