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1.1 root 1: /*
2: SDL - Simple DirectMedia Layer
3: Copyright (C) 1997-2004 Sam Lantinga
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
20: [email protected]
21: */
22:
23: #ifdef SAVE_RCSID
24: static char rcsid =
25: "@(#) $Id: SDL_audio.h,v 1.1.1.1 2005/11/18 23:23:29 rswindell Exp $";
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 SDL_AudioSpec {
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) */
53: Uint16 samples; /* Audio buffer size in samples (power of 2) */
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 (SDLCALL *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 (SDLCALL *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: */
108: extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
109: extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
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: */
115: extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen);
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: */
158: extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);
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;
168: extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);
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: */
177: extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
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: */
196: extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
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: */
205: extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf);
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: */
214: extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
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: */
226: extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt);
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
236: extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume);
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: */
244: extern DECLSPEC void SDLCALL SDL_LockAudio(void);
245: extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
246:
247: /*
248: * This function shuts down audio processing and closes the audio device.
249: */
250: extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
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 */
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