Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 4070:b8f2db95145e SDL-1.2
Patch from Christian Walther
Yes, the idea to use a cursor rectangle instead of [NSCursor set] has occurred
to me too, and it does seem to be the most elegant way. Here's my attempt at an
implementation
author | Sam Lantinga <slouken@libsdl.org> |
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date | Sun, 15 Jul 2007 15:58:00 +0000 |
parents | d7c9d7f42881 |
children | c121d94672cb a1b03ba2fcd0 |
rev | line source |
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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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20 slouken@libsdl.org |
0 | 21 */ |
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22 #include "SDL_config.h" |
0 | 23 |
24 /* Allow access to a raw mixing buffer */ | |
25 | |
26 #include <sys/types.h> | |
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27 #include <signal.h> /* For kill() */ |
0 | 28 |
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29 #include "SDL_timer.h" |
0 | 30 #include "SDL_audio.h" |
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31 #include "../SDL_audiomem.h" |
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32 #include "../SDL_audio_c.h" |
0 | 33 #include "SDL_alsa_audio.h" |
34 | |
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35 #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
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36 #include <dlfcn.h> |
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37 #include "SDL_name.h" |
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38 #include "SDL_loadso.h" |
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39 #else |
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40 #define SDL_NAME(X) X |
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41 #endif |
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42 |
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43 |
0 | 44 /* The tag name used by ALSA audio */ |
45 #define DRIVER_NAME "alsa" | |
46 | |
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47 /* The default ALSA audio driver */ |
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48 #define DEFAULT_DEVICE "default" |
0 | 49 |
50 /* Audio driver functions */ | |
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51 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec); |
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52 static void ALSA_WaitAudio(_THIS); |
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53 static void ALSA_PlayAudio(_THIS); |
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54 static Uint8 *ALSA_GetAudioBuf(_THIS); |
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55 static void ALSA_CloseAudio(_THIS); |
0 | 56 |
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57 #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
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58 |
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59 static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC; |
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60 static void *alsa_handle = NULL; |
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61 static int alsa_loaded = 0; |
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62 |
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63 static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); |
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64 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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65 static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm); |
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66 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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67 static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm); |
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68 static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm); |
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69 static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm); |
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70 static const char *(*SDL_NAME(snd_strerror))(int errnum); |
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71 static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void); |
1552 | 72 static size_t (*SDL_NAME(snd_pcm_sw_params_sizeof))(void); |
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73 static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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74 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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75 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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76 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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77 static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params); |
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78 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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79 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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80 static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params); |
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81 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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82 static int (*SDL_NAME(snd_pcm_hw_params_get_periods))(snd_pcm_hw_params_t *params); |
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83 static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
1552 | 84 /* |
85 */ | |
86 static int (*SDL_NAME(snd_pcm_sw_params_current))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams); | |
87 static int (*SDL_NAME(snd_pcm_sw_params_set_start_threshold))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val); | |
88 static int (*SDL_NAME(snd_pcm_sw_params_set_avail_min))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val); | |
89 static int (*SDL_NAME(snd_pcm_sw_params))(snd_pcm_t *pcm, snd_pcm_sw_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) |
1552 | 92 #define snd_pcm_sw_params_sizeof SDL_NAME(snd_pcm_sw_params_sizeof) |
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93 |
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94 /* cast funcs to char* first, to please GCC's strict aliasing rules. */ |
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95 static struct { |
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96 const char *name; |
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97 void **func; |
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98 } alsa_functions[] = { |
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99 { "snd_pcm_open", (void**)(char*)&SDL_NAME(snd_pcm_open) }, |
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100 { "snd_pcm_close", (void**)(char*)&SDL_NAME(snd_pcm_close) }, |
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101 { "snd_pcm_writei", (void**)(char*)&SDL_NAME(snd_pcm_writei) }, |
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102 { "snd_pcm_resume", (void**)(char*)&SDL_NAME(snd_pcm_resume) }, |
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103 { "snd_pcm_prepare", (void**)(char*)&SDL_NAME(snd_pcm_prepare) }, |
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104 { "snd_pcm_drain", (void**)(char*)&SDL_NAME(snd_pcm_drain) }, |
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105 { "snd_strerror", (void**)(char*)&SDL_NAME(snd_strerror) }, |
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106 { "snd_pcm_hw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof) }, |
1552 | 107 { "snd_pcm_sw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_sizeof) }, |
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108 { "snd_pcm_hw_params_any", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_any) }, |
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109 { "snd_pcm_hw_params_set_access", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access) }, |
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110 { "snd_pcm_hw_params_set_format", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format) }, |
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111 { "snd_pcm_hw_params_set_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels) }, |
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112 { "snd_pcm_hw_params_get_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels) }, |
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113 { "snd_pcm_hw_params_set_rate_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near) }, |
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114 { "snd_pcm_hw_params_set_period_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) }, |
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115 { "snd_pcm_hw_params_get_period_size", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_period_size) }, |
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116 { "snd_pcm_hw_params_set_periods_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near) }, |
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117 { "snd_pcm_hw_params_get_periods", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_periods) }, |
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118 { "snd_pcm_hw_params", (void**)(char*)&SDL_NAME(snd_pcm_hw_params) }, |
1552 | 119 { "snd_pcm_sw_params_current", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_current) }, |
120 { "snd_pcm_sw_params_set_start_threshold", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_start_threshold) }, | |
121 { "snd_pcm_sw_params_set_avail_min", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_avail_min) }, | |
122 { "snd_pcm_sw_params", (void**)(char*)&SDL_NAME(snd_pcm_sw_params) }, | |
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123 { "snd_pcm_nonblock", (void**)(char*)&SDL_NAME(snd_pcm_nonblock) }, |
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124 }; |
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125 |
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126 static void UnloadALSALibrary(void) { |
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127 if (alsa_loaded) { |
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128 /* SDL_UnloadObject(alsa_handle);*/ |
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129 dlclose(alsa_handle); |
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130 alsa_handle = NULL; |
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131 alsa_loaded = 0; |
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132 } |
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133 } |
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134 |
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135 static int LoadALSALibrary(void) { |
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136 int i, retval = -1; |
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137 |
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138 /* alsa_handle = SDL_LoadObject(alsa_library);*/ |
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139 alsa_handle = dlopen(alsa_library,RTLD_NOW); |
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140 if (alsa_handle) { |
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141 alsa_loaded = 1; |
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142 retval = 0; |
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143 for (i = 0; i < SDL_arraysize(alsa_functions); i++) { |
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144 /* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/ |
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145 #if HAVE_DLVSYM |
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146 *alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9"); |
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147 if (!*alsa_functions[i].func) |
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148 #endif |
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149 *alsa_functions[i].func = dlsym(alsa_handle,alsa_functions[i].name); |
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150 if (!*alsa_functions[i].func) { |
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151 retval = -1; |
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152 UnloadALSALibrary(); |
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153 break; |
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154 } |
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155 } |
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156 } |
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157 return retval; |
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158 } |
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159 |
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160 #else |
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161 |
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162 static void UnloadALSALibrary(void) { |
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163 return; |
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164 } |
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165 |
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166 static int LoadALSALibrary(void) { |
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167 return 0; |
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168 } |
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169 |
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170 #endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */ |
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171 |
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172 static const char *get_audio_device(int channels) |
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173 { |
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174 const char *device; |
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175 |
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176 device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */ |
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177 if ( device == NULL ) { |
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178 if (channels == 6) device = "surround51"; |
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179 else if (channels == 4) device = "surround40"; |
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180 else device = DEFAULT_DEVICE; |
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181 } |
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182 return device; |
0 | 183 } |
184 | |
185 /* Audio driver bootstrap functions */ | |
186 | |
187 static int Audio_Available(void) | |
188 { | |
189 int available; | |
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190 int status; |
0 | 191 snd_pcm_t *handle; |
192 | |
193 available = 0; | |
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194 if (LoadALSALibrary() < 0) { |
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195 return available; |
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196 } |
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197 status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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198 if ( status >= 0 ) { |
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199 available = 1; |
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200 SDL_NAME(snd_pcm_close)(handle); |
0 | 201 } |
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202 UnloadALSALibrary(); |
0 | 203 return(available); |
204 } | |
205 | |
206 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
207 { | |
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208 SDL_free(device->hidden); |
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209 SDL_free(device); |
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210 UnloadALSALibrary(); |
0 | 211 } |
212 | |
213 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
214 { | |
215 SDL_AudioDevice *this; | |
216 | |
217 /* Initialize all variables that we clean on shutdown */ | |
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218 LoadALSALibrary(); |
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219 this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); |
0 | 220 if ( this ) { |
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221 SDL_memset(this, 0, (sizeof *this)); |
0 | 222 this->hidden = (struct SDL_PrivateAudioData *) |
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223 SDL_malloc((sizeof *this->hidden)); |
0 | 224 } |
225 if ( (this == NULL) || (this->hidden == NULL) ) { | |
226 SDL_OutOfMemory(); | |
227 if ( this ) { | |
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228 SDL_free(this); |
0 | 229 } |
230 return(0); | |
231 } | |
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232 SDL_memset(this->hidden, 0, (sizeof *this->hidden)); |
0 | 233 |
234 /* Set the function pointers */ | |
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235 this->OpenAudio = ALSA_OpenAudio; |
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236 this->WaitAudio = ALSA_WaitAudio; |
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237 this->PlayAudio = ALSA_PlayAudio; |
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238 this->GetAudioBuf = ALSA_GetAudioBuf; |
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239 this->CloseAudio = ALSA_CloseAudio; |
0 | 240 |
241 this->free = Audio_DeleteDevice; | |
242 | |
243 return this; | |
244 } | |
245 | |
246 AudioBootStrap ALSA_bootstrap = { | |
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247 DRIVER_NAME, "ALSA 0.9 PCM audio", |
0 | 248 Audio_Available, Audio_CreateDevice |
249 }; | |
250 | |
251 /* This function waits until it is possible to write a full sound buffer */ | |
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252 static void ALSA_WaitAudio(_THIS) |
0 | 253 { |
254 /* Check to see if the thread-parent process is still alive */ | |
255 { static int cnt = 0; | |
256 /* Note that this only works with thread implementations | |
257 that use a different process id for each thread. | |
258 */ | |
259 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ | |
260 if ( kill(parent, 0) < 0 ) { | |
261 this->enabled = 0; | |
262 } | |
263 } | |
264 } | |
265 } | |
266 | |
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267 |
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268 /* |
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269 * http://bugzilla.libsdl.org/show_bug.cgi?id=110 |
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270 * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE |
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271 * and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR" |
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272 */ |
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273 #define SWIZ6(T) \ |
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274 T *ptr = (T *) mixbuf; \ |
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275 const Uint32 count = (this->spec.samples / 6); \ |
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276 Uint32 i; \ |
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277 for (i = 0; i < count; i++, ptr += 6) { \ |
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278 T tmp; \ |
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279 tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \ |
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280 tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \ |
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281 } |
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282 |
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283 static __inline__ void swizzle_alsa_channels_6_64bit(_THIS) { SWIZ6(Uint64); } |
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284 static __inline__ void swizzle_alsa_channels_6_32bit(_THIS) { SWIZ6(Uint32); } |
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285 static __inline__ void swizzle_alsa_channels_6_16bit(_THIS) { SWIZ6(Uint16); } |
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286 static __inline__ void swizzle_alsa_channels_6_8bit(_THIS) { SWIZ6(Uint8); } |
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287 |
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288 #undef SWIZ6 |
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289 |
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290 |
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291 /* |
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292 * Called right before feeding this->mixbuf to the hardware. Swizzle channels |
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293 * from Windows/Mac order to the format alsalib will want. |
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294 */ |
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295 static __inline__ void swizzle_alsa_channels(_THIS) |
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296 { |
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297 if (this->spec.channels == 6) { |
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298 const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */ |
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299 if (fmtsize == 16) |
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300 swizzle_alsa_channels_6_16bit(this); |
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301 else if (fmtsize == 8) |
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302 swizzle_alsa_channels_6_8bit(this); |
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303 else if (fmtsize == 32) |
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304 swizzle_alsa_channels_6_32bit(this); |
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305 else if (fmtsize == 64) |
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306 swizzle_alsa_channels_6_64bit(this); |
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307 } |
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308 |
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309 /* !!! FIXME: update this for 7.1 if needed, later. */ |
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310 } |
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311 |
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312 |
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313 static void ALSA_PlayAudio(_THIS) |
0 | 314 { |
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315 int status; |
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316 int sample_len; |
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317 signed short *sample_buf; |
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318 |
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319 swizzle_alsa_channels(this); |
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320 |
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321 sample_len = this->spec.samples; |
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322 sample_buf = (signed short *)mixbuf; |
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323 |
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324 while ( sample_len > 0 ) { |
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325 status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len); |
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326 if ( status < 0 ) { |
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327 if ( status == -EAGAIN ) { |
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328 SDL_Delay(1); |
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329 continue; |
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330 } |
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331 if ( status == -ESTRPIPE ) { |
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332 do { |
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333 SDL_Delay(1); |
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334 status = SDL_NAME(snd_pcm_resume)(pcm_handle); |
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335 } while ( status == -EAGAIN ); |
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336 } |
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337 if ( status < 0 ) { |
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338 status = SDL_NAME(snd_pcm_prepare)(pcm_handle); |
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339 } |
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340 if ( status < 0 ) { |
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341 /* Hmm, not much we can do - abort */ |
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342 this->enabled = 0; |
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343 return; |
0 | 344 } |
356
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345 continue; |
0 | 346 } |
354
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347 sample_buf += status * this->spec.channels; |
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348 sample_len -= status; |
0 | 349 } |
350 } | |
351 | |
354
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352 static Uint8 *ALSA_GetAudioBuf(_THIS) |
0 | 353 { |
354
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354 return(mixbuf); |
0 | 355 } |
356 | |
354
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357 static void ALSA_CloseAudio(_THIS) |
0 | 358 { |
354
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359 if ( mixbuf != NULL ) { |
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360 SDL_FreeAudioMem(mixbuf); |
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361 mixbuf = NULL; |
0 | 362 } |
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363 if ( pcm_handle ) { |
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364 SDL_NAME(snd_pcm_drain)(pcm_handle); |
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365 SDL_NAME(snd_pcm_close)(pcm_handle); |
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366 pcm_handle = NULL; |
0 | 367 } |
368 } | |
369 | |
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370 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec) |
0 | 371 { |
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372 int status; |
1552 | 373 snd_pcm_hw_params_t *hwparams; |
374 snd_pcm_sw_params_t *swparams; | |
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375 snd_pcm_format_t format; |
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376 snd_pcm_uframes_t frames; |
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377 Uint16 test_format; |
0 | 378 |
379 /* Open the audio device */ | |
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380 /* Name of device should depend on # channels in spec */ |
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381 status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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382 |
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383 if ( status < 0 ) { |
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384 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); |
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385 return(-1); |
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386 } |
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387 |
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388 /* Figure out what the hardware is capable of */ |
1552 | 389 snd_pcm_hw_params_alloca(&hwparams); |
390 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams); | |
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391 if ( status < 0 ) { |
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392 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); |
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393 ALSA_CloseAudio(this); |
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394 return(-1); |
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395 } |
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396 |
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397 /* SDL only uses interleaved sample output */ |
1552 | 398 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); |
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399 if ( status < 0 ) { |
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400 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); |
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401 ALSA_CloseAudio(this); |
0 | 402 return(-1); |
403 } | |
404 | |
405 /* Try for a closest match on audio format */ | |
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406 status = -1; |
0 | 407 for ( test_format = SDL_FirstAudioFormat(spec->format); |
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408 test_format && (status < 0); ) { |
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409 switch ( test_format ) { |
0 | 410 case AUDIO_U8: |
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411 format = SND_PCM_FORMAT_U8; |
0 | 412 break; |
413 case AUDIO_S8: | |
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414 format = SND_PCM_FORMAT_S8; |
0 | 415 break; |
416 case AUDIO_S16LSB: | |
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417 format = SND_PCM_FORMAT_S16_LE; |
0 | 418 break; |
419 case AUDIO_S16MSB: | |
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420 format = SND_PCM_FORMAT_S16_BE; |
0 | 421 break; |
422 case AUDIO_U16LSB: | |
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423 format = SND_PCM_FORMAT_U16_LE; |
0 | 424 break; |
425 case AUDIO_U16MSB: | |
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426 format = SND_PCM_FORMAT_U16_BE; |
0 | 427 break; |
428 default: | |
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429 format = 0; |
0 | 430 break; |
431 } | |
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432 if ( format != 0 ) { |
1552 | 433 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format); |
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434 } |
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435 if ( status < 0 ) { |
0 | 436 test_format = SDL_NextAudioFormat(); |
437 } | |
438 } | |
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439 if ( status < 0 ) { |
0 | 440 SDL_SetError("Couldn't find any hardware audio formats"); |
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441 ALSA_CloseAudio(this); |
0 | 442 return(-1); |
443 } | |
444 spec->format = test_format; | |
445 | |
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446 /* Set the number of channels */ |
1552 | 447 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels); |
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448 if ( status < 0 ) { |
1552 | 449 status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams); |
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450 if ( (status <= 0) || (status > 2) ) { |
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451 SDL_SetError("Couldn't set audio channels"); |
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452 ALSA_CloseAudio(this); |
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453 return(-1); |
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454 } |
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455 spec->channels = status; |
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456 } |
0 | 457 |
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458 /* Set the audio rate */ |
1552 | 459 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, spec->freq, NULL); |
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460 if ( status < 0 ) { |
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461 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); |
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462 ALSA_CloseAudio(this); |
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463 return(-1); |
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464 } |
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465 spec->freq = status; |
0 | 466 |
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467 /* Set the buffer size, in samples */ |
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468 frames = spec->samples; |
1552 | 469 frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, frames, NULL); |
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470 spec->samples = frames; |
1552 | 471 SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, 2, NULL); |
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472 |
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473 /* "set" the hardware with the desired parameters */ |
1552 | 474 status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams); |
475 if ( status < 0 ) { | |
476 SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status)); | |
477 ALSA_CloseAudio(this); | |
478 return(-1); | |
479 } | |
480 | |
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481 /* This is useful for debugging... */ |
1552 | 482 /* |
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483 { snd_pcm_sframes_t bufsize; int fragments; |
1552 | 484 bufsize = SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams); |
485 fragments = SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams); | |
486 | |
487 fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, fragments); | |
488 } | |
489 */ | |
490 | |
491 /* Set the software parameters */ | |
492 snd_pcm_sw_params_alloca(&swparams); | |
493 status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams); | |
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494 if ( status < 0 ) { |
1552 | 495 SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status)); |
496 ALSA_CloseAudio(this); | |
497 return(-1); | |
498 } | |
499 status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 0); | |
500 if ( status < 0 ) { | |
501 SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status)); | |
502 ALSA_CloseAudio(this); | |
503 return(-1); | |
504 } | |
505 status = SDL_NAME(snd_pcm_sw_params_set_avail_min)(pcm_handle, swparams, frames); | |
506 if ( status < 0 ) { | |
507 SDL_SetError("Couldn't set avail min: %s", SDL_NAME(snd_strerror)(status)); | |
508 ALSA_CloseAudio(this); | |
509 return(-1); | |
510 } | |
511 status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams); | |
512 if ( status < 0 ) { | |
513 SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status)); | |
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514 ALSA_CloseAudio(this); |
0 | 515 return(-1); |
516 } | |
517 | |
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518 /* Calculate the final parameters for this audio specification */ |
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519 SDL_CalculateAudioSpec(spec); |
0 | 520 |
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521 /* Allocate mixing buffer */ |
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522 mixlen = spec->size; |
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523 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); |
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524 if ( mixbuf == NULL ) { |
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525 ALSA_CloseAudio(this); |
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526 return(-1); |
0 | 527 } |
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528 SDL_memset(mixbuf, spec->silence, spec->size); |
0 | 529 |
530 /* Get the parent process id (we're the parent of the audio thread) */ | |
531 parent = getpid(); | |
532 | |
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533 /* Switch to blocking mode for playback */ |
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534 SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); |
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535 |
0 | 536 /* We're ready to rock and roll. :-) */ |
537 return(0); | |
538 } |