Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 1336:3692456e7b0f
Use SDL_ prefixed versions of C library functions.
FIXME:
Change #include <stdlib.h> to #include "SDL_stdlib.h"
Change #include <string.h> to #include "SDL_string.h"
Make sure nothing else broke because of this...
author | Sam Lantinga <slouken@libsdl.org> |
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date | Tue, 07 Feb 2006 06:59:48 +0000 |
parents | 1cbaeee565b1 |
children | 604d73db6802 |
rev | line source |
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0 | 1 /* |
2 SDL - Simple DirectMedia Layer | |
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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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20 slouken@libsdl.org |
0 | 21 */ |
22 | |
23 | |
24 | |
25 /* Allow access to a raw mixing buffer */ | |
26 | |
27 #include <stdlib.h> | |
28 #include <stdio.h> | |
29 #include <string.h> | |
30 #include <errno.h> | |
31 #include <unistd.h> | |
32 #include <fcntl.h> | |
33 #include <signal.h> | |
34 #include <sys/types.h> | |
35 #include <sys/time.h> | |
36 | |
37 #include "SDL_audio.h" | |
38 #include "SDL_error.h" | |
39 #include "SDL_audiomem.h" | |
40 #include "SDL_audio_c.h" | |
41 #include "SDL_timer.h" | |
42 #include "SDL_alsa_audio.h" | |
43 | |
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44 #ifdef ALSA_DYNAMIC |
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45 #ifdef USE_DLVSYM |
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46 #ifndef __USE_GNU |
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47 #define __USE_GNU |
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48 #endif |
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49 #endif |
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50 #include <dlfcn.h> |
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51 #include "SDL_name.h" |
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52 #include "SDL_loadso.h" |
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53 #else |
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54 #define SDL_NAME(X) X |
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55 #endif |
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56 |
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57 |
0 | 58 /* The tag name used by ALSA audio */ |
59 #define DRIVER_NAME "alsa" | |
60 | |
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61 /* The default ALSA audio driver */ |
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62 #define DEFAULT_DEVICE "default" |
0 | 63 |
64 /* Audio driver functions */ | |
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65 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec); |
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66 static void ALSA_WaitAudio(_THIS); |
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67 static void ALSA_PlayAudio(_THIS); |
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68 static Uint8 *ALSA_GetAudioBuf(_THIS); |
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69 static void ALSA_CloseAudio(_THIS); |
0 | 70 |
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71 #ifdef ALSA_DYNAMIC |
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72 |
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73 static const char *alsa_library = ALSA_DYNAMIC; |
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74 static void *alsa_handle = NULL; |
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75 static int alsa_loaded = 0; |
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76 |
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77 static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); |
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78 static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); |
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79 static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm); |
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80 static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size); |
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81 static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm); |
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82 static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm); |
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83 static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm); |
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84 static const char *(*SDL_NAME(snd_strerror))(int errnum); |
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85 static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void); |
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86 static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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87 static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access); |
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88 static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val); |
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89 static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val); |
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90 static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params); |
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91 static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir); |
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92 static snd_pcm_uframes_t (*SDL_NAME(snd_pcm_hw_params_set_period_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val, int *dir); |
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93 static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir); |
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94 static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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95 static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock); |
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96 #define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof) |
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97 |
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98 /* cast funcs to char* first, to please GCC's strict aliasing rules. */ |
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99 static struct { |
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100 const char *name; |
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101 void **func; |
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102 } alsa_functions[] = { |
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103 { "snd_pcm_open", (void**)(char*)&SDL_NAME(snd_pcm_open) }, |
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104 { "snd_pcm_close", (void**)(char*)&SDL_NAME(snd_pcm_close) }, |
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105 { "snd_pcm_writei", (void**)(char*)&SDL_NAME(snd_pcm_writei) }, |
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106 { "snd_pcm_resume", (void**)(char*)&SDL_NAME(snd_pcm_resume) }, |
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107 { "snd_pcm_prepare", (void**)(char*)&SDL_NAME(snd_pcm_prepare) }, |
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108 { "snd_pcm_drain", (void**)(char*)&SDL_NAME(snd_pcm_drain) }, |
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109 { "snd_strerror", (void**)(char*)&SDL_NAME(snd_strerror) }, |
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110 { "snd_pcm_hw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof) }, |
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111 { "snd_pcm_hw_params_any", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_any) }, |
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112 { "snd_pcm_hw_params_set_access", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access) }, |
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113 { "snd_pcm_hw_params_set_format", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format) }, |
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114 { "snd_pcm_hw_params_set_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels) }, |
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115 { "snd_pcm_hw_params_get_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels) }, |
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116 { "snd_pcm_hw_params_set_rate_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near) }, |
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117 { "snd_pcm_hw_params_set_period_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) }, |
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118 { "snd_pcm_hw_params_set_periods_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near) }, |
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119 { "snd_pcm_hw_params", (void**)(char*)&SDL_NAME(snd_pcm_hw_params) }, |
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120 { "snd_pcm_nonblock", (void**)(char*)&SDL_NAME(snd_pcm_nonblock) }, |
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121 }; |
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122 |
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123 static void UnloadALSALibrary(void) { |
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124 if (alsa_loaded) { |
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125 /* SDL_UnloadObject(alsa_handle);*/ |
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126 dlclose(alsa_handle); |
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127 alsa_handle = NULL; |
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128 alsa_loaded = 0; |
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129 } |
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130 } |
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131 |
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132 static int LoadALSALibrary(void) { |
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133 int i, retval = -1; |
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134 |
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135 /* alsa_handle = SDL_LoadObject(alsa_library);*/ |
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136 alsa_handle = dlopen(alsa_library,RTLD_NOW); |
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137 if (alsa_handle) { |
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138 alsa_loaded = 1; |
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139 retval = 0; |
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140 for (i = 0; i < SDL_TABLESIZE(alsa_functions); i++) { |
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141 /* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/ |
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142 #ifdef USE_DLVSYM |
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143 *alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9"); |
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144 if (!*alsa_functions[i].func) |
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145 #endif |
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146 *alsa_functions[i].func = dlsym(alsa_handle,alsa_functions[i].name); |
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147 if (!*alsa_functions[i].func) { |
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148 retval = -1; |
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149 UnloadALSALibrary(); |
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150 break; |
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151 } |
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152 } |
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153 } |
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154 return retval; |
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155 } |
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156 |
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157 #else |
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158 |
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159 static void UnloadALSALibrary(void) { |
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160 return; |
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161 } |
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162 |
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163 static int LoadALSALibrary(void) { |
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164 return 0; |
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165 } |
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166 |
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167 #endif /* ALSA_DYNAMIC */ |
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168 |
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169 static const char *get_audio_device(int channels) |
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170 { |
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171 const char *device; |
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172 |
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173 device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */ |
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174 if ( device == NULL ) { |
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175 if (channels == 6) device = "surround51"; |
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176 else if (channels == 4) device = "surround40"; |
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177 else device = DEFAULT_DEVICE; |
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178 } |
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179 return device; |
0 | 180 } |
181 | |
182 /* Audio driver bootstrap functions */ | |
183 | |
184 static int Audio_Available(void) | |
185 { | |
186 int available; | |
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187 int status; |
0 | 188 snd_pcm_t *handle; |
189 | |
190 available = 0; | |
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191 if (LoadALSALibrary() < 0) { |
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192 return available; |
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193 } |
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194 status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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195 if ( status >= 0 ) { |
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196 available = 1; |
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197 SDL_NAME(snd_pcm_close)(handle); |
0 | 198 } |
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199 UnloadALSALibrary(); |
0 | 200 return(available); |
201 } | |
202 | |
203 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
204 { | |
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205 SDL_free(device->hidden); |
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206 SDL_free(device); |
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207 UnloadALSALibrary(); |
0 | 208 } |
209 | |
210 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
211 { | |
212 SDL_AudioDevice *this; | |
213 | |
214 /* Initialize all variables that we clean on shutdown */ | |
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215 LoadALSALibrary(); |
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216 this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); |
0 | 217 if ( this ) { |
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218 SDL_memset(this, 0, (sizeof *this)); |
0 | 219 this->hidden = (struct SDL_PrivateAudioData *) |
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220 SDL_malloc((sizeof *this->hidden)); |
0 | 221 } |
222 if ( (this == NULL) || (this->hidden == NULL) ) { | |
223 SDL_OutOfMemory(); | |
224 if ( this ) { | |
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225 SDL_free(this); |
0 | 226 } |
227 return(0); | |
228 } | |
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229 SDL_memset(this->hidden, 0, (sizeof *this->hidden)); |
0 | 230 |
231 /* Set the function pointers */ | |
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232 this->OpenAudio = ALSA_OpenAudio; |
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233 this->WaitAudio = ALSA_WaitAudio; |
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234 this->PlayAudio = ALSA_PlayAudio; |
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235 this->GetAudioBuf = ALSA_GetAudioBuf; |
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236 this->CloseAudio = ALSA_CloseAudio; |
0 | 237 |
238 this->free = Audio_DeleteDevice; | |
239 | |
240 return this; | |
241 } | |
242 | |
243 AudioBootStrap ALSA_bootstrap = { | |
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244 DRIVER_NAME, "ALSA 0.9 PCM audio", |
0 | 245 Audio_Available, Audio_CreateDevice |
246 }; | |
247 | |
248 /* This function waits until it is possible to write a full sound buffer */ | |
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249 static void ALSA_WaitAudio(_THIS) |
0 | 250 { |
251 /* Check to see if the thread-parent process is still alive */ | |
252 { static int cnt = 0; | |
253 /* Note that this only works with thread implementations | |
254 that use a different process id for each thread. | |
255 */ | |
256 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ | |
257 if ( kill(parent, 0) < 0 ) { | |
258 this->enabled = 0; | |
259 } | |
260 } | |
261 } | |
262 } | |
263 | |
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264 static void ALSA_PlayAudio(_THIS) |
0 | 265 { |
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266 int status; |
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267 int sample_len; |
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268 signed short *sample_buf; |
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269 |
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270 sample_len = this->spec.samples; |
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271 sample_buf = (signed short *)mixbuf; |
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272 while ( sample_len > 0 ) { |
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273 status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len); |
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274 if ( status < 0 ) { |
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275 if ( status == -EAGAIN ) { |
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276 SDL_Delay(1); |
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277 continue; |
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278 } |
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279 if ( status == -ESTRPIPE ) { |
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280 do { |
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281 SDL_Delay(1); |
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282 status = SDL_NAME(snd_pcm_resume)(pcm_handle); |
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283 } while ( status == -EAGAIN ); |
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284 } |
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285 if ( status < 0 ) { |
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286 status = SDL_NAME(snd_pcm_prepare)(pcm_handle); |
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287 } |
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288 if ( status < 0 ) { |
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289 /* Hmm, not much we can do - abort */ |
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290 this->enabled = 0; |
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291 return; |
0 | 292 } |
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293 continue; |
0 | 294 } |
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295 sample_buf += status * this->spec.channels; |
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296 sample_len -= status; |
0 | 297 } |
298 } | |
299 | |
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300 static Uint8 *ALSA_GetAudioBuf(_THIS) |
0 | 301 { |
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302 return(mixbuf); |
0 | 303 } |
304 | |
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305 static void ALSA_CloseAudio(_THIS) |
0 | 306 { |
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307 if ( mixbuf != NULL ) { |
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308 SDL_FreeAudioMem(mixbuf); |
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309 mixbuf = NULL; |
0 | 310 } |
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311 if ( pcm_handle ) { |
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312 SDL_NAME(snd_pcm_drain)(pcm_handle); |
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313 SDL_NAME(snd_pcm_close)(pcm_handle); |
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314 pcm_handle = NULL; |
0 | 315 } |
316 } | |
317 | |
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318 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec) |
0 | 319 { |
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320 int status; |
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321 snd_pcm_hw_params_t *params; |
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322 snd_pcm_format_t format; |
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323 snd_pcm_uframes_t frames; |
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324 Uint16 test_format; |
0 | 325 |
326 /* Open the audio device */ | |
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327 /* Name of device should depend on # channels in spec */ |
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328 status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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329 |
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330 if ( status < 0 ) { |
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331 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); |
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332 return(-1); |
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333 } |
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334 |
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335 /* Figure out what the hardware is capable of */ |
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336 snd_pcm_hw_params_alloca(¶ms); |
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337 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params); |
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338 if ( status < 0 ) { |
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339 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); |
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340 ALSA_CloseAudio(this); |
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341 return(-1); |
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342 } |
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343 |
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344 /* SDL only uses interleaved sample output */ |
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345 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); |
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346 if ( status < 0 ) { |
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347 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); |
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348 ALSA_CloseAudio(this); |
0 | 349 return(-1); |
350 } | |
351 | |
352 /* Try for a closest match on audio format */ | |
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353 status = -1; |
0 | 354 for ( test_format = SDL_FirstAudioFormat(spec->format); |
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355 test_format && (status < 0); ) { |
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356 switch ( test_format ) { |
0 | 357 case AUDIO_U8: |
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358 format = SND_PCM_FORMAT_U8; |
0 | 359 break; |
360 case AUDIO_S8: | |
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361 format = SND_PCM_FORMAT_S8; |
0 | 362 break; |
363 case AUDIO_S16LSB: | |
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364 format = SND_PCM_FORMAT_S16_LE; |
0 | 365 break; |
366 case AUDIO_S16MSB: | |
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367 format = SND_PCM_FORMAT_S16_BE; |
0 | 368 break; |
369 case AUDIO_U16LSB: | |
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370 format = SND_PCM_FORMAT_U16_LE; |
0 | 371 break; |
372 case AUDIO_U16MSB: | |
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373 format = SND_PCM_FORMAT_U16_BE; |
0 | 374 break; |
375 default: | |
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376 format = 0; |
0 | 377 break; |
378 } | |
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379 if ( format != 0 ) { |
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380 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format); |
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381 } |
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382 if ( status < 0 ) { |
0 | 383 test_format = SDL_NextAudioFormat(); |
384 } | |
385 } | |
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386 if ( status < 0 ) { |
0 | 387 SDL_SetError("Couldn't find any hardware audio formats"); |
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388 ALSA_CloseAudio(this); |
0 | 389 return(-1); |
390 } | |
391 spec->format = test_format; | |
392 | |
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393 /* Set the number of channels */ |
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394 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels); |
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395 if ( status < 0 ) { |
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396 status = SDL_NAME(snd_pcm_hw_params_get_channels)(params); |
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397 if ( (status <= 0) || (status > 2) ) { |
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398 SDL_SetError("Couldn't set audio channels"); |
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399 ALSA_CloseAudio(this); |
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400 return(-1); |
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401 } |
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402 spec->channels = status; |
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403 } |
0 | 404 |
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405 /* Set the audio rate */ |
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406 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL); |
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407 if ( status < 0 ) { |
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408 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); |
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409 ALSA_CloseAudio(this); |
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410 return(-1); |
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411 } |
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412 spec->freq = status; |
0 | 413 |
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414 /* Set the buffer size, in samples */ |
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415 frames = spec->samples; |
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416 frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL); |
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417 spec->samples = frames; |
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418 SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL); |
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419 |
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420 /* "set" the hardware with the desired parameters */ |
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421 status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params); |
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422 if ( status < 0 ) { |
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423 SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(snd_strerror)(status)); |
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424 ALSA_CloseAudio(this); |
0 | 425 return(-1); |
426 } | |
427 | |
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428 /* Calculate the final parameters for this audio specification */ |
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429 SDL_CalculateAudioSpec(spec); |
0 | 430 |
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431 /* Allocate mixing buffer */ |
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432 mixlen = spec->size; |
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433 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); |
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434 if ( mixbuf == NULL ) { |
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435 ALSA_CloseAudio(this); |
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436 return(-1); |
0 | 437 } |
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438 SDL_memset(mixbuf, spec->silence, spec->size); |
0 | 439 |
440 /* Get the parent process id (we're the parent of the audio thread) */ | |
441 parent = getpid(); | |
442 | |
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443 /* Switch to blocking mode for playback */ |
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444 SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); |
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445 |
0 | 446 /* We're ready to rock and roll. :-) */ |
447 return(0); | |
448 } |