Mercurial > sdl-ios-xcode
annotate include/SDL_audio.h @ 3899:081aecdb0911 SDL-1.2
From: Gabriel Gambetta
To: SDL Mailing List <sdl@libsdl.org>
Date: Fri, 17 Nov 2006 00:39:29 -0200
Subject: [SDL] Alpha blending bug - fixed?
It turns out mm5 never got the correct value. This failed :
[...]
"movd %1, %%mm5\n\t"
: : "m" (amask), "m" (sf->Ashift) );
mm5 got 0xFF000018 instead of 0x00000018. However I did this :
Uint32 ashift = sf->Ashift;
[...]
"movd %1, %%mm5\n\t"
: : "m" (amask), "m" (ashift) );
and everything worked fine.
The only thing I could find out was that ashift is 32-bit aligned but
sf->Ashift isn't:
printf("ashift %8X [%d]\n", &ashift, (int)(&ashift) % 4);
printf("sf->Ashift %8X [%d]\n", &sf->Ashift, (int)(&sf->Ashift) % 4);
ashift at BFF0E0C4 [0]
sf->Ashift at 08AFB9CD [1]
Could this be the problem? If it is, the fix is extremely trivial.
Should I submit a patch?
--Gabriel
| author | Ryan C. Gordon <icculus@icculus.org> |
|---|---|
| date | Tue, 21 Nov 2006 23:17:00 +0000 |
| parents | c71e05b4dc2e |
| children | 8b9d79e7eacf c121d94672cb a1b03ba2fcd0 |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
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3 Copyright (C) 1997-2006 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 | |
| 23 /* Access to the raw audio mixing buffer for the SDL library */ | |
| 24 | |
| 25 #ifndef _SDL_audio_h | |
| 26 #define _SDL_audio_h | |
| 27 | |
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28 #include "SDL_stdinc.h" |
| 0 | 29 #include "SDL_error.h" |
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30 #include "SDL_endian.h" |
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31 #include "SDL_mutex.h" |
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32 #include "SDL_thread.h" |
| 0 | 33 #include "SDL_rwops.h" |
| 34 | |
| 35 #include "begin_code.h" | |
| 36 /* Set up for C function definitions, even when using C++ */ | |
| 37 #ifdef __cplusplus | |
| 38 extern "C" { | |
| 39 #endif | |
| 40 | |
| 41 /* The calculated values in this structure are calculated by SDL_OpenAudio() */ | |
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42 typedef struct SDL_AudioSpec { |
| 0 | 43 int freq; /* DSP frequency -- samples per second */ |
| 44 Uint16 format; /* Audio data format */ | |
| 45 Uint8 channels; /* Number of channels: 1 mono, 2 stereo */ | |
| 46 Uint8 silence; /* Audio buffer silence value (calculated) */ | |
| 188 | 47 Uint16 samples; /* Audio buffer size in samples (power of 2) */ |
| 0 | 48 Uint16 padding; /* Necessary for some compile environments */ |
| 49 Uint32 size; /* Audio buffer size in bytes (calculated) */ | |
| 50 /* This function is called when the audio device needs more data. | |
| 51 'stream' is a pointer to the audio data buffer | |
| 52 'len' is the length of that buffer in bytes. | |
| 53 Once the callback returns, the buffer will no longer be valid. | |
| 54 Stereo samples are stored in a LRLRLR ordering. | |
| 55 */ | |
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56 void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len); |
| 0 | 57 void *userdata; |
| 58 } SDL_AudioSpec; | |
| 59 | |
| 60 /* Audio format flags (defaults to LSB byte order) */ | |
| 61 #define AUDIO_U8 0x0008 /* Unsigned 8-bit samples */ | |
| 62 #define AUDIO_S8 0x8008 /* Signed 8-bit samples */ | |
| 63 #define AUDIO_U16LSB 0x0010 /* Unsigned 16-bit samples */ | |
| 64 #define AUDIO_S16LSB 0x8010 /* Signed 16-bit samples */ | |
| 65 #define AUDIO_U16MSB 0x1010 /* As above, but big-endian byte order */ | |
| 66 #define AUDIO_S16MSB 0x9010 /* As above, but big-endian byte order */ | |
| 67 #define AUDIO_U16 AUDIO_U16LSB | |
| 68 #define AUDIO_S16 AUDIO_S16LSB | |
| 69 | |
| 70 /* Native audio byte ordering */ | |
| 71 #if SDL_BYTEORDER == SDL_LIL_ENDIAN | |
| 72 #define AUDIO_U16SYS AUDIO_U16LSB | |
| 73 #define AUDIO_S16SYS AUDIO_S16LSB | |
| 74 #else | |
| 75 #define AUDIO_U16SYS AUDIO_U16MSB | |
| 76 #define AUDIO_S16SYS AUDIO_S16MSB | |
| 77 #endif | |
| 78 | |
| 79 | |
| 80 /* A structure to hold a set of audio conversion filters and buffers */ | |
| 81 typedef struct SDL_AudioCVT { | |
| 82 int needed; /* Set to 1 if conversion possible */ | |
| 83 Uint16 src_format; /* Source audio format */ | |
| 84 Uint16 dst_format; /* Target audio format */ | |
| 85 double rate_incr; /* Rate conversion increment */ | |
| 86 Uint8 *buf; /* Buffer to hold entire audio data */ | |
| 87 int len; /* Length of original audio buffer */ | |
| 88 int len_cvt; /* Length of converted audio buffer */ | |
| 89 int len_mult; /* buffer must be len*len_mult big */ | |
| 90 double len_ratio; /* Given len, final size is len*len_ratio */ | |
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91 void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format); |
| 0 | 92 int filter_index; /* Current audio conversion function */ |
| 93 } SDL_AudioCVT; | |
| 94 | |
| 95 | |
| 96 /* Function prototypes */ | |
| 97 | |
| 98 /* These functions are used internally, and should not be used unless you | |
| 99 * have a specific need to specify the audio driver you want to use. | |
| 100 * You should normally use SDL_Init() or SDL_InitSubSystem(). | |
| 101 */ | |
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102 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); |
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103 extern DECLSPEC void SDLCALL SDL_AudioQuit(void); |
| 0 | 104 |
| 105 /* This function fills the given character buffer with the name of the | |
| 106 * current audio driver, and returns a pointer to it if the audio driver has | |
| 107 * been initialized. It returns NULL if no driver has been initialized. | |
| 108 */ | |
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109 extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen); |
| 0 | 110 |
| 111 /* | |
| 112 * This function opens the audio device with the desired parameters, and | |
| 113 * returns 0 if successful, placing the actual hardware parameters in the | |
| 114 * structure pointed to by 'obtained'. If 'obtained' is NULL, the audio | |
| 115 * data passed to the callback function will be guaranteed to be in the | |
| 116 * requested format, and will be automatically converted to the hardware | |
| 117 * audio format if necessary. This function returns -1 if it failed | |
| 118 * to open the audio device, or couldn't set up the audio thread. | |
| 119 * | |
| 120 * When filling in the desired audio spec structure, | |
| 121 * 'desired->freq' should be the desired audio frequency in samples-per-second. | |
| 122 * 'desired->format' should be the desired audio format. | |
| 123 * 'desired->samples' is the desired size of the audio buffer, in samples. | |
| 124 * This number should be a power of two, and may be adjusted by the audio | |
| 125 * driver to a value more suitable for the hardware. Good values seem to | |
| 126 * range between 512 and 8096 inclusive, depending on the application and | |
| 127 * CPU speed. Smaller values yield faster response time, but can lead | |
| 128 * to underflow if the application is doing heavy processing and cannot | |
| 129 * fill the audio buffer in time. A stereo sample consists of both right | |
| 130 * and left channels in LR ordering. | |
| 131 * Note that the number of samples is directly related to time by the | |
| 132 * following formula: ms = (samples*1000)/freq | |
| 133 * 'desired->size' is the size in bytes of the audio buffer, and is | |
| 134 * calculated by SDL_OpenAudio(). | |
| 135 * 'desired->silence' is the value used to set the buffer to silence, | |
| 136 * and is calculated by SDL_OpenAudio(). | |
| 137 * 'desired->callback' should be set to a function that will be called | |
| 138 * when the audio device is ready for more data. It is passed a pointer | |
| 139 * to the audio buffer, and the length in bytes of the audio buffer. | |
| 140 * This function usually runs in a separate thread, and so you should | |
| 141 * protect data structures that it accesses by calling SDL_LockAudio() | |
| 142 * and SDL_UnlockAudio() in your code. | |
| 143 * 'desired->userdata' is passed as the first parameter to your callback | |
| 144 * function. | |
| 145 * | |
| 146 * The audio device starts out playing silence when it's opened, and should | |
| 147 * be enabled for playing by calling SDL_PauseAudio(0) when you are ready | |
| 148 * for your audio callback function to be called. Since the audio driver | |
| 149 * may modify the requested size of the audio buffer, you should allocate | |
| 150 * any local mixing buffers after you open the audio device. | |
| 151 */ | |
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152 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained); |
| 0 | 153 |
| 154 /* | |
| 155 * Get the current audio state: | |
| 156 */ | |
| 157 typedef enum { | |
| 158 SDL_AUDIO_STOPPED = 0, | |
| 159 SDL_AUDIO_PLAYING, | |
| 160 SDL_AUDIO_PAUSED | |
| 161 } SDL_audiostatus; | |
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162 extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void); |
| 0 | 163 |
| 164 /* | |
| 165 * This function pauses and unpauses the audio callback processing. | |
| 166 * It should be called with a parameter of 0 after opening the audio | |
| 167 * device to start playing sound. This is so you can safely initialize | |
| 168 * data for your callback function after opening the audio device. | |
| 169 * Silence will be written to the audio device during the pause. | |
| 170 */ | |
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171 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); |
| 0 | 172 |
| 173 /* | |
| 174 * This function loads a WAVE from the data source, automatically freeing | |
| 175 * that source if 'freesrc' is non-zero. For example, to load a WAVE file, | |
| 176 * you could do: | |
| 177 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); | |
| 178 * | |
| 179 * If this function succeeds, it returns the given SDL_AudioSpec, | |
| 180 * filled with the audio data format of the wave data, and sets | |
| 181 * 'audio_buf' to a malloc()'d buffer containing the audio data, | |
| 182 * and sets 'audio_len' to the length of that audio buffer, in bytes. | |
| 183 * You need to free the audio buffer with SDL_FreeWAV() when you are | |
| 184 * done with it. | |
| 185 * | |
| 186 * This function returns NULL and sets the SDL error message if the | |
| 187 * wave file cannot be opened, uses an unknown data format, or is | |
| 188 * corrupt. Currently raw and MS-ADPCM WAVE files are supported. | |
| 189 */ | |
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190 extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len); |
| 0 | 191 |
| 192 /* Compatibility convenience function -- loads a WAV from a file */ | |
| 193 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ | |
| 194 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) | |
| 195 | |
| 196 /* | |
| 197 * This function frees data previously allocated with SDL_LoadWAV_RW() | |
| 198 */ | |
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199 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf); |
| 0 | 200 |
| 201 /* | |
| 202 * This function takes a source format and rate and a destination format | |
| 203 * and rate, and initializes the 'cvt' structure with information needed | |
| 204 * by SDL_ConvertAudio() to convert a buffer of audio data from one format | |
| 205 * to the other. | |
| 206 * This function returns 0, or -1 if there was an error. | |
| 207 */ | |
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208 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt, |
| 0 | 209 Uint16 src_format, Uint8 src_channels, int src_rate, |
| 210 Uint16 dst_format, Uint8 dst_channels, int dst_rate); | |
| 211 | |
| 212 /* Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(), | |
| 213 * created an audio buffer cvt->buf, and filled it with cvt->len bytes of | |
| 214 * audio data in the source format, this function will convert it in-place | |
| 215 * to the desired format. | |
| 216 * The data conversion may expand the size of the audio data, so the buffer | |
| 217 * cvt->buf should be allocated after the cvt structure is initialized by | |
| 218 * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long. | |
| 219 */ | |
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220 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt); |
| 0 | 221 |
| 222 /* | |
| 223 * This takes two audio buffers of the playing audio format and mixes | |
| 224 * them, performing addition, volume adjustment, and overflow clipping. | |
| 225 * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME | |
| 226 * for full audio volume. Note this does not change hardware volume. | |
| 227 * This is provided for convenience -- you can mix your own audio data. | |
| 228 */ | |
| 229 #define SDL_MIX_MAXVOLUME 128 | |
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230 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume); |
| 0 | 231 |
| 232 /* | |
| 233 * The lock manipulated by these functions protects the callback function. | |
| 234 * During a LockAudio/UnlockAudio pair, you can be guaranteed that the | |
| 235 * callback function is not running. Do not call these from the callback | |
| 236 * function or you will cause deadlock. | |
| 237 */ | |
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238 extern DECLSPEC void SDLCALL SDL_LockAudio(void); |
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239 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); |
| 0 | 240 |
| 241 /* | |
| 242 * This function shuts down audio processing and closes the audio device. | |
| 243 */ | |
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244 extern DECLSPEC void SDLCALL SDL_CloseAudio(void); |
| 0 | 245 |
| 246 | |
| 247 /* Ends C function definitions when using C++ */ | |
| 248 #ifdef __cplusplus | |
| 249 } | |
| 250 #endif | |
| 251 #include "close_code.h" | |
| 252 | |
| 253 #endif /* _SDL_audio_h */ |