Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 1338:604d73db6802
Removed uses of stdlib.h and string.h
author | Sam Lantinga <slouken@libsdl.org> |
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date | Tue, 07 Feb 2006 09:29:18 +0000 |
parents | 3692456e7b0f |
children | c71e05b4dc2e |
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 <sys/types.h> | |
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28 #include <signal.h> /* For kill() */ |
0 | 29 |
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30 #include "SDL_stdlib.h" |
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31 #include "SDL_string.h" |
0 | 32 #include "SDL_audio.h" |
33 #include "SDL_error.h" | |
34 #include "SDL_audiomem.h" | |
35 #include "SDL_audio_c.h" | |
36 #include "SDL_timer.h" | |
37 #include "SDL_alsa_audio.h" | |
38 | |
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39 #ifdef ALSA_DYNAMIC |
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40 #ifdef USE_DLVSYM |
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41 #ifndef __USE_GNU |
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42 #define __USE_GNU |
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43 #endif |
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44 #endif |
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45 #include <dlfcn.h> |
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46 #include "SDL_name.h" |
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47 #include "SDL_loadso.h" |
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48 #else |
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49 #define SDL_NAME(X) X |
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50 #endif |
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51 |
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52 |
0 | 53 /* The tag name used by ALSA audio */ |
54 #define DRIVER_NAME "alsa" | |
55 | |
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56 /* The default ALSA audio driver */ |
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57 #define DEFAULT_DEVICE "default" |
0 | 58 |
59 /* Audio driver functions */ | |
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60 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec); |
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61 static void ALSA_WaitAudio(_THIS); |
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62 static void ALSA_PlayAudio(_THIS); |
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63 static Uint8 *ALSA_GetAudioBuf(_THIS); |
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64 static void ALSA_CloseAudio(_THIS); |
0 | 65 |
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66 #ifdef ALSA_DYNAMIC |
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67 |
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68 static const char *alsa_library = ALSA_DYNAMIC; |
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69 static void *alsa_handle = NULL; |
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70 static int alsa_loaded = 0; |
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71 |
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72 static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); |
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73 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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74 static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm); |
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75 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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76 static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm); |
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77 static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm); |
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78 static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm); |
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79 static const char *(*SDL_NAME(snd_strerror))(int errnum); |
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80 static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void); |
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81 static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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82 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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83 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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84 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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85 static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params); |
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86 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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87 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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88 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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89 static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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90 static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock); |
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91 #define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof) |
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92 |
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93 /* cast funcs to char* first, to please GCC's strict aliasing rules. */ |
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94 static struct { |
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95 const char *name; |
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96 void **func; |
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97 } alsa_functions[] = { |
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98 { "snd_pcm_open", (void**)(char*)&SDL_NAME(snd_pcm_open) }, |
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99 { "snd_pcm_close", (void**)(char*)&SDL_NAME(snd_pcm_close) }, |
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100 { "snd_pcm_writei", (void**)(char*)&SDL_NAME(snd_pcm_writei) }, |
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101 { "snd_pcm_resume", (void**)(char*)&SDL_NAME(snd_pcm_resume) }, |
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102 { "snd_pcm_prepare", (void**)(char*)&SDL_NAME(snd_pcm_prepare) }, |
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103 { "snd_pcm_drain", (void**)(char*)&SDL_NAME(snd_pcm_drain) }, |
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104 { "snd_strerror", (void**)(char*)&SDL_NAME(snd_strerror) }, |
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105 { "snd_pcm_hw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof) }, |
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106 { "snd_pcm_hw_params_any", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_any) }, |
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107 { "snd_pcm_hw_params_set_access", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access) }, |
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108 { "snd_pcm_hw_params_set_format", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format) }, |
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109 { "snd_pcm_hw_params_set_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels) }, |
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110 { "snd_pcm_hw_params_get_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels) }, |
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111 { "snd_pcm_hw_params_set_rate_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near) }, |
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112 { "snd_pcm_hw_params_set_period_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) }, |
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113 { "snd_pcm_hw_params_set_periods_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near) }, |
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114 { "snd_pcm_hw_params", (void**)(char*)&SDL_NAME(snd_pcm_hw_params) }, |
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115 { "snd_pcm_nonblock", (void**)(char*)&SDL_NAME(snd_pcm_nonblock) }, |
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116 }; |
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117 |
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118 static void UnloadALSALibrary(void) { |
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119 if (alsa_loaded) { |
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120 /* SDL_UnloadObject(alsa_handle);*/ |
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121 dlclose(alsa_handle); |
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122 alsa_handle = NULL; |
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123 alsa_loaded = 0; |
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124 } |
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125 } |
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126 |
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127 static int LoadALSALibrary(void) { |
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128 int i, retval = -1; |
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129 |
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130 /* alsa_handle = SDL_LoadObject(alsa_library);*/ |
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131 alsa_handle = dlopen(alsa_library,RTLD_NOW); |
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132 if (alsa_handle) { |
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133 alsa_loaded = 1; |
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134 retval = 0; |
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135 for (i = 0; i < SDL_TABLESIZE(alsa_functions); i++) { |
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136 /* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/ |
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137 #ifdef USE_DLVSYM |
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138 *alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9"); |
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139 if (!*alsa_functions[i].func) |
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140 #endif |
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141 *alsa_functions[i].func = dlsym(alsa_handle,alsa_functions[i].name); |
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142 if (!*alsa_functions[i].func) { |
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143 retval = -1; |
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144 UnloadALSALibrary(); |
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145 break; |
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146 } |
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147 } |
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148 } |
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149 return retval; |
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150 } |
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151 |
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152 #else |
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153 |
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154 static void UnloadALSALibrary(void) { |
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155 return; |
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156 } |
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157 |
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158 static int LoadALSALibrary(void) { |
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159 return 0; |
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160 } |
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161 |
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162 #endif /* ALSA_DYNAMIC */ |
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163 |
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164 static const char *get_audio_device(int channels) |
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165 { |
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166 const char *device; |
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167 |
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168 device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */ |
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169 if ( device == NULL ) { |
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170 if (channels == 6) device = "surround51"; |
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171 else if (channels == 4) device = "surround40"; |
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172 else device = DEFAULT_DEVICE; |
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173 } |
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174 return device; |
0 | 175 } |
176 | |
177 /* Audio driver bootstrap functions */ | |
178 | |
179 static int Audio_Available(void) | |
180 { | |
181 int available; | |
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182 int status; |
0 | 183 snd_pcm_t *handle; |
184 | |
185 available = 0; | |
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186 if (LoadALSALibrary() < 0) { |
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187 return available; |
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188 } |
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189 status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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190 if ( status >= 0 ) { |
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191 available = 1; |
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192 SDL_NAME(snd_pcm_close)(handle); |
0 | 193 } |
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194 UnloadALSALibrary(); |
0 | 195 return(available); |
196 } | |
197 | |
198 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
199 { | |
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200 SDL_free(device->hidden); |
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201 SDL_free(device); |
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202 UnloadALSALibrary(); |
0 | 203 } |
204 | |
205 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
206 { | |
207 SDL_AudioDevice *this; | |
208 | |
209 /* Initialize all variables that we clean on shutdown */ | |
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210 LoadALSALibrary(); |
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211 this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); |
0 | 212 if ( this ) { |
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213 SDL_memset(this, 0, (sizeof *this)); |
0 | 214 this->hidden = (struct SDL_PrivateAudioData *) |
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215 SDL_malloc((sizeof *this->hidden)); |
0 | 216 } |
217 if ( (this == NULL) || (this->hidden == NULL) ) { | |
218 SDL_OutOfMemory(); | |
219 if ( this ) { | |
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220 SDL_free(this); |
0 | 221 } |
222 return(0); | |
223 } | |
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224 SDL_memset(this->hidden, 0, (sizeof *this->hidden)); |
0 | 225 |
226 /* Set the function pointers */ | |
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227 this->OpenAudio = ALSA_OpenAudio; |
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228 this->WaitAudio = ALSA_WaitAudio; |
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229 this->PlayAudio = ALSA_PlayAudio; |
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230 this->GetAudioBuf = ALSA_GetAudioBuf; |
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231 this->CloseAudio = ALSA_CloseAudio; |
0 | 232 |
233 this->free = Audio_DeleteDevice; | |
234 | |
235 return this; | |
236 } | |
237 | |
238 AudioBootStrap ALSA_bootstrap = { | |
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239 DRIVER_NAME, "ALSA 0.9 PCM audio", |
0 | 240 Audio_Available, Audio_CreateDevice |
241 }; | |
242 | |
243 /* This function waits until it is possible to write a full sound buffer */ | |
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244 static void ALSA_WaitAudio(_THIS) |
0 | 245 { |
246 /* Check to see if the thread-parent process is still alive */ | |
247 { static int cnt = 0; | |
248 /* Note that this only works with thread implementations | |
249 that use a different process id for each thread. | |
250 */ | |
251 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ | |
252 if ( kill(parent, 0) < 0 ) { | |
253 this->enabled = 0; | |
254 } | |
255 } | |
256 } | |
257 } | |
258 | |
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259 static void ALSA_PlayAudio(_THIS) |
0 | 260 { |
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261 int status; |
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262 int sample_len; |
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263 signed short *sample_buf; |
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264 |
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265 sample_len = this->spec.samples; |
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266 sample_buf = (signed short *)mixbuf; |
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267 while ( sample_len > 0 ) { |
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268 status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len); |
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269 if ( status < 0 ) { |
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270 if ( status == -EAGAIN ) { |
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271 SDL_Delay(1); |
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272 continue; |
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273 } |
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274 if ( status == -ESTRPIPE ) { |
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275 do { |
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276 SDL_Delay(1); |
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277 status = SDL_NAME(snd_pcm_resume)(pcm_handle); |
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278 } while ( status == -EAGAIN ); |
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279 } |
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280 if ( status < 0 ) { |
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281 status = SDL_NAME(snd_pcm_prepare)(pcm_handle); |
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282 } |
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283 if ( status < 0 ) { |
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284 /* Hmm, not much we can do - abort */ |
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285 this->enabled = 0; |
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286 return; |
0 | 287 } |
356
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288 continue; |
0 | 289 } |
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290 sample_buf += status * this->spec.channels; |
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291 sample_len -= status; |
0 | 292 } |
293 } | |
294 | |
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295 static Uint8 *ALSA_GetAudioBuf(_THIS) |
0 | 296 { |
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297 return(mixbuf); |
0 | 298 } |
299 | |
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300 static void ALSA_CloseAudio(_THIS) |
0 | 301 { |
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302 if ( mixbuf != NULL ) { |
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303 SDL_FreeAudioMem(mixbuf); |
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304 mixbuf = NULL; |
0 | 305 } |
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306 if ( pcm_handle ) { |
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307 SDL_NAME(snd_pcm_drain)(pcm_handle); |
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308 SDL_NAME(snd_pcm_close)(pcm_handle); |
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309 pcm_handle = NULL; |
0 | 310 } |
311 } | |
312 | |
354
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313 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec) |
0 | 314 { |
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315 int status; |
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316 snd_pcm_hw_params_t *params; |
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317 snd_pcm_format_t format; |
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318 snd_pcm_uframes_t frames; |
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319 Uint16 test_format; |
0 | 320 |
321 /* Open the audio device */ | |
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322 /* Name of device should depend on # channels in spec */ |
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323 status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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324 |
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325 if ( status < 0 ) { |
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326 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); |
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327 return(-1); |
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328 } |
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329 |
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330 /* Figure out what the hardware is capable of */ |
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331 snd_pcm_hw_params_alloca(¶ms); |
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332 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params); |
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333 if ( status < 0 ) { |
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334 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); |
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335 ALSA_CloseAudio(this); |
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336 return(-1); |
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337 } |
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338 |
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339 /* SDL only uses interleaved sample output */ |
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340 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); |
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341 if ( status < 0 ) { |
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342 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); |
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343 ALSA_CloseAudio(this); |
0 | 344 return(-1); |
345 } | |
346 | |
347 /* Try for a closest match on audio format */ | |
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348 status = -1; |
0 | 349 for ( test_format = SDL_FirstAudioFormat(spec->format); |
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350 test_format && (status < 0); ) { |
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351 switch ( test_format ) { |
0 | 352 case AUDIO_U8: |
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353 format = SND_PCM_FORMAT_U8; |
0 | 354 break; |
355 case AUDIO_S8: | |
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356 format = SND_PCM_FORMAT_S8; |
0 | 357 break; |
358 case AUDIO_S16LSB: | |
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359 format = SND_PCM_FORMAT_S16_LE; |
0 | 360 break; |
361 case AUDIO_S16MSB: | |
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362 format = SND_PCM_FORMAT_S16_BE; |
0 | 363 break; |
364 case AUDIO_U16LSB: | |
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365 format = SND_PCM_FORMAT_U16_LE; |
0 | 366 break; |
367 case AUDIO_U16MSB: | |
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368 format = SND_PCM_FORMAT_U16_BE; |
0 | 369 break; |
370 default: | |
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371 format = 0; |
0 | 372 break; |
373 } | |
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374 if ( format != 0 ) { |
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375 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format); |
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376 } |
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377 if ( status < 0 ) { |
0 | 378 test_format = SDL_NextAudioFormat(); |
379 } | |
380 } | |
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381 if ( status < 0 ) { |
0 | 382 SDL_SetError("Couldn't find any hardware audio formats"); |
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383 ALSA_CloseAudio(this); |
0 | 384 return(-1); |
385 } | |
386 spec->format = test_format; | |
387 | |
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388 /* Set the number of channels */ |
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389 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels); |
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390 if ( status < 0 ) { |
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391 status = SDL_NAME(snd_pcm_hw_params_get_channels)(params); |
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392 if ( (status <= 0) || (status > 2) ) { |
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393 SDL_SetError("Couldn't set audio channels"); |
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394 ALSA_CloseAudio(this); |
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395 return(-1); |
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396 } |
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397 spec->channels = status; |
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398 } |
0 | 399 |
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400 /* Set the audio rate */ |
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401 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL); |
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402 if ( status < 0 ) { |
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403 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); |
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404 ALSA_CloseAudio(this); |
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405 return(-1); |
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406 } |
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407 spec->freq = status; |
0 | 408 |
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409 /* Set the buffer size, in samples */ |
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410 frames = spec->samples; |
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411 frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL); |
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412 spec->samples = frames; |
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413 SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL); |
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414 |
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415 /* "set" the hardware with the desired parameters */ |
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416 status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params); |
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417 if ( status < 0 ) { |
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418 SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(snd_strerror)(status)); |
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419 ALSA_CloseAudio(this); |
0 | 420 return(-1); |
421 } | |
422 | |
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423 /* Calculate the final parameters for this audio specification */ |
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424 SDL_CalculateAudioSpec(spec); |
0 | 425 |
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426 /* Allocate mixing buffer */ |
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427 mixlen = spec->size; |
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428 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); |
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429 if ( mixbuf == NULL ) { |
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430 ALSA_CloseAudio(this); |
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431 return(-1); |
0 | 432 } |
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433 SDL_memset(mixbuf, spec->silence, spec->size); |
0 | 434 |
435 /* Get the parent process id (we're the parent of the audio thread) */ | |
436 parent = getpid(); | |
437 | |
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438 /* Switch to blocking mode for playback */ |
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439 SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); |
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440 |
0 | 441 /* We're ready to rock and roll. :-) */ |
442 return(0); | |
443 } |