standalone: refactor for becoming also a SDL driver
[sdl_omap.git] / include / SDL_audio.h
CommitLineData
e14743d1 1/*
2 SDL - Simple DirectMedia Layer
3 Copyright (C) 1997-2009 Sam Lantinga
4
5 This library is free software; you can redistribute it and/or
6 modify it under the terms of the GNU Lesser General Public
7 License as published by the Free Software Foundation; either
8 version 2.1 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 Lesser General Public License for more details.
14
15 You should have received a copy of the GNU Lesser General Public
16 License along with this library; if not, write to the Free Software
17 Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
18
19 Sam Lantinga
20 slouken@libsdl.org
21*/
22
23/**
24 * @file SDL_audio.h
25 * Access to the raw audio mixing buffer for the SDL library
26 */
27
28#ifndef _SDL_audio_h
29#define _SDL_audio_h
30
31#include "SDL_stdinc.h"
32#include "SDL_error.h"
33#include "SDL_endian.h"
34#include "SDL_mutex.h"
35#include "SDL_thread.h"
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
41extern "C" {
42#endif
43
44/**
45 * When filling in the desired audio spec structure,
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.
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
58 * - 'desired->size' is the size in bytes of the audio buffer, and is
59 * calculated by SDL_OpenAudio().
60 * - 'desired->silence' is the value used to set the buffer to silence,
61 * and is calculated by SDL_OpenAudio().
62 * - 'desired->callback' should be set to a function that will be called
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.
68 * - 'desired->userdata' is passed as the first parameter to your callback
69 * function.
70 *
71 * @note The calculated values in this structure are calculated by SDL_OpenAudio()
72 *
73 */
74typedef 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 */
126typedef 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/*@{*/
150extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
151extern 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 */
159extern 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 *
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.
175 *
176 * @sa SDL_AudioSpec
177 */
178extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);
179
180typedef enum {
181 SDL_AUDIO_STOPPED = 0,
182 SDL_AUDIO_PLAYING,
183 SDL_AUDIO_PAUSED
184} SDL_audiostatus;
185
186/** Get the current audio state */
187extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);
188
189/**
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 */
196extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
197
198/**
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:
202 * @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode
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 */
215extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
216
217/** Compatibility convenience function -- loads a WAV from a file */
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
221/**
222 * This function frees data previously allocated with SDL_LoadWAV_RW()
223 */
224extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf);
225
226/**
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.
231 *
232 * @return This function returns 0, or -1 if there was an error.
233 */
234extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
235 Uint16 src_format, Uint8 src_channels, int src_rate,
236 Uint16 dst_format, Uint8 dst_channels, int dst_rate);
237
238/**
239 * Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),
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 */
247extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt);
248
249
250#define SDL_MIX_MAXVOLUME 128
251/**
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 */
258extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume);
259
260/**
261 * @name Audio Locks
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 */
267/*@{*/
268extern DECLSPEC void SDLCALL SDL_LockAudio(void);
269extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
270/*@}*/
271
272/**
273 * This function shuts down audio processing and closes the audio device.
274 */
275extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
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 */