Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 1768:814f9f2c7a33
Fixed bug #80
Date: 21 Apr 2003 17:20:20 +0100
From: Alan Swanson <swanson@uklinux.net>
Subject: [SDL] New XFree 4.3 Video Mode Patch
If you look at the unsorted list of modes returned by X, here's mine;
1280 x 1024 @ 85.0 >
1024 x 768 @ 100.3 > USER
800 x 600 @ 125.5 > SET
640 x 480 @ 124.9 >
1280 x 1024 @ 75.0 ]
1280 x 1024 @ 60.0 ]
1280 x 960 @ 85.0 ] X11
1280 x 960 @ 60.0 ] AUTO
1152 x 864 @ 75.0 ]=20
1152 x 768 @ 54.8 ]
960 x 720 @ 120.0 ]
...
640 x 400 @ 85.1 ] 256k
576 x 432 @ 150.0 ] 249k PIXEL
640 x 350 @ 85.1 ] 224k COUNT
576 x 384 @ 109.6 ] 221k
...
The user set modes come first followed by X set modes which are ordered
by decreasing number of pixels and refresh.
The reason why every other library or program not using SDL working is
due to SDL scanning the modes in reverse getting X11 provided modes
modes with the lowest refresh.
author | Sam Lantinga <slouken@libsdl.org> |
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date | Fri, 05 May 2006 05:50:26 +0000 |
parents | 63fa37538842 |
children | 782fd950bd46 d7c9d7f42881 |
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 static void ALSA_PlayAudio(_THIS) |
0 | 268 { |
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269 int status; |
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270 int sample_len; |
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271 signed short *sample_buf; |
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272 |
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273 sample_len = this->spec.samples; |
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274 sample_buf = (signed short *)mixbuf; |
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275 while ( sample_len > 0 ) { |
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276 status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len); |
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277 if ( status < 0 ) { |
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278 if ( status == -EAGAIN ) { |
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279 SDL_Delay(1); |
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280 continue; |
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281 } |
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282 if ( status == -ESTRPIPE ) { |
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283 do { |
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284 SDL_Delay(1); |
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285 status = SDL_NAME(snd_pcm_resume)(pcm_handle); |
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286 } while ( status == -EAGAIN ); |
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287 } |
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288 if ( status < 0 ) { |
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289 status = SDL_NAME(snd_pcm_prepare)(pcm_handle); |
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290 } |
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291 if ( status < 0 ) { |
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292 /* Hmm, not much we can do - abort */ |
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293 this->enabled = 0; |
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294 return; |
0 | 295 } |
356
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296 continue; |
0 | 297 } |
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298 sample_buf += status * this->spec.channels; |
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299 sample_len -= status; |
0 | 300 } |
301 } | |
302 | |
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303 static Uint8 *ALSA_GetAudioBuf(_THIS) |
0 | 304 { |
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305 return(mixbuf); |
0 | 306 } |
307 | |
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308 static void ALSA_CloseAudio(_THIS) |
0 | 309 { |
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310 if ( mixbuf != NULL ) { |
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311 SDL_FreeAudioMem(mixbuf); |
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312 mixbuf = NULL; |
0 | 313 } |
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314 if ( pcm_handle ) { |
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315 SDL_NAME(snd_pcm_drain)(pcm_handle); |
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316 SDL_NAME(snd_pcm_close)(pcm_handle); |
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317 pcm_handle = NULL; |
0 | 318 } |
319 } | |
320 | |
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321 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec) |
0 | 322 { |
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323 int status; |
1552 | 324 snd_pcm_hw_params_t *hwparams; |
325 snd_pcm_sw_params_t *swparams; | |
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326 snd_pcm_format_t format; |
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327 snd_pcm_uframes_t frames; |
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328 Uint16 test_format; |
0 | 329 |
330 /* Open the audio device */ | |
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331 /* Name of device should depend on # channels in spec */ |
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332 status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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333 |
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334 if ( status < 0 ) { |
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335 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); |
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336 return(-1); |
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337 } |
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338 |
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339 /* Figure out what the hardware is capable of */ |
1552 | 340 snd_pcm_hw_params_alloca(&hwparams); |
341 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams); | |
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342 if ( status < 0 ) { |
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343 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); |
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344 ALSA_CloseAudio(this); |
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345 return(-1); |
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346 } |
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347 |
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348 /* SDL only uses interleaved sample output */ |
1552 | 349 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); |
354
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350 if ( status < 0 ) { |
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351 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); |
354
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352 ALSA_CloseAudio(this); |
0 | 353 return(-1); |
354 } | |
355 | |
356 /* Try for a closest match on audio format */ | |
354
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357 status = -1; |
0 | 358 for ( test_format = SDL_FirstAudioFormat(spec->format); |
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359 test_format && (status < 0); ) { |
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360 switch ( test_format ) { |
0 | 361 case AUDIO_U8: |
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362 format = SND_PCM_FORMAT_U8; |
0 | 363 break; |
364 case AUDIO_S8: | |
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365 format = SND_PCM_FORMAT_S8; |
0 | 366 break; |
367 case AUDIO_S16LSB: | |
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368 format = SND_PCM_FORMAT_S16_LE; |
0 | 369 break; |
370 case AUDIO_S16MSB: | |
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371 format = SND_PCM_FORMAT_S16_BE; |
0 | 372 break; |
373 case AUDIO_U16LSB: | |
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374 format = SND_PCM_FORMAT_U16_LE; |
0 | 375 break; |
376 case AUDIO_U16MSB: | |
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377 format = SND_PCM_FORMAT_U16_BE; |
0 | 378 break; |
379 default: | |
354
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380 format = 0; |
0 | 381 break; |
382 } | |
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383 if ( format != 0 ) { |
1552 | 384 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format); |
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385 } |
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386 if ( status < 0 ) { |
0 | 387 test_format = SDL_NextAudioFormat(); |
388 } | |
389 } | |
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390 if ( status < 0 ) { |
0 | 391 SDL_SetError("Couldn't find any hardware audio formats"); |
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392 ALSA_CloseAudio(this); |
0 | 393 return(-1); |
394 } | |
395 spec->format = test_format; | |
396 | |
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397 /* Set the number of channels */ |
1552 | 398 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels); |
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399 if ( status < 0 ) { |
1552 | 400 status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams); |
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401 if ( (status <= 0) || (status > 2) ) { |
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402 SDL_SetError("Couldn't set audio channels"); |
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403 ALSA_CloseAudio(this); |
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404 return(-1); |
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405 } |
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406 spec->channels = status; |
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407 } |
0 | 408 |
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409 /* Set the audio rate */ |
1552 | 410 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, spec->freq, NULL); |
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411 if ( status < 0 ) { |
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412 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); |
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413 ALSA_CloseAudio(this); |
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414 return(-1); |
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415 } |
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416 spec->freq = status; |
0 | 417 |
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418 /* Set the buffer size, in samples */ |
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419 frames = spec->samples; |
1552 | 420 frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, frames, NULL); |
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421 spec->samples = frames; |
1552 | 422 SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, 2, NULL); |
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423 |
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424 /* "set" the hardware with the desired parameters */ |
1552 | 425 status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams); |
426 if ( status < 0 ) { | |
427 SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status)); | |
428 ALSA_CloseAudio(this); | |
429 return(-1); | |
430 } | |
431 | |
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432 /* This is useful for debugging... */ |
1552 | 433 /* |
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434 { snd_pcm_sframes_t bufsize; int fragments; |
1552 | 435 bufsize = SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams); |
436 fragments = SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams); | |
437 | |
438 fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, fragments); | |
439 } | |
440 */ | |
441 | |
442 /* Set the software parameters */ | |
443 snd_pcm_sw_params_alloca(&swparams); | |
444 status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams); | |
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445 if ( status < 0 ) { |
1552 | 446 SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status)); |
447 ALSA_CloseAudio(this); | |
448 return(-1); | |
449 } | |
450 status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 0); | |
451 if ( status < 0 ) { | |
452 SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status)); | |
453 ALSA_CloseAudio(this); | |
454 return(-1); | |
455 } | |
456 status = SDL_NAME(snd_pcm_sw_params_set_avail_min)(pcm_handle, swparams, frames); | |
457 if ( status < 0 ) { | |
458 SDL_SetError("Couldn't set avail min: %s", SDL_NAME(snd_strerror)(status)); | |
459 ALSA_CloseAudio(this); | |
460 return(-1); | |
461 } | |
462 status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams); | |
463 if ( status < 0 ) { | |
464 SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status)); | |
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465 ALSA_CloseAudio(this); |
0 | 466 return(-1); |
467 } | |
468 | |
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469 /* Calculate the final parameters for this audio specification */ |
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470 SDL_CalculateAudioSpec(spec); |
0 | 471 |
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472 /* Allocate mixing buffer */ |
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473 mixlen = spec->size; |
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474 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); |
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475 if ( mixbuf == NULL ) { |
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476 ALSA_CloseAudio(this); |
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477 return(-1); |
0 | 478 } |
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479 SDL_memset(mixbuf, spec->silence, spec->size); |
0 | 480 |
481 /* Get the parent process id (we're the parent of the audio thread) */ | |
482 parent = getpid(); | |
483 | |
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484 /* Switch to blocking mode for playback */ |
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485 SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); |
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486 |
0 | 487 /* We're ready to rock and roll. :-) */ |
488 return(0); | |
489 } |