Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 3490:42e33f89a167
Added the automated test to the Visual Studio 2008 project
author | Sam Lantinga <slouken@libsdl.org> |
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date | Wed, 25 Nov 2009 04:41:09 +0000 |
parents | 4e83cdb58134 |
children | 631173ffd68f |
rev | line source |
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0 | 1 /* |
2 SDL - Simple DirectMedia Layer | |
2859 | 3 Copyright (C) 1997-2009 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() */ |
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28 #include <errno.h> |
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29 #include <string.h> |
0 | 30 |
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31 #include "SDL_timer.h" |
0 | 32 #include "SDL_audio.h" |
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33 #include "../SDL_audiomem.h" |
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34 #include "../SDL_audio_c.h" |
0 | 35 #include "SDL_alsa_audio.h" |
36 | |
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37 #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
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38 #include "SDL_loadso.h" |
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39 #endif |
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40 |
0 | 41 /* The tag name used by ALSA audio */ |
42 #define DRIVER_NAME "alsa" | |
43 | |
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44 /* The default ALSA audio driver */ |
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45 #define DEFAULT_DEVICE "default" |
0 | 46 |
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47 static int (*ALSA_snd_pcm_open) |
2060 | 48 (snd_pcm_t **, const char *, snd_pcm_stream_t, int); |
49 static int (*ALSA_snd_pcm_close) (snd_pcm_t * pcm); | |
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50 static snd_pcm_sframes_t(*ALSA_snd_pcm_writei) |
2060 | 51 (snd_pcm_t *, const void *, snd_pcm_uframes_t); |
52 static int (*ALSA_snd_pcm_resume) (snd_pcm_t *); | |
53 static int (*ALSA_snd_pcm_prepare) (snd_pcm_t *); | |
54 static int (*ALSA_snd_pcm_drain) (snd_pcm_t *); | |
55 static const char *(*ALSA_snd_strerror) (int); | |
56 static size_t(*ALSA_snd_pcm_hw_params_sizeof) (void); | |
57 static size_t(*ALSA_snd_pcm_sw_params_sizeof) (void); | |
58 static int (*ALSA_snd_pcm_hw_params_any) (snd_pcm_t *, snd_pcm_hw_params_t *); | |
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59 static int (*ALSA_snd_pcm_hw_params_set_access) |
2060 | 60 (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t); |
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61 static int (*ALSA_snd_pcm_hw_params_set_format) |
2060 | 62 (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t); |
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63 static int (*ALSA_snd_pcm_hw_params_set_channels) |
2060 | 64 (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int); |
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65 static int (*ALSA_snd_pcm_hw_params_set_rate_near) |
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66 (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *); |
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67 static int (*ALSA_snd_pcm_hw_params_set_period_size_near) |
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68 (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t *, int *); |
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69 static int (*ALSA_snd_pcm_hw_params_get_channels) |
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70 (const snd_pcm_hw_params_t *, unsigned int *); |
2060 | 71 static snd_pcm_sframes_t(*ALSA_snd_pcm_hw_params_get_period_size) |
72 (const snd_pcm_hw_params_t *); | |
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73 static int (*ALSA_snd_pcm_hw_params_set_periods_near) |
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74 (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int *, int *); |
2060 | 75 static int (*ALSA_snd_pcm_hw_params_get_periods) (snd_pcm_hw_params_t *); |
76 static int (*ALSA_snd_pcm_hw_params) (snd_pcm_t *, snd_pcm_hw_params_t *); | |
77 static int (*ALSA_snd_pcm_sw_params_current) (snd_pcm_t *, | |
78 snd_pcm_sw_params_t *); | |
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79 static int (*ALSA_snd_pcm_sw_params_set_start_threshold) |
2060 | 80 (snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t); |
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81 static int (*ALSA_snd_pcm_sw_params_set_avail_min) |
2060 | 82 (snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t); |
83 static int (*ALSA_snd_pcm_sw_params) (snd_pcm_t *, snd_pcm_sw_params_t *); | |
84 static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int); | |
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85 #define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof |
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86 #define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof |
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87 |
0 | 88 |
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89 #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
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90 |
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91 static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC; |
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92 static void *alsa_handle = NULL; |
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93 |
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94 static int |
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95 load_alsa_sym(const char *fn, void **addr) |
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96 { |
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97 *addr = SDL_LoadFunction(alsa_handle, fn); |
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98 if (*addr == NULL) { |
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99 /* Don't call SDL_SetError(): SDL_LoadFunction already did. */ |
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100 return 0; |
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101 } |
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102 |
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103 return 1; |
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104 } |
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105 |
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106 /* cast funcs to char* first, to please GCC's strict aliasing rules. */ |
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107 #define SDL_ALSA_SYM(x) \ |
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108 if (!load_alsa_sym(#x, (void **) (char *) &ALSA_##x)) return -1 |
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109 #else |
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110 #define SDL_ALSA_SYM(x) ALSA_##x = x |
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111 #endif |
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112 |
2060 | 113 static int |
114 load_alsa_syms(void) | |
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115 { |
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116 SDL_ALSA_SYM(snd_pcm_open); |
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117 SDL_ALSA_SYM(snd_pcm_close); |
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118 SDL_ALSA_SYM(snd_pcm_writei); |
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119 SDL_ALSA_SYM(snd_pcm_resume); |
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120 SDL_ALSA_SYM(snd_pcm_prepare); |
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121 SDL_ALSA_SYM(snd_pcm_drain); |
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122 SDL_ALSA_SYM(snd_strerror); |
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123 SDL_ALSA_SYM(snd_pcm_hw_params_sizeof); |
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124 SDL_ALSA_SYM(snd_pcm_sw_params_sizeof); |
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125 SDL_ALSA_SYM(snd_pcm_hw_params_any); |
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126 SDL_ALSA_SYM(snd_pcm_hw_params_set_access); |
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127 SDL_ALSA_SYM(snd_pcm_hw_params_set_format); |
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128 SDL_ALSA_SYM(snd_pcm_hw_params_set_channels); |
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129 SDL_ALSA_SYM(snd_pcm_hw_params_get_channels); |
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130 SDL_ALSA_SYM(snd_pcm_hw_params_set_rate_near); |
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131 SDL_ALSA_SYM(snd_pcm_hw_params_set_period_size_near); |
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132 SDL_ALSA_SYM(snd_pcm_hw_params_get_period_size); |
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133 SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_near); |
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134 SDL_ALSA_SYM(snd_pcm_hw_params_get_periods); |
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135 SDL_ALSA_SYM(snd_pcm_hw_params); |
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136 SDL_ALSA_SYM(snd_pcm_sw_params_current); |
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137 SDL_ALSA_SYM(snd_pcm_sw_params_set_start_threshold); |
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138 SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min); |
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139 SDL_ALSA_SYM(snd_pcm_sw_params); |
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140 SDL_ALSA_SYM(snd_pcm_nonblock); |
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141 return 0; |
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142 } |
2060 | 143 |
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144 #undef SDL_ALSA_SYM |
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145 |
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146 #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
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147 |
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148 static void |
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149 UnloadALSALibrary(void) |
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150 { |
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151 if (alsa_handle != NULL) { |
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152 SDL_UnloadObject(alsa_handle); |
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153 alsa_handle = NULL; |
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154 } |
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155 } |
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156 |
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157 static int |
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158 LoadALSALibrary(void) |
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159 { |
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160 int retval = 0; |
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161 if (alsa_handle == NULL) { |
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162 alsa_handle = SDL_LoadObject(alsa_library); |
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163 if (alsa_handle == NULL) { |
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164 retval = -1; |
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165 /* Don't call SDL_SetError(): SDL_LoadObject already did. */ |
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166 } else { |
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167 retval = load_alsa_syms(); |
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168 if (retval < 0) { |
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169 UnloadALSALibrary(); |
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170 } |
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171 } |
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172 } |
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173 return retval; |
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174 } |
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175 |
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176 #else |
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177 |
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178 static void |
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179 UnloadALSALibrary(void) |
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180 { |
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181 } |
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182 |
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183 static int |
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184 LoadALSALibrary(void) |
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185 { |
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186 load_alsa_syms(); |
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187 return 0; |
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188 } |
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189 |
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190 #endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */ |
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191 |
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192 static const char * |
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193 get_audio_device(int channels) |
354
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194 { |
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195 const char *device; |
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196 |
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197 device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */ |
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198 if (device == NULL) { |
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199 if (channels == 6) |
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200 device = "surround51"; |
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201 else if (channels == 4) |
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202 device = "surround40"; |
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203 else |
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204 device = DEFAULT_DEVICE; |
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205 } |
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206 return device; |
0 | 207 } |
208 | |
209 | |
210 /* This function waits until it is possible to write a full sound buffer */ | |
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211 static void |
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212 ALSA_WaitDevice(_THIS) |
0 | 213 { |
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214 /* Check to see if the thread-parent process is still alive */ |
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215 { |
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216 static int cnt = 0; |
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217 /* Note that this only works with thread implementations |
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218 that use a different process id for each thread. |
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219 */ |
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220 /* Check every 10 loops */ |
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221 if (this->hidden->parent && (((++cnt) % 10) == 0)) { |
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223 this->enabled = 0; |
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224 } |
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225 } |
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226 } |
0 | 227 } |
228 | |
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229 |
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230 /* !!! FIXME: is there a channel swizzler in alsalib instead? */ |
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231 /* |
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232 * http://bugzilla.libsdl.org/show_bug.cgi?id=110 |
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233 * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE |
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234 * and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR" |
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235 */ |
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236 #define SWIZ6(T) \ |
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237 T *ptr = (T *) this->hidden->mixbuf; \ |
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238 const Uint32 count = (this->spec.samples / 6); \ |
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239 Uint32 i; \ |
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240 for (i = 0; i < count; i++, ptr += 6) { \ |
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241 T tmp; \ |
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242 tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \ |
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243 tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \ |
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244 } |
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245 |
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246 static __inline__ void |
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247 swizzle_alsa_channels_6_64bit(_THIS) |
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248 { |
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249 SWIZ6(Uint64); |
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250 } |
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251 |
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252 static __inline__ void |
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253 swizzle_alsa_channels_6_32bit(_THIS) |
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254 { |
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255 SWIZ6(Uint32); |
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256 } |
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257 |
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258 static __inline__ void |
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259 swizzle_alsa_channels_6_16bit(_THIS) |
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260 { |
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261 SWIZ6(Uint16); |
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262 } |
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263 |
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264 static __inline__ void |
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265 swizzle_alsa_channels_6_8bit(_THIS) |
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266 { |
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267 SWIZ6(Uint8); |
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268 } |
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269 |
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270 #undef SWIZ6 |
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271 |
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272 |
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273 /* |
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274 * Called right before feeding this->hidden->mixbuf to the hardware. Swizzle |
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275 * channels from Windows/Mac order to the format alsalib will want. |
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276 */ |
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277 static __inline__ void |
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278 swizzle_alsa_channels(_THIS) |
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279 { |
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280 if (this->spec.channels == 6) { |
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281 const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */ |
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282 if (fmtsize == 16) |
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283 swizzle_alsa_channels_6_16bit(this); |
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284 else if (fmtsize == 8) |
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285 swizzle_alsa_channels_6_8bit(this); |
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286 else if (fmtsize == 32) |
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287 swizzle_alsa_channels_6_32bit(this); |
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288 else if (fmtsize == 64) |
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289 swizzle_alsa_channels_6_64bit(this); |
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290 } |
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291 |
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292 /* !!! FIXME: update this for 7.1 if needed, later. */ |
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293 } |
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294 |
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295 |
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296 static void |
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297 ALSA_PlayDevice(_THIS) |
0 | 298 { |
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299 int status; |
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300 int sample_len; |
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301 signed short *sample_buf; |
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302 |
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303 swizzle_alsa_channels(this); |
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304 |
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305 sample_len = this->spec.samples; |
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306 sample_buf = (signed short *) this->hidden->mixbuf; |
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307 |
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308 while (sample_len > 0) { |
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309 status = ALSA_snd_pcm_writei(this->hidden->pcm_handle, |
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310 sample_buf, sample_len); |
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311 |
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312 if (status < 0) { |
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313 if (status == -EAGAIN) { |
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314 SDL_Delay(1); |
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315 continue; |
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316 } |
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317 if (status == -ESTRPIPE) { |
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318 do { |
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319 SDL_Delay(1); |
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320 status = ALSA_snd_pcm_resume(this->hidden->pcm_handle); |
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321 } while (status == -EAGAIN); |
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322 } |
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323 if (status < 0) { |
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324 status = ALSA_snd_pcm_prepare(this->hidden->pcm_handle); |
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325 } |
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326 if (status < 0) { |
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327 /* Hmm, not much we can do - abort */ |
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328 this->enabled = 0; |
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329 return; |
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330 } |
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331 continue; |
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332 } |
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333 sample_buf += status * this->spec.channels; |
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334 sample_len -= status; |
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335 } |
0 | 336 } |
337 | |
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338 static Uint8 * |
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339 ALSA_GetDeviceBuf(_THIS) |
0 | 340 { |
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341 return (this->hidden->mixbuf); |
0 | 342 } |
343 | |
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344 static void |
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345 ALSA_CloseDevice(_THIS) |
0 | 346 { |
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347 if (this->hidden != NULL) { |
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348 if (this->hidden->mixbuf != NULL) { |
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349 SDL_FreeAudioMem(this->hidden->mixbuf); |
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350 this->hidden->mixbuf = NULL; |
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351 } |
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352 if (this->hidden->pcm_handle) { |
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353 ALSA_snd_pcm_drain(this->hidden->pcm_handle); |
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354 ALSA_snd_pcm_close(this->hidden->pcm_handle); |
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355 this->hidden->pcm_handle = NULL; |
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356 } |
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357 SDL_free(this->hidden); |
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358 this->hidden = NULL; |
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359 } |
0 | 360 } |
361 | |
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362 static int |
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363 ALSA_OpenDevice(_THIS, const char *devname, int iscapture) |
0 | 364 { |
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365 int status = 0; |
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366 snd_pcm_t *pcm_handle = NULL; |
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367 snd_pcm_hw_params_t *hwparams = NULL; |
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368 snd_pcm_sw_params_t *swparams = NULL; |
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369 snd_pcm_format_t format = 0; |
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370 snd_pcm_uframes_t frames = 0; |
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371 SDL_AudioFormat test_format = 0; |
3362
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Merged r4990:4991 from branches/SDL-1.2: ALSA 1.0 API and dlvsym() removal.
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372 unsigned int rate = 0; |
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373 unsigned int periods = 0; |
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374 unsigned int channels = 0; |
2049
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375 |
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376 /* Initialize all variables that we clean on shutdown */ |
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377 this->hidden = (struct SDL_PrivateAudioData *) |
2060 | 378 SDL_malloc((sizeof *this->hidden)); |
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379 if (this->hidden == NULL) { |
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380 SDL_OutOfMemory(); |
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381 return 0; |
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382 } |
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383 SDL_memset(this->hidden, 0, (sizeof *this->hidden)); |
0 | 384 |
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385 /* Open the audio device */ |
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386 /* Name of device should depend on # channels in spec */ |
2049
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387 status = ALSA_snd_pcm_open(&pcm_handle, |
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388 get_audio_device(this->spec.channels), |
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389 SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
942
41a59de7f2ed
Here are patches for SDL12 and SDL_mixer for 4 or 6 channel
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390 |
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391 if (status < 0) { |
2049
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392 ALSA_CloseDevice(this); |
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393 SDL_SetError("ALSA: Couldn't open audio device: %s", |
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394 ALSA_snd_strerror(status)); |
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395 return 0; |
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396 } |
354
30935e76acb5
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397 |
2049
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398 this->hidden->pcm_handle = pcm_handle; |
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399 |
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400 /* Figure out what the hardware is capable of */ |
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401 snd_pcm_hw_params_alloca(&hwparams); |
2049
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402 status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams); |
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403 if (status < 0) { |
2049
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404 ALSA_CloseDevice(this); |
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405 SDL_SetError("ALSA: Couldn't get hardware config: %s", |
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406 ALSA_snd_strerror(status)); |
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407 return 0; |
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408 } |
354
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Updated ALSA audio support for ALSA 0.9
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409 |
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410 /* SDL only uses interleaved sample output */ |
2049
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411 status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams, |
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412 SND_PCM_ACCESS_RW_INTERLEAVED); |
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413 if (status < 0) { |
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414 ALSA_CloseDevice(this); |
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415 SDL_SetError("ALSA: Couldn't set interleaved access: %s", |
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416 ALSA_snd_strerror(status)); |
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417 return 0; |
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418 } |
0 | 419 |
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420 /* Try for a closest match on audio format */ |
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421 status = -1; |
2049
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422 for (test_format = SDL_FirstAudioFormat(this->spec.format); |
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423 test_format && (status < 0);) { |
2060 | 424 status = 0; /* if we can't support a format, it'll become -1. */ |
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425 switch (test_format) { |
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426 case AUDIO_U8: |
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427 format = SND_PCM_FORMAT_U8; |
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428 break; |
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429 case AUDIO_S8: |
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430 format = SND_PCM_FORMAT_S8; |
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431 break; |
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432 case AUDIO_S16LSB: |
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433 format = SND_PCM_FORMAT_S16_LE; |
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434 break; |
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435 case AUDIO_S16MSB: |
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436 format = SND_PCM_FORMAT_S16_BE; |
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437 break; |
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438 case AUDIO_U16LSB: |
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439 format = SND_PCM_FORMAT_U16_LE; |
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440 break; |
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441 case AUDIO_U16MSB: |
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442 format = SND_PCM_FORMAT_U16_BE; |
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443 break; |
1995
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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parents:
1982
diff
changeset
|
444 case AUDIO_S32LSB: |
2010
39897da56f63
Whoops, wrong tokens for int32 support in ALSA driver (specified unsigned
Ryan C. Gordon <icculus@icculus.org>
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2009
diff
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|
445 format = SND_PCM_FORMAT_S32_LE; |
1995
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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diff
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|
446 break; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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|
447 case AUDIO_S32MSB: |
2010
39897da56f63
Whoops, wrong tokens for int32 support in ALSA driver (specified unsigned
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2009
diff
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448 format = SND_PCM_FORMAT_S32_BE; |
1995
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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diff
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449 break; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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diff
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|
450 case AUDIO_F32LSB: |
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ALSA backend can handle int32 and float32 data directly.
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451 format = SND_PCM_FORMAT_FLOAT_LE; |
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ALSA backend can handle int32 and float32 data directly.
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diff
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|
452 break; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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diff
changeset
|
453 case AUDIO_F32MSB: |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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454 format = SND_PCM_FORMAT_FLOAT_BE; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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|
455 break; |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
456 default: |
2009
f2058fb367e4
ALSA was testing if (format) was set to zero as an error condition, but
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diff
changeset
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457 status = -1; |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
458 break; |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
459 } |
2009
f2058fb367e4
ALSA was testing if (format) was set to zero as an error condition, but
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diff
changeset
|
460 if (status >= 0) { |
2049
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461 status = ALSA_snd_pcm_hw_params_set_format(pcm_handle, |
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|
462 hwparams, format); |
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463 } |
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464 if (status < 0) { |
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465 test_format = SDL_NextAudioFormat(); |
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466 } |
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467 } |
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|
468 if (status < 0) { |
2049
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|
469 ALSA_CloseDevice(this); |
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|
470 SDL_SetError("ALSA: Couldn't find any hardware audio formats"); |
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|
471 return 0; |
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|
472 } |
2049
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|
473 this->spec.format = test_format; |
0 | 474 |
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|
475 /* Set the number of channels */ |
2049
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|
476 status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams, |
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477 this->spec.channels); |
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478 channels = this->spec.channels; |
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479 if (status < 0) { |
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480 status = ALSA_snd_pcm_hw_params_get_channels(hwparams, &channels); |
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481 if (status < 0) { |
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482 ALSA_CloseDevice(this); |
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483 SDL_SetError("ALSA: Couldn't set audio channels"); |
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484 return 0; |
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485 } |
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486 this->spec.channels = channels; |
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487 } |
0 | 488 |
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489 /* Set the audio rate */ |
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490 rate = this->spec.freq; |
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491 status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, |
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492 &rate, NULL); |
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493 if (status < 0) { |
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494 ALSA_CloseDevice(this); |
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495 SDL_SetError("ALSA: Couldn't set audio frequency: %s", |
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496 ALSA_snd_strerror(status)); |
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497 return 0; |
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498 } |
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499 this->spec.freq = rate; |
0 | 500 |
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501 /* Set the buffer size, in samples */ |
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502 frames = this->spec.samples; |
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503 status = ALSA_snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, |
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504 &frames, NULL); |
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505 if ( status < 0 ) { |
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506 ALSA_CloseDevice(this); |
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507 SDL_SetError("ALSA: Couldn't set audio frequency: %s", |
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508 ALSA_snd_strerror(status)); |
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509 return(-1); |
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510 } |
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511 this->spec.samples = frames; |
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512 |
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513 periods = 2; |
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514 ALSA_snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, |
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515 &periods, NULL); |
354
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516 |
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517 /* "set" the hardware with the desired parameters */ |
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518 status = ALSA_snd_pcm_hw_params(pcm_handle, hwparams); |
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519 if (status < 0) { |
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520 ALSA_CloseDevice(this); |
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521 SDL_SetError("ALSA: Couldn't set hardware audio parameters: %s", |
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522 ALSA_snd_strerror(status)); |
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523 return 0; |
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524 } |
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525 #if AUDIO_DEBUG |
2060 | 526 { |
527 snd_pcm_sframes_t bufsize; | |
528 int fragments; | |
529 bufsize = ALSA_snd_pcm_hw_params_get_period_size(hwparams); | |
530 fragments = ALSA_snd_pcm_hw_params_get_periods(hwparams); | |
531 fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, | |
532 fragments); | |
533 } | |
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534 #endif |
1552 | 535 |
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536 /* Set the software parameters */ |
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537 snd_pcm_sw_params_alloca(&swparams); |
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538 status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams); |
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539 if (status < 0) { |
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540 ALSA_CloseDevice(this); |
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541 SDL_SetError("ALSA: Couldn't get software config: %s", |
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542 ALSA_snd_strerror(status)); |
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543 return 0; |
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544 } |
2060 | 545 status = |
546 ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 0); | |
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547 if (status < 0) { |
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548 ALSA_CloseDevice(this); |
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549 SDL_SetError("ALSA: Couldn't set start threshold: %s", |
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550 ALSA_snd_strerror(status)); |
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551 return 0; |
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552 } |
2060 | 553 status = |
554 ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, frames); | |
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555 if (status < 0) { |
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556 ALSA_CloseDevice(this); |
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557 SDL_SetError("Couldn't set avail min: %s", ALSA_snd_strerror(status)); |
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558 return 0; |
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559 } |
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560 status = ALSA_snd_pcm_sw_params(pcm_handle, swparams); |
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561 if (status < 0) { |
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562 ALSA_CloseDevice(this); |
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563 SDL_SetError("Couldn't set software audio parameters: %s", |
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564 ALSA_snd_strerror(status)); |
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565 return 0; |
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566 } |
0 | 567 |
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568 /* Calculate the final parameters for this audio specification */ |
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569 SDL_CalculateAudioSpec(&this->spec); |
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570 |
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571 /* Allocate mixing buffer */ |
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572 this->hidden->mixlen = this->spec.size; |
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573 this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen); |
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574 if (this->hidden->mixbuf == NULL) { |
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575 ALSA_CloseDevice(this); |
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576 SDL_OutOfMemory(); |
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577 return 0; |
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578 } |
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579 SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size); |
0 | 580 |
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581 /* Get the parent process id (we're the parent of the audio thread) */ |
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582 this->hidden->parent = getpid(); |
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583 |
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584 /* Switch to blocking mode for playback */ |
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585 ALSA_snd_pcm_nonblock(pcm_handle, 0); |
0 | 586 |
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587 /* We're ready to rock and roll. :-) */ |
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588 return 1; |
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589 } |
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590 |
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591 static void |
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592 ALSA_Deinitialize(void) |
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593 { |
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594 UnloadALSALibrary(); |
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595 } |
0 | 596 |
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597 static int |
2060 | 598 ALSA_Init(SDL_AudioDriverImpl * impl) |
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599 { |
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600 if (LoadALSALibrary() < 0) { |
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601 return 0; |
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602 } |
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603 |
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604 /* Set the function pointers */ |
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605 impl->OpenDevice = ALSA_OpenDevice; |
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606 impl->WaitDevice = ALSA_WaitDevice; |
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607 impl->GetDeviceBuf = ALSA_GetDeviceBuf; |
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608 impl->PlayDevice = ALSA_PlayDevice; |
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609 impl->CloseDevice = ALSA_CloseDevice; |
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610 impl->Deinitialize = ALSA_Deinitialize; |
2060 | 611 impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: Add device enum! */ |
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612 |
2942 | 613 return 1; /* !!! FIXME: return 2 once device enum is implemented. */ |
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614 } |
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615 |
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616 |
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617 AudioBootStrap ALSA_bootstrap = { |
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618 DRIVER_NAME, "ALSA 0.9 PCM audio", ALSA_Init, 0 |
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619 }; |
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620 |
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621 /* vi: set ts=4 sw=4 expandtab: */ |