Mercurial > sdl-ios-xcode
annotate include/SDL_audio.h @ 4286:9ea4413f0a9e SDL-1.2
Fixed bug #831
Andrey 2009-10-09 08:08:04 PDT
gapi_hires_fix
I made a mistake with the conclusions.
And last time I as was mistaken with conclusions.
Now I thought about in more detail the algorithm works.
I have made it that the project fheroes2 well worked.
The result of my work in attachment.
I have a possibility to check up it on different models pocketpc.
I checked various modes screen of 320x320, 320x240, 640x480, 800x400.
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Sat, 10 Oct 2009 02:03:53 +0000 |
| parents | 4c4113c2162c |
| children |
| rev | line source |
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| 0 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
| 4159 | 3 Copyright (C) 1997-2009 Sam Lantinga |
| 0 | 4 |
| 5 This library is free software; you can redistribute it and/or | |
|
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6 modify it under the terms of the GNU Lesser General Public |
| 0 | 7 License as published by the Free Software Foundation; either |
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8 version 2.1 of the License, or (at your option) any later version. |
| 0 | 9 |
| 10 This library is distributed in the hope that it will be useful, | |
| 11 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
| 12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
|
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13 Lesser General Public License for more details. |
| 0 | 14 |
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15 You should have received a copy of the GNU Lesser General Public |
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16 License along with this library; if not, write to the Free Software |
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17 Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA |
| 0 | 18 |
| 19 Sam Lantinga | |
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20 slouken@libsdl.org |
| 0 | 21 */ |
| 22 | |
| 4217 | 23 /** |
| 24 * @file SDL_audio.h | |
| 25 * Access to the raw audio mixing buffer for the SDL library | |
| 26 */ | |
| 0 | 27 |
| 28 #ifndef _SDL_audio_h | |
| 29 #define _SDL_audio_h | |
| 30 | |
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31 #include "SDL_stdinc.h" |
| 0 | 32 #include "SDL_error.h" |
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33 #include "SDL_endian.h" |
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34 #include "SDL_mutex.h" |
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35 #include "SDL_thread.h" |
| 0 | 36 #include "SDL_rwops.h" |
| 37 | |
| 38 #include "begin_code.h" | |
| 39 /* Set up for C function definitions, even when using C++ */ | |
| 40 #ifdef __cplusplus | |
| 41 extern "C" { | |
| 42 #endif | |
| 43 | |
| 4217 | 44 /** |
| 0 | 45 * When filling in the desired audio spec structure, |
| 4217 | 46 * - 'desired->freq' should be the desired audio frequency in samples-per-second. |
| 47 * - 'desired->format' should be the desired audio format. | |
| 48 * - 'desired->samples' is the desired size of the audio buffer, in samples. | |
| 0 | 49 * This number should be a power of two, and may be adjusted by the audio |
| 50 * driver to a value more suitable for the hardware. Good values seem to | |
| 51 * range between 512 and 8096 inclusive, depending on the application and | |
| 52 * CPU speed. Smaller values yield faster response time, but can lead | |
| 53 * to underflow if the application is doing heavy processing and cannot | |
| 54 * fill the audio buffer in time. A stereo sample consists of both right | |
| 55 * and left channels in LR ordering. | |
| 56 * Note that the number of samples is directly related to time by the | |
| 57 * following formula: ms = (samples*1000)/freq | |
| 4217 | 58 * - 'desired->size' is the size in bytes of the audio buffer, and is |
| 0 | 59 * calculated by SDL_OpenAudio(). |
| 4217 | 60 * - 'desired->silence' is the value used to set the buffer to silence, |
| 0 | 61 * and is calculated by SDL_OpenAudio(). |
| 4217 | 62 * - 'desired->callback' should be set to a function that will be called |
| 0 | 63 * when the audio device is ready for more data. It is passed a pointer |
| 64 * to the audio buffer, and the length in bytes of the audio buffer. | |
| 65 * This function usually runs in a separate thread, and so you should | |
| 66 * protect data structures that it accesses by calling SDL_LockAudio() | |
| 67 * and SDL_UnlockAudio() in your code. | |
| 4217 | 68 * - 'desired->userdata' is passed as the first parameter to your callback |
| 0 | 69 * function. |
| 70 * | |
| 4217 | 71 * @note The calculated values in this structure are calculated by SDL_OpenAudio() |
| 72 * | |
| 73 */ | |
| 74 typedef struct SDL_AudioSpec { | |
| 75 int freq; /**< DSP frequency -- samples per second */ | |
| 76 Uint16 format; /**< Audio data format */ | |
| 77 Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */ | |
| 78 Uint8 silence; /**< Audio buffer silence value (calculated) */ | |
| 79 Uint16 samples; /**< Audio buffer size in samples (power of 2) */ | |
| 80 Uint16 padding; /**< Necessary for some compile environments */ | |
| 81 Uint32 size; /**< Audio buffer size in bytes (calculated) */ | |
| 82 /** | |
| 83 * This function is called when the audio device needs more data. | |
| 84 * | |
| 85 * @param[out] stream A pointer to the audio data buffer | |
| 86 * @param[in] len The length of the audio buffer in bytes. | |
| 87 * | |
| 88 * Once the callback returns, the buffer will no longer be valid. | |
| 89 * Stereo samples are stored in a LRLRLR ordering. | |
| 90 */ | |
| 91 void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len); | |
| 92 void *userdata; | |
| 93 } SDL_AudioSpec; | |
| 94 | |
| 95 /** | |
| 96 * @name Audio format flags | |
| 97 * defaults to LSB byte order | |
| 98 */ | |
| 99 /*@{*/ | |
| 100 #define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */ | |
| 101 #define AUDIO_S8 0x8008 /**< Signed 8-bit samples */ | |
| 102 #define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */ | |
| 103 #define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */ | |
| 104 #define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */ | |
| 105 #define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */ | |
| 106 #define AUDIO_U16 AUDIO_U16LSB | |
| 107 #define AUDIO_S16 AUDIO_S16LSB | |
| 108 | |
| 109 /** | |
| 110 * @name Native audio byte ordering | |
| 111 */ | |
| 112 /*@{*/ | |
| 113 #if SDL_BYTEORDER == SDL_LIL_ENDIAN | |
| 114 #define AUDIO_U16SYS AUDIO_U16LSB | |
| 115 #define AUDIO_S16SYS AUDIO_S16LSB | |
| 116 #else | |
| 117 #define AUDIO_U16SYS AUDIO_U16MSB | |
| 118 #define AUDIO_S16SYS AUDIO_S16MSB | |
| 119 #endif | |
| 120 /*@}*/ | |
| 121 | |
| 122 /*@}*/ | |
| 123 | |
| 124 | |
| 125 /** A structure to hold a set of audio conversion filters and buffers */ | |
| 126 typedef struct SDL_AudioCVT { | |
| 127 int needed; /**< Set to 1 if conversion possible */ | |
| 128 Uint16 src_format; /**< Source audio format */ | |
| 129 Uint16 dst_format; /**< Target audio format */ | |
| 130 double rate_incr; /**< Rate conversion increment */ | |
| 131 Uint8 *buf; /**< Buffer to hold entire audio data */ | |
| 132 int len; /**< Length of original audio buffer */ | |
| 133 int len_cvt; /**< Length of converted audio buffer */ | |
| 134 int len_mult; /**< buffer must be len*len_mult big */ | |
| 135 double len_ratio; /**< Given len, final size is len*len_ratio */ | |
| 136 void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format); | |
| 137 int filter_index; /**< Current audio conversion function */ | |
| 138 } SDL_AudioCVT; | |
| 139 | |
| 140 | |
| 141 /* Function prototypes */ | |
| 142 | |
| 143 /** | |
| 144 * @name Audio Init and Quit | |
| 145 * These functions are used internally, and should not be used unless you | |
| 146 * have a specific need to specify the audio driver you want to use. | |
| 147 * You should normally use SDL_Init() or SDL_InitSubSystem(). | |
| 148 */ | |
| 149 /*@{*/ | |
| 150 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); | |
| 151 extern DECLSPEC void SDLCALL SDL_AudioQuit(void); | |
| 152 /*@}*/ | |
| 153 | |
| 154 /** | |
| 155 * This function fills the given character buffer with the name of the | |
| 156 * current audio driver, and returns a pointer to it if the audio driver has | |
| 157 * been initialized. It returns NULL if no driver has been initialized. | |
| 158 */ | |
| 159 extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen); | |
| 160 | |
| 161 /** | |
| 162 * This function opens the audio device with the desired parameters, and | |
| 163 * returns 0 if successful, placing the actual hardware parameters in the | |
| 164 * structure pointed to by 'obtained'. If 'obtained' is NULL, the audio | |
| 165 * data passed to the callback function will be guaranteed to be in the | |
| 166 * requested format, and will be automatically converted to the hardware | |
| 167 * audio format if necessary. This function returns -1 if it failed | |
| 168 * to open the audio device, or couldn't set up the audio thread. | |
| 169 * | |
| 0 | 170 * The audio device starts out playing silence when it's opened, and should |
| 171 * be enabled for playing by calling SDL_PauseAudio(0) when you are ready | |
| 172 * for your audio callback function to be called. Since the audio driver | |
| 173 * may modify the requested size of the audio buffer, you should allocate | |
| 174 * any local mixing buffers after you open the audio device. | |
| 4217 | 175 * |
| 176 * @sa SDL_AudioSpec | |
| 0 | 177 */ |
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178 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained); |
| 0 | 179 |
| 180 typedef enum { | |
| 181 SDL_AUDIO_STOPPED = 0, | |
| 182 SDL_AUDIO_PLAYING, | |
| 183 SDL_AUDIO_PAUSED | |
| 184 } SDL_audiostatus; | |
| 4217 | 185 |
| 186 /** Get the current audio state */ | |
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187 extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void); |
| 0 | 188 |
| 4217 | 189 /** |
| 0 | 190 * This function pauses and unpauses the audio callback processing. |
| 191 * It should be called with a parameter of 0 after opening the audio | |
| 192 * device to start playing sound. This is so you can safely initialize | |
| 193 * data for your callback function after opening the audio device. | |
| 194 * Silence will be written to the audio device during the pause. | |
| 195 */ | |
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196 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); |
| 0 | 197 |
| 4217 | 198 /** |
| 0 | 199 * This function loads a WAVE from the data source, automatically freeing |
| 200 * that source if 'freesrc' is non-zero. For example, to load a WAVE file, | |
| 201 * you could do: | |
| 4217 | 202 * @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode |
| 0 | 203 * |
| 204 * If this function succeeds, it returns the given SDL_AudioSpec, | |
| 205 * filled with the audio data format of the wave data, and sets | |
| 206 * 'audio_buf' to a malloc()'d buffer containing the audio data, | |
| 207 * and sets 'audio_len' to the length of that audio buffer, in bytes. | |
| 208 * You need to free the audio buffer with SDL_FreeWAV() when you are | |
| 209 * done with it. | |
| 210 * | |
| 211 * This function returns NULL and sets the SDL error message if the | |
| 212 * wave file cannot be opened, uses an unknown data format, or is | |
| 213 * corrupt. Currently raw and MS-ADPCM WAVE files are supported. | |
| 214 */ | |
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215 extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len); |
| 0 | 216 |
| 4217 | 217 /** Compatibility convenience function -- loads a WAV from a file */ |
| 0 | 218 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ |
| 219 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) | |
| 220 | |
| 4217 | 221 /** |
| 0 | 222 * This function frees data previously allocated with SDL_LoadWAV_RW() |
| 223 */ | |
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224 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf); |
| 0 | 225 |
| 4217 | 226 /** |
| 0 | 227 * This function takes a source format and rate and a destination format |
| 228 * and rate, and initializes the 'cvt' structure with information needed | |
| 229 * by SDL_ConvertAudio() to convert a buffer of audio data from one format | |
| 230 * to the other. | |
| 4217 | 231 * |
| 232 * @return This function returns 0, or -1 if there was an error. | |
| 0 | 233 */ |
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234 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt, |
| 0 | 235 Uint16 src_format, Uint8 src_channels, int src_rate, |
| 236 Uint16 dst_format, Uint8 dst_channels, int dst_rate); | |
| 237 | |
| 4217 | 238 /** |
| 239 * Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(), | |
| 0 | 240 * created an audio buffer cvt->buf, and filled it with cvt->len bytes of |
| 241 * audio data in the source format, this function will convert it in-place | |
| 242 * to the desired format. | |
| 243 * The data conversion may expand the size of the audio data, so the buffer | |
| 244 * cvt->buf should be allocated after the cvt structure is initialized by | |
| 245 * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long. | |
| 246 */ | |
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247 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt); |
| 0 | 248 |
| 4217 | 249 |
| 250 #define SDL_MIX_MAXVOLUME 128 | |
| 251 /** | |
| 0 | 252 * This takes two audio buffers of the playing audio format and mixes |
| 253 * them, performing addition, volume adjustment, and overflow clipping. | |
| 254 * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME | |
| 255 * for full audio volume. Note this does not change hardware volume. | |
| 256 * This is provided for convenience -- you can mix your own audio data. | |
| 257 */ | |
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258 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume); |
| 0 | 259 |
| 4217 | 260 /** |
| 261 * @name Audio Locks | |
| 0 | 262 * The lock manipulated by these functions protects the callback function. |
| 263 * During a LockAudio/UnlockAudio pair, you can be guaranteed that the | |
| 264 * callback function is not running. Do not call these from the callback | |
| 265 * function or you will cause deadlock. | |
| 266 */ | |
| 4217 | 267 /*@{*/ |
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268 extern DECLSPEC void SDLCALL SDL_LockAudio(void); |
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269 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); |
| 4217 | 270 /*@}*/ |
| 0 | 271 |
| 4217 | 272 /** |
| 0 | 273 * This function shuts down audio processing and closes the audio device. |
| 274 */ | |
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275 extern DECLSPEC void SDLCALL SDL_CloseAudio(void); |
| 0 | 276 |
| 277 | |
| 278 /* Ends C function definitions when using C++ */ | |
| 279 #ifdef __cplusplus | |
| 280 } | |
| 281 #endif | |
| 282 #include "close_code.h" | |
| 283 | |
| 284 #endif /* _SDL_audio_h */ |