Mercurial > sdl-ios-xcode
annotate src/audio/SDL_audio.c @ 1035:974ba6ae0fa3
Date: Wed, 26 Jan 2005 13:37:09 GMT
From: Peter Naulls
Subject: RISC OS SDL Patches
Sam, I've attached a diff of the latest changes to libSDL for RISC OS
support. These changes are by Alan Buckley and myself.
The most significant of these are:
Optimised assembler blit rountines - I've attached the file
src/video/riscos/SDL_riscosASM.s which is needed for this.
Move to using /dev/dsp instead of its own audio implementation.
This means that src/audio/riscos/SDL_drenderer.c should be removed
Typo fixes. Mainly correct spelling of "RISC OS", but some from elsewhere
too.
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Sat, 12 Feb 2005 18:01:31 +0000 |
| parents | d74fbf56f2f6 |
| children | 173c063d4f55 |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
|
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Updated copyright information for 2004 (Happy New Year!)
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3 Copyright (C) 1997-2004 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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Updated the source with the correct e-mail address
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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 /* Allow access to a raw mixing buffer */ | |
| 29 #include <stdlib.h> | |
| 30 #include <stdio.h> | |
| 31 #include <string.h> | |
| 32 | |
| 33 #include "SDL.h" | |
| 34 #include "SDL_audio.h" | |
| 35 #include "SDL_timer.h" | |
| 36 #include "SDL_error.h" | |
| 37 #include "SDL_audio_c.h" | |
| 38 #include "SDL_audiomem.h" | |
| 39 #include "SDL_sysaudio.h" | |
| 40 | |
| 41 /* Available audio drivers */ | |
| 42 static AudioBootStrap *bootstrap[] = { | |
|
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43 #ifdef OPENBSD_AUDIO_SUPPORT |
|
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44 &OPENBSD_AUDIO_bootstrap, |
|
94
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Cleaned up the OpenBSD port, thanks to Peter Valchev
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45 #endif |
| 0 | 46 #ifdef OSS_SUPPORT |
| 47 &DSP_bootstrap, | |
| 48 &DMA_bootstrap, | |
| 49 #endif | |
| 50 #ifdef ALSA_SUPPORT | |
| 51 &ALSA_bootstrap, | |
| 52 #endif | |
|
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Date: Sat, 2 Aug 2003 16:22:51 +0300
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53 #ifdef QNXNTOAUDIO_SUPPORT |
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54 &QNXNTOAUDIO_bootstrap, |
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55 #endif |
|
148
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Audio subsystem no longer assumes sun audio API on UNIX systems
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56 #ifdef SUNAUDIO_SUPPORT |
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57 &SUNAUDIO_bootstrap, |
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58 #endif |
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59 #ifdef DMEDIA_SUPPORT |
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60 &DMEDIA_bootstrap, |
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35
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Added detection of Open Sound System on Solaris x86
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61 #endif |
| 0 | 62 #ifdef ARTSC_SUPPORT |
| 63 &ARTSC_bootstrap, | |
| 64 #endif | |
| 65 #ifdef ESD_SUPPORT | |
| 66 &ESD_bootstrap, | |
| 67 #endif | |
| 68 #ifdef NAS_SUPPORT | |
| 69 &NAS_bootstrap, | |
| 70 #endif | |
| 71 #ifdef ENABLE_DIRECTX | |
| 72 &DSOUND_bootstrap, | |
| 73 #endif | |
| 74 #ifdef ENABLE_WINDIB | |
| 75 &WAVEOUT_bootstrap, | |
| 76 #endif | |
| 77 #ifdef __BEOS__ | |
| 78 &BAUDIO_bootstrap, | |
| 79 #endif | |
|
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CoreAudio driver works on Mac OSX 10.1
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80 #ifdef MACOSX |
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Audio improvements from Max Horn, including a new CoreAudio driver for MacOSX
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81 &COREAUDIO_bootstrap, |
|
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82 #endif |
| 0 | 83 #if defined(macintosh) || TARGET_API_MAC_CARBON |
| 84 &SNDMGR_bootstrap, | |
| 85 #endif | |
| 86 #ifdef _AIX | |
| 87 &Paud_bootstrap, | |
| 88 #endif | |
|
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89 #ifdef ENABLE_AHI |
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90 &AHI_bootstrap, |
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91 #endif |
|
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Added audio and CD-ROM support for OSF/Tru64 (thanks Hayashi!)
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92 #ifdef MMEAUDIO_SUPPORT |
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93 &MMEAUDIO_bootstrap, |
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94 #endif |
|
398
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Added Atari audio support (thanks Patrice!)
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95 #ifdef MINTAUDIO_SUPPORT |
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644
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Atari MiNT: added more audio drivers
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96 &MINTAUDIO_GSXB_bootstrap, |
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97 &MINTAUDIO_MCSN_bootstrap, |
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98 &MINTAUDIO_STFA_bootstrap, |
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99 &MINTAUDIO_XBIOS_bootstrap, |
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100 &MINTAUDIO_DMA8_bootstrap, |
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398
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101 #endif |
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Added an audio driver that writes to disk (thanks Ryan!)
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102 #ifdef DISKAUD_SUPPORT |
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103 &DISKAUD_bootstrap, |
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104 #endif |
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509
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Added initial support for Dreamcast (thanks HERO!)
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105 #ifdef ENABLE_DC |
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Added initial support for Dreamcast (thanks HERO!)
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106 &DCAUD_bootstrap, |
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107 #endif |
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Added initial support for RISC OS (thanks Peter Naulls!)
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108 #ifdef DRENDERER_SUPPORT |
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109 &DRENDERER_bootstrap, |
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110 #endif |
| 0 | 111 NULL |
| 112 }; | |
| 113 SDL_AudioDevice *current_audio = NULL; | |
| 114 | |
| 115 /* Various local functions */ | |
| 116 int SDL_AudioInit(const char *driver_name); | |
| 117 void SDL_AudioQuit(void); | |
| 118 | |
|
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119 #ifdef ENABLE_AHI |
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120 static int audio_configured = 0; |
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121 #endif |
| 0 | 122 |
| 123 /* The general mixing thread function */ | |
| 124 int SDL_RunAudio(void *audiop) | |
| 125 { | |
| 126 SDL_AudioDevice *audio = (SDL_AudioDevice *)audiop; | |
| 127 Uint8 *stream; | |
| 128 int stream_len; | |
| 129 void *udata; | |
| 130 void (*fill)(void *userdata,Uint8 *stream, int len); | |
| 131 int silence; | |
|
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132 #ifdef ENABLE_AHI |
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133 int started = 0; |
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134 |
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135 /* AmigaOS NEEDS that the audio driver is opened in the thread that uses it! */ |
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136 |
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137 D(bug("Task audio started audio struct:<%lx>...\n",audiop)); |
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138 |
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139 D(bug("Before Openaudio...")); |
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140 if(audio->OpenAudio(audio, &audio->spec)==-1) |
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141 { |
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142 D(bug("Open audio failed...\n")); |
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143 return(-1); |
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144 } |
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145 D(bug("OpenAudio...OK\n")); |
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146 #endif |
| 0 | 147 |
| 148 /* Perform any thread setup */ | |
| 149 if ( audio->ThreadInit ) { | |
| 150 audio->ThreadInit(audio); | |
| 151 } | |
| 152 audio->threadid = SDL_ThreadID(); | |
| 153 | |
| 154 /* Set up the mixing function */ | |
| 155 fill = audio->spec.callback; | |
| 156 udata = audio->spec.userdata; | |
|
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157 |
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158 #ifdef ENABLE_AHI |
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159 audio_configured = 1; |
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160 |
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161 D(bug("Audio configured... Checking for conversion\n")); |
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162 SDL_mutexP(audio->mixer_lock); |
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163 D(bug("Semaphore obtained...\n")); |
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164 #endif |
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165 |
| 0 | 166 if ( audio->convert.needed ) { |
| 167 if ( audio->convert.src_format == AUDIO_U8 ) { | |
| 168 silence = 0x80; | |
| 169 } else { | |
| 170 silence = 0; | |
| 171 } | |
| 172 stream_len = audio->convert.len; | |
| 173 } else { | |
| 174 silence = audio->spec.silence; | |
| 175 stream_len = audio->spec.size; | |
| 176 } | |
| 177 stream = audio->fake_stream; | |
| 178 | |
|
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179 #ifdef ENABLE_AHI |
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180 SDL_mutexV(audio->mixer_lock); |
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181 D(bug("Entering audio loop...\n")); |
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182 #endif |
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183 |
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184 |
| 0 | 185 /* Loop, filling the audio buffers */ |
| 186 while ( audio->enabled ) { | |
| 187 | |
| 188 /* Wait for new current buffer to finish playing */ | |
| 189 if ( stream == audio->fake_stream ) { | |
| 190 SDL_Delay((audio->spec.samples*1000)/audio->spec.freq); | |
| 191 } else { | |
|
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192 #ifdef ENABLE_AHI |
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193 if ( started > 1 ) |
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194 #endif |
| 0 | 195 audio->WaitAudio(audio); |
| 196 } | |
| 197 | |
| 198 /* Fill the current buffer with sound */ | |
| 199 if ( audio->convert.needed ) { | |
| 200 if ( audio->convert.buf ) { | |
| 201 stream = audio->convert.buf; | |
| 202 } else { | |
| 203 continue; | |
| 204 } | |
| 205 } else { | |
| 206 stream = audio->GetAudioBuf(audio); | |
| 207 if ( stream == NULL ) { | |
| 208 stream = audio->fake_stream; | |
| 209 } | |
| 210 } | |
| 211 memset(stream, silence, stream_len); | |
| 212 | |
| 213 if ( ! audio->paused ) { | |
| 214 SDL_mutexP(audio->mixer_lock); | |
| 215 (*fill)(udata, stream, stream_len); | |
| 216 SDL_mutexV(audio->mixer_lock); | |
| 217 } | |
| 218 | |
| 219 /* Convert the audio if necessary */ | |
| 220 if ( audio->convert.needed ) { | |
| 221 SDL_ConvertAudio(&audio->convert); | |
| 222 stream = audio->GetAudioBuf(audio); | |
| 223 if ( stream == NULL ) { | |
| 224 stream = audio->fake_stream; | |
| 225 } | |
| 226 memcpy(stream, audio->convert.buf, | |
| 227 audio->convert.len_cvt); | |
| 228 } | |
| 229 | |
| 230 /* Ready current buffer for play and change current buffer */ | |
| 231 if ( stream != audio->fake_stream ) { | |
| 232 audio->PlayAudio(audio); | |
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233 #ifdef ENABLE_AHI |
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234 /* AmigaOS don't have to wait the first time audio is played! */ |
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235 started++; |
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236 #endif |
| 0 | 237 } |
| 238 } | |
| 239 /* Wait for the audio to drain.. */ | |
| 240 if ( audio->WaitDone ) { | |
| 241 audio->WaitDone(audio); | |
| 242 } | |
|
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243 |
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244 #ifdef ENABLE_AHI |
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245 D(bug("WaitAudio...Done\n")); |
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246 |
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247 audio->CloseAudio(audio); |
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248 |
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249 D(bug("CloseAudio..Done, subtask exiting...\n")); |
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250 audio_configured = 0; |
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251 #endif |
| 0 | 252 return(0); |
| 253 } | |
| 254 | |
|
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255 static void SDL_LockAudio_Default(SDL_AudioDevice *audio) |
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256 { |
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257 if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) { |
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258 return; |
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259 } |
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260 SDL_mutexP(audio->mixer_lock); |
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261 } |
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262 |
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263 static void SDL_UnlockAudio_Default(SDL_AudioDevice *audio) |
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264 { |
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265 if ( audio->thread && (SDL_ThreadID() == audio->threadid) ) { |
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266 return; |
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267 } |
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268 SDL_mutexV(audio->mixer_lock); |
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269 } |
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270 |
| 0 | 271 int SDL_AudioInit(const char *driver_name) |
| 272 { | |
| 273 SDL_AudioDevice *audio; | |
| 274 int i = 0, idx; | |
| 275 | |
| 276 /* Check to make sure we don't overwrite 'current_audio' */ | |
| 277 if ( current_audio != NULL ) { | |
| 278 SDL_AudioQuit(); | |
| 279 } | |
| 280 | |
| 281 /* Select the proper audio driver */ | |
| 282 audio = NULL; | |
| 283 idx = 0; | |
| 284 #ifdef unix | |
| 285 if ( (driver_name == NULL) && (getenv("ESPEAKER") != NULL) ) { | |
| 286 /* Ahem, we know that if ESPEAKER is set, user probably wants | |
| 287 to use ESD, but don't start it if it's not already running. | |
| 288 This probably isn't the place to do this, but... Shh! :) | |
| 289 */ | |
| 290 for ( i=0; bootstrap[i]; ++i ) { | |
| 291 if ( strcmp(bootstrap[i]->name, "esd") == 0 ) { | |
| 292 const char *esd_no_spawn; | |
| 293 | |
| 294 /* Don't start ESD if it's not running */ | |
| 295 esd_no_spawn = getenv("ESD_NO_SPAWN"); | |
| 296 if ( esd_no_spawn == NULL ) { | |
| 297 putenv("ESD_NO_SPAWN=1"); | |
| 298 } | |
| 299 if ( bootstrap[i]->available() ) { | |
| 300 audio = bootstrap[i]->create(0); | |
| 301 break; | |
| 302 } | |
| 303 #ifdef linux /* No unsetenv() on most platforms */ | |
| 304 if ( esd_no_spawn == NULL ) { | |
| 305 unsetenv("ESD_NO_SPAWN"); | |
| 306 } | |
| 307 #endif | |
| 308 } | |
| 309 } | |
| 310 } | |
| 311 #endif /* unix */ | |
| 312 if ( audio == NULL ) { | |
| 313 if ( driver_name != NULL ) { | |
| 314 #if 0 /* This will be replaced with a better driver selection API */ | |
| 315 if ( strrchr(driver_name, ':') != NULL ) { | |
| 316 idx = atoi(strrchr(driver_name, ':')+1); | |
| 317 } | |
| 318 #endif | |
| 319 for ( i=0; bootstrap[i]; ++i ) { | |
| 320 if (strncmp(bootstrap[i]->name, driver_name, | |
| 321 strlen(bootstrap[i]->name)) == 0) { | |
| 322 if ( bootstrap[i]->available() ) { | |
| 323 audio=bootstrap[i]->create(idx); | |
| 324 break; | |
| 325 } | |
| 326 } | |
| 327 } | |
| 328 } else { | |
| 329 for ( i=0; bootstrap[i]; ++i ) { | |
| 330 if ( bootstrap[i]->available() ) { | |
| 331 audio = bootstrap[i]->create(idx); | |
| 332 if ( audio != NULL ) { | |
| 333 break; | |
| 334 } | |
| 335 } | |
| 336 } | |
| 337 } | |
| 338 if ( audio == NULL ) { | |
| 339 SDL_SetError("No available audio device"); | |
| 340 #if 0 /* Don't fail SDL_Init() if audio isn't available. | |
| 341 SDL_OpenAudio() will handle it at that point. *sigh* | |
| 342 */ | |
| 343 return(-1); | |
| 344 #endif | |
| 345 } | |
| 346 } | |
| 347 current_audio = audio; | |
| 348 if ( current_audio ) { | |
| 349 current_audio->name = bootstrap[i]->name; | |
|
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350 if ( !current_audio->LockAudio && !current_audio->UnlockAudio ) { |
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351 current_audio->LockAudio = SDL_LockAudio_Default; |
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352 current_audio->UnlockAudio = SDL_UnlockAudio_Default; |
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353 } |
| 0 | 354 } |
| 355 return(0); | |
| 356 } | |
| 357 | |
| 358 char *SDL_AudioDriverName(char *namebuf, int maxlen) | |
| 359 { | |
| 360 if ( current_audio != NULL ) { | |
| 361 strncpy(namebuf, current_audio->name, maxlen-1); | |
| 362 namebuf[maxlen-1] = '\0'; | |
| 363 return(namebuf); | |
| 364 } | |
| 365 return(NULL); | |
| 366 } | |
| 367 | |
| 368 int SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained) | |
| 369 { | |
| 370 SDL_AudioDevice *audio; | |
| 371 | |
| 372 /* Start up the audio driver, if necessary */ | |
| 373 if ( ! current_audio ) { | |
| 374 if ( (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) || | |
| 375 (current_audio == NULL) ) { | |
| 376 return(-1); | |
| 377 } | |
| 378 } | |
| 379 audio = current_audio; | |
| 380 | |
|
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381 if (audio->opened) { |
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382 SDL_SetError("Audio device is already opened"); |
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383 return(-1); |
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384 } |
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385 |
| 0 | 386 /* Verify some parameters */ |
| 387 if ( desired->callback == NULL ) { | |
| 388 SDL_SetError("SDL_OpenAudio() passed a NULL callback"); | |
| 389 return(-1); | |
| 390 } | |
| 391 switch ( desired->channels ) { | |
| 392 case 1: /* Mono */ | |
| 393 case 2: /* Stereo */ | |
|
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394 case 4: /* surround */ |
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395 case 6: /* surround with center and lfe */ |
| 0 | 396 break; |
| 397 default: | |
| 398 SDL_SetError("1 (mono) and 2 (stereo) channels supported"); | |
| 399 return(-1); | |
| 400 } | |
| 401 | |
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402 #if defined(macintosh) || (defined(__riscos__) && !defined(DISABLE_THREADS)) |
| 0 | 403 /* FIXME: Need to implement PPC interrupt asm for SDL_LockAudio() */ |
| 404 #else | |
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405 #if defined(__MINT__) && !defined(ENABLE_THREADS) |
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406 /* Uses interrupt driven audio, without thread */ |
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407 #else |
| 0 | 408 /* Create a semaphore for locking the sound buffers */ |
| 409 audio->mixer_lock = SDL_CreateMutex(); | |
| 410 if ( audio->mixer_lock == NULL ) { | |
| 411 SDL_SetError("Couldn't create mixer lock"); | |
| 412 SDL_CloseAudio(); | |
| 413 return(-1); | |
| 414 } | |
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415 #endif /* __MINT__ */ |
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416 #endif /* macintosh */ |
| 0 | 417 |
| 418 /* Calculate the silence and size of the audio specification */ | |
| 419 SDL_CalculateAudioSpec(desired); | |
| 420 | |
| 421 /* Open the audio subsystem */ | |
| 422 memcpy(&audio->spec, desired, sizeof(audio->spec)); | |
| 423 audio->convert.needed = 0; | |
| 424 audio->enabled = 1; | |
| 425 audio->paused = 1; | |
|
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426 |
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427 #ifndef ENABLE_AHI |
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428 |
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429 /* AmigaOS opens audio inside the main loop */ |
| 0 | 430 audio->opened = audio->OpenAudio(audio, &audio->spec)+1; |
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431 |
| 0 | 432 if ( ! audio->opened ) { |
| 433 SDL_CloseAudio(); | |
| 434 return(-1); | |
| 435 } | |
|
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436 #else |
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437 D(bug("Locking semaphore...")); |
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438 SDL_mutexP(audio->mixer_lock); |
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439 |
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440 audio->thread = SDL_CreateThread(SDL_RunAudio, audio); |
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441 D(bug("Created thread...\n")); |
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442 |
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443 if ( audio->thread == NULL ) { |
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444 SDL_mutexV(audio->mixer_lock); |
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445 SDL_CloseAudio(); |
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446 SDL_SetError("Couldn't create audio thread"); |
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447 return(-1); |
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448 } |
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449 |
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450 while(!audio_configured) |
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451 SDL_Delay(100); |
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452 #endif |
| 0 | 453 |
| 454 /* If the audio driver changes the buffer size, accept it */ | |
| 455 if ( audio->spec.samples != desired->samples ) { | |
| 456 desired->samples = audio->spec.samples; | |
| 457 SDL_CalculateAudioSpec(desired); | |
| 458 } | |
| 459 | |
| 460 /* Allocate a fake audio memory buffer */ | |
| 461 audio->fake_stream = SDL_AllocAudioMem(audio->spec.size); | |
| 462 if ( audio->fake_stream == NULL ) { | |
| 463 SDL_CloseAudio(); | |
| 464 SDL_OutOfMemory(); | |
| 465 return(-1); | |
| 466 } | |
| 467 | |
| 468 /* See if we need to do any conversion */ | |
|
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469 if ( obtained != NULL ) { |
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470 memcpy(obtained, &audio->spec, sizeof(audio->spec)); |
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471 } else if ( desired->freq != audio->spec.freq || |
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472 desired->format != audio->spec.format || |
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473 desired->channels != audio->spec.channels ) { |
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474 /* Build an audio conversion block */ |
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475 if ( SDL_BuildAudioCVT(&audio->convert, |
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476 desired->format, desired->channels, |
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477 desired->freq, |
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478 audio->spec.format, audio->spec.channels, |
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479 audio->spec.freq) < 0 ) { |
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480 SDL_CloseAudio(); |
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481 return(-1); |
| 0 | 482 } |
|
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483 if ( audio->convert.needed ) { |
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484 audio->convert.len = desired->size; |
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485 audio->convert.buf =(Uint8 *)SDL_AllocAudioMem( |
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486 audio->convert.len*audio->convert.len_mult); |
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487 if ( audio->convert.buf == NULL ) { |
| 0 | 488 SDL_CloseAudio(); |
|
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489 SDL_OutOfMemory(); |
| 0 | 490 return(-1); |
| 491 } | |
| 492 } | |
| 493 } | |
| 494 | |
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495 #ifndef ENABLE_AHI |
| 0 | 496 /* Start the audio thread if necessary */ |
| 497 switch (audio->opened) { | |
| 498 case 1: | |
| 499 /* Start the audio thread */ | |
| 500 audio->thread = SDL_CreateThread(SDL_RunAudio, audio); | |
| 501 if ( audio->thread == NULL ) { | |
| 502 SDL_CloseAudio(); | |
| 503 SDL_SetError("Couldn't create audio thread"); | |
| 504 return(-1); | |
| 505 } | |
| 506 break; | |
| 507 | |
| 508 default: | |
| 509 /* The audio is now playing */ | |
| 510 break; | |
| 511 } | |
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512 #else |
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513 SDL_mutexV(audio->mixer_lock); |
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514 D(bug("SDL_OpenAudio USCITA...\n")); |
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515 |
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516 #endif |
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517 |
| 0 | 518 return(0); |
| 519 } | |
| 520 | |
| 521 SDL_audiostatus SDL_GetAudioStatus(void) | |
| 522 { | |
| 523 SDL_AudioDevice *audio = current_audio; | |
| 524 SDL_audiostatus status; | |
| 525 | |
| 526 status = SDL_AUDIO_STOPPED; | |
| 527 if ( audio && audio->enabled ) { | |
| 528 if ( audio->paused ) { | |
| 529 status = SDL_AUDIO_PAUSED; | |
| 530 } else { | |
| 531 status = SDL_AUDIO_PLAYING; | |
| 532 } | |
| 533 } | |
| 534 return(status); | |
| 535 } | |
| 536 | |
| 537 void SDL_PauseAudio (int pause_on) | |
| 538 { | |
| 539 SDL_AudioDevice *audio = current_audio; | |
| 540 | |
| 541 if ( audio ) { | |
| 542 audio->paused = pause_on; | |
| 543 } | |
| 544 } | |
| 545 | |
| 546 void SDL_LockAudio (void) | |
| 547 { | |
| 548 SDL_AudioDevice *audio = current_audio; | |
| 549 | |
| 550 /* Obtain a lock on the mixing buffers */ | |
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551 if ( audio && audio->LockAudio ) { |
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552 audio->LockAudio(audio); |
| 0 | 553 } |
| 554 } | |
| 555 | |
| 556 void SDL_UnlockAudio (void) | |
| 557 { | |
| 558 SDL_AudioDevice *audio = current_audio; | |
| 559 | |
| 560 /* Release lock on the mixing buffers */ | |
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561 if ( audio && audio->UnlockAudio ) { |
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562 audio->UnlockAudio(audio); |
| 0 | 563 } |
| 564 } | |
| 565 | |
| 566 void SDL_CloseAudio (void) | |
| 567 { | |
| 568 SDL_QuitSubSystem(SDL_INIT_AUDIO); | |
| 569 } | |
| 570 | |
| 571 void SDL_AudioQuit(void) | |
| 572 { | |
| 573 SDL_AudioDevice *audio = current_audio; | |
| 574 | |
| 575 if ( audio ) { | |
| 576 audio->enabled = 0; | |
| 577 if ( audio->thread != NULL ) { | |
| 578 SDL_WaitThread(audio->thread, NULL); | |
| 579 } | |
| 580 if ( audio->mixer_lock != NULL ) { | |
| 581 SDL_DestroyMutex(audio->mixer_lock); | |
| 582 } | |
| 583 if ( audio->fake_stream != NULL ) { | |
| 584 SDL_FreeAudioMem(audio->fake_stream); | |
| 585 } | |
| 586 if ( audio->convert.needed ) { | |
| 587 SDL_FreeAudioMem(audio->convert.buf); | |
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588 |
| 0 | 589 } |
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590 #ifndef ENABLE_AHI |
| 0 | 591 if ( audio->opened ) { |
| 592 audio->CloseAudio(audio); | |
| 593 audio->opened = 0; | |
| 594 } | |
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595 #endif |
| 0 | 596 /* Free the driver data */ |
| 597 audio->free(audio); | |
| 598 current_audio = NULL; | |
| 599 } | |
| 600 } | |
| 601 | |
| 602 #define NUM_FORMATS 6 | |
| 603 static int format_idx; | |
| 604 static int format_idx_sub; | |
| 605 static Uint16 format_list[NUM_FORMATS][NUM_FORMATS] = { | |
| 606 { AUDIO_U8, AUDIO_S8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB }, | |
| 607 { AUDIO_S8, AUDIO_U8, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB }, | |
| 608 { AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, AUDIO_S8 }, | |
| 609 { AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_U8, AUDIO_S8 }, | |
| 610 { AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_U8, AUDIO_S8 }, | |
| 611 { AUDIO_U16MSB, AUDIO_U16LSB, AUDIO_S16MSB, AUDIO_S16LSB, AUDIO_U8, AUDIO_S8 }, | |
| 612 }; | |
| 613 | |
| 614 Uint16 SDL_FirstAudioFormat(Uint16 format) | |
| 615 { | |
| 616 for ( format_idx=0; format_idx < NUM_FORMATS; ++format_idx ) { | |
| 617 if ( format_list[format_idx][0] == format ) { | |
| 618 break; | |
| 619 } | |
| 620 } | |
| 621 format_idx_sub = 0; | |
| 622 return(SDL_NextAudioFormat()); | |
| 623 } | |
| 624 | |
| 625 Uint16 SDL_NextAudioFormat(void) | |
| 626 { | |
| 627 if ( (format_idx == NUM_FORMATS) || (format_idx_sub == NUM_FORMATS) ) { | |
| 628 return(0); | |
| 629 } | |
| 630 return(format_list[format_idx][format_idx_sub++]); | |
| 631 } | |
| 632 | |
| 633 void SDL_CalculateAudioSpec(SDL_AudioSpec *spec) | |
| 634 { | |
| 635 switch (spec->format) { | |
| 636 case AUDIO_U8: | |
| 637 spec->silence = 0x80; | |
| 638 break; | |
| 639 default: | |
| 640 spec->silence = 0x00; | |
| 641 break; | |
| 642 } | |
| 643 spec->size = (spec->format&0xFF)/8; | |
| 644 spec->size *= spec->channels; | |
| 645 spec->size *= spec->samples; | |
| 646 } |