Mercurial > sdl-ios-xcode
annotate include/SDL_audio.h @ 757:4f46fee887fe
Date: Sun, 14 Dec 2003 15:16:12 +0100
From: Stephane Marchesin
Subject: Re: [SDL] compile problem with sdl <-> alsa
The attathced patch does the trick (although there will have to be code
changes when the alsa 0.9 api get deprecated).
On a side-note, the good news is that using alsa 1.0.0rc1 solves the
previous issues I had with my ac97 via sound chipset.
author | Sam Lantinga <slouken@libsdl.org> |
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date | Tue, 16 Dec 2003 13:16:50 +0000 |
parents | 9154ec9ca3d2 |
children | b8d311d90021 |
rev | line source |
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0 | 1 /* |
2 SDL - Simple DirectMedia Layer | |
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3 Copyright (C) 1997, 1998, 1999, 2000, 2001, 2002 Sam Lantinga |
0 | 4 |
5 This library is free software; you can redistribute it and/or | |
6 modify it under the terms of the GNU Library General Public | |
7 License as published by the Free Software Foundation; either | |
8 version 2 of the License, or (at your option) any later version. | |
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 | |
13 Library General Public License for more details. | |
14 | |
15 You should have received a copy of the GNU Library General Public | |
16 License along with this library; if not, write to the Free | |
17 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
18 | |
19 Sam Lantinga | |
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20 slouken@libsdl.org |
0 | 21 */ |
22 | |
23 #ifdef SAVE_RCSID | |
24 static char rcsid = | |
25 "@(#) $Id$"; | |
26 #endif | |
27 | |
28 /* Access to the raw audio mixing buffer for the SDL library */ | |
29 | |
30 #ifndef _SDL_audio_h | |
31 #define _SDL_audio_h | |
32 | |
33 #include <stdio.h> | |
34 | |
35 #include "SDL_main.h" | |
36 #include "SDL_types.h" | |
37 #include "SDL_error.h" | |
38 #include "SDL_rwops.h" | |
39 #include "SDL_byteorder.h" | |
40 | |
41 #include "begin_code.h" | |
42 /* Set up for C function definitions, even when using C++ */ | |
43 #ifdef __cplusplus | |
44 extern "C" { | |
45 #endif | |
46 | |
47 /* The calculated values in this structure are calculated by SDL_OpenAudio() */ | |
48 typedef struct { | |
49 int freq; /* DSP frequency -- samples per second */ | |
50 Uint16 format; /* Audio data format */ | |
51 Uint8 channels; /* Number of channels: 1 mono, 2 stereo */ | |
52 Uint8 silence; /* Audio buffer silence value (calculated) */ | |
188 | 53 Uint16 samples; /* Audio buffer size in samples (power of 2) */ |
0 | 54 Uint16 padding; /* Necessary for some compile environments */ |
55 Uint32 size; /* Audio buffer size in bytes (calculated) */ | |
56 /* This function is called when the audio device needs more data. | |
57 'stream' is a pointer to the audio data buffer | |
58 'len' is the length of that buffer in bytes. | |
59 Once the callback returns, the buffer will no longer be valid. | |
60 Stereo samples are stored in a LRLRLR ordering. | |
61 */ | |
62 void (*callback)(void *userdata, Uint8 *stream, int len); | |
63 void *userdata; | |
64 } SDL_AudioSpec; | |
65 | |
66 /* Audio format flags (defaults to LSB byte order) */ | |
67 #define AUDIO_U8 0x0008 /* Unsigned 8-bit samples */ | |
68 #define AUDIO_S8 0x8008 /* Signed 8-bit samples */ | |
69 #define AUDIO_U16LSB 0x0010 /* Unsigned 16-bit samples */ | |
70 #define AUDIO_S16LSB 0x8010 /* Signed 16-bit samples */ | |
71 #define AUDIO_U16MSB 0x1010 /* As above, but big-endian byte order */ | |
72 #define AUDIO_S16MSB 0x9010 /* As above, but big-endian byte order */ | |
73 #define AUDIO_U16 AUDIO_U16LSB | |
74 #define AUDIO_S16 AUDIO_S16LSB | |
75 | |
76 /* Native audio byte ordering */ | |
77 #if SDL_BYTEORDER == SDL_LIL_ENDIAN | |
78 #define AUDIO_U16SYS AUDIO_U16LSB | |
79 #define AUDIO_S16SYS AUDIO_S16LSB | |
80 #else | |
81 #define AUDIO_U16SYS AUDIO_U16MSB | |
82 #define AUDIO_S16SYS AUDIO_S16MSB | |
83 #endif | |
84 | |
85 | |
86 /* A structure to hold a set of audio conversion filters and buffers */ | |
87 typedef struct SDL_AudioCVT { | |
88 int needed; /* Set to 1 if conversion possible */ | |
89 Uint16 src_format; /* Source audio format */ | |
90 Uint16 dst_format; /* Target audio format */ | |
91 double rate_incr; /* Rate conversion increment */ | |
92 Uint8 *buf; /* Buffer to hold entire audio data */ | |
93 int len; /* Length of original audio buffer */ | |
94 int len_cvt; /* Length of converted audio buffer */ | |
95 int len_mult; /* buffer must be len*len_mult big */ | |
96 double len_ratio; /* Given len, final size is len*len_ratio */ | |
97 void (*filters[10])(struct SDL_AudioCVT *cvt, Uint16 format); | |
98 int filter_index; /* Current audio conversion function */ | |
99 } SDL_AudioCVT; | |
100 | |
101 | |
102 /* Function prototypes */ | |
103 | |
104 /* These functions are used internally, and should not be used unless you | |
105 * have a specific need to specify the audio driver you want to use. | |
106 * You should normally use SDL_Init() or SDL_InitSubSystem(). | |
107 */ | |
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108 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); |
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109 extern DECLSPEC void SDLCALL SDL_AudioQuit(void); |
0 | 110 |
111 /* This function fills the given character buffer with the name of the | |
112 * current audio driver, and returns a pointer to it if the audio driver has | |
113 * been initialized. It returns NULL if no driver has been initialized. | |
114 */ | |
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115 extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen); |
0 | 116 |
117 /* | |
118 * This function opens the audio device with the desired parameters, and | |
119 * returns 0 if successful, placing the actual hardware parameters in the | |
120 * structure pointed to by 'obtained'. If 'obtained' is NULL, the audio | |
121 * data passed to the callback function will be guaranteed to be in the | |
122 * requested format, and will be automatically converted to the hardware | |
123 * audio format if necessary. This function returns -1 if it failed | |
124 * to open the audio device, or couldn't set up the audio thread. | |
125 * | |
126 * When filling in the desired audio spec structure, | |
127 * 'desired->freq' should be the desired audio frequency in samples-per-second. | |
128 * 'desired->format' should be the desired audio format. | |
129 * 'desired->samples' is the desired size of the audio buffer, in samples. | |
130 * This number should be a power of two, and may be adjusted by the audio | |
131 * driver to a value more suitable for the hardware. Good values seem to | |
132 * range between 512 and 8096 inclusive, depending on the application and | |
133 * CPU speed. Smaller values yield faster response time, but can lead | |
134 * to underflow if the application is doing heavy processing and cannot | |
135 * fill the audio buffer in time. A stereo sample consists of both right | |
136 * and left channels in LR ordering. | |
137 * Note that the number of samples is directly related to time by the | |
138 * following formula: ms = (samples*1000)/freq | |
139 * 'desired->size' is the size in bytes of the audio buffer, and is | |
140 * calculated by SDL_OpenAudio(). | |
141 * 'desired->silence' is the value used to set the buffer to silence, | |
142 * and is calculated by SDL_OpenAudio(). | |
143 * 'desired->callback' should be set to a function that will be called | |
144 * when the audio device is ready for more data. It is passed a pointer | |
145 * to the audio buffer, and the length in bytes of the audio buffer. | |
146 * This function usually runs in a separate thread, and so you should | |
147 * protect data structures that it accesses by calling SDL_LockAudio() | |
148 * and SDL_UnlockAudio() in your code. | |
149 * 'desired->userdata' is passed as the first parameter to your callback | |
150 * function. | |
151 * | |
152 * The audio device starts out playing silence when it's opened, and should | |
153 * be enabled for playing by calling SDL_PauseAudio(0) when you are ready | |
154 * for your audio callback function to be called. Since the audio driver | |
155 * may modify the requested size of the audio buffer, you should allocate | |
156 * any local mixing buffers after you open the audio device. | |
157 */ | |
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158 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained); |
0 | 159 |
160 /* | |
161 * Get the current audio state: | |
162 */ | |
163 typedef enum { | |
164 SDL_AUDIO_STOPPED = 0, | |
165 SDL_AUDIO_PLAYING, | |
166 SDL_AUDIO_PAUSED | |
167 } SDL_audiostatus; | |
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168 extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void); |
0 | 169 |
170 /* | |
171 * This function pauses and unpauses the audio callback processing. | |
172 * It should be called with a parameter of 0 after opening the audio | |
173 * device to start playing sound. This is so you can safely initialize | |
174 * data for your callback function after opening the audio device. | |
175 * Silence will be written to the audio device during the pause. | |
176 */ | |
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177 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); |
0 | 178 |
179 /* | |
180 * This function loads a WAVE from the data source, automatically freeing | |
181 * that source if 'freesrc' is non-zero. For example, to load a WAVE file, | |
182 * you could do: | |
183 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); | |
184 * | |
185 * If this function succeeds, it returns the given SDL_AudioSpec, | |
186 * filled with the audio data format of the wave data, and sets | |
187 * 'audio_buf' to a malloc()'d buffer containing the audio data, | |
188 * and sets 'audio_len' to the length of that audio buffer, in bytes. | |
189 * You need to free the audio buffer with SDL_FreeWAV() when you are | |
190 * done with it. | |
191 * | |
192 * This function returns NULL and sets the SDL error message if the | |
193 * wave file cannot be opened, uses an unknown data format, or is | |
194 * corrupt. Currently raw and MS-ADPCM WAVE files are supported. | |
195 */ | |
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196 extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len); |
0 | 197 |
198 /* Compatibility convenience function -- loads a WAV from a file */ | |
199 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ | |
200 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) | |
201 | |
202 /* | |
203 * This function frees data previously allocated with SDL_LoadWAV_RW() | |
204 */ | |
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205 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf); |
0 | 206 |
207 /* | |
208 * This function takes a source format and rate and a destination format | |
209 * and rate, and initializes the 'cvt' structure with information needed | |
210 * by SDL_ConvertAudio() to convert a buffer of audio data from one format | |
211 * to the other. | |
212 * This function returns 0, or -1 if there was an error. | |
213 */ | |
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214 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt, |
0 | 215 Uint16 src_format, Uint8 src_channels, int src_rate, |
216 Uint16 dst_format, Uint8 dst_channels, int dst_rate); | |
217 | |
218 /* Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(), | |
219 * created an audio buffer cvt->buf, and filled it with cvt->len bytes of | |
220 * audio data in the source format, this function will convert it in-place | |
221 * to the desired format. | |
222 * The data conversion may expand the size of the audio data, so the buffer | |
223 * cvt->buf should be allocated after the cvt structure is initialized by | |
224 * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long. | |
225 */ | |
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226 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt); |
0 | 227 |
228 /* | |
229 * This takes two audio buffers of the playing audio format and mixes | |
230 * them, performing addition, volume adjustment, and overflow clipping. | |
231 * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME | |
232 * for full audio volume. Note this does not change hardware volume. | |
233 * This is provided for convenience -- you can mix your own audio data. | |
234 */ | |
235 #define SDL_MIX_MAXVOLUME 128 | |
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236 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume); |
0 | 237 |
238 /* | |
239 * The lock manipulated by these functions protects the callback function. | |
240 * During a LockAudio/UnlockAudio pair, you can be guaranteed that the | |
241 * callback function is not running. Do not call these from the callback | |
242 * function or you will cause deadlock. | |
243 */ | |
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244 extern DECLSPEC void SDLCALL SDL_LockAudio(void); |
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245 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); |
0 | 246 |
247 /* | |
248 * This function shuts down audio processing and closes the audio device. | |
249 */ | |
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250 extern DECLSPEC void SDLCALL SDL_CloseAudio(void); |
0 | 251 |
252 | |
253 /* Ends C function definitions when using C++ */ | |
254 #ifdef __cplusplus | |
255 } | |
256 #endif | |
257 #include "close_code.h" | |
258 | |
259 #endif /* _SDL_audio_h */ |