Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 4348:b312352d8c8d SDL-1.2
Enable the ALSA debug code via environment variable
author | Sam Lantinga <slouken@libsdl.org> |
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date | Sat, 17 Oct 2009 07:04:58 +0000 |
parents | 38f22ed3a433 |
children | 471dac4b41dd |
rev | line source |
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0 | 1 /* |
2 SDL - Simple DirectMedia Layer | |
4159 | 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() */ |
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 "SDL_name.h" |
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37 #include "SDL_loadso.h" |
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38 #else |
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39 #define SDL_NAME(X) X |
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40 #endif |
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41 |
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42 |
0 | 43 /* The tag name used by ALSA audio */ |
44 #define DRIVER_NAME "alsa" | |
45 | |
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46 /*#define DEBUG_PERIOD_SIZE*/ |
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47 |
0 | 48 /* Audio driver functions */ |
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49 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec); |
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50 static void ALSA_WaitAudio(_THIS); |
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51 static void ALSA_PlayAudio(_THIS); |
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52 static Uint8 *ALSA_GetAudioBuf(_THIS); |
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53 static void ALSA_CloseAudio(_THIS); |
0 | 54 |
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55 #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
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56 |
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57 static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC; |
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58 static void *alsa_handle = NULL; |
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59 static int alsa_loaded = 0; |
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60 |
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61 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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62 static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm); |
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63 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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64 static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm); |
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65 static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm); |
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66 static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm); |
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67 static const char *(*SDL_NAME(snd_strerror))(int errnum); |
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68 static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void); |
1552 | 69 static size_t (*SDL_NAME(snd_pcm_sw_params_sizeof))(void); |
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70 static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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71 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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72 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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73 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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74 static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params, unsigned int *val); |
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75 static 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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76 static int (*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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77 static int (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir); |
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78 static 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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79 static int (*SDL_NAME(snd_pcm_hw_params_get_periods))(const snd_pcm_hw_params_t *params, unsigned int *val, int *dir); |
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80 static int (*SDL_NAME(snd_pcm_hw_params_set_buffer_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val); |
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81 static int (*SDL_NAME(snd_pcm_hw_params_get_buffer_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val); |
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82 static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
1552 | 83 /* |
84 */ | |
85 static int (*SDL_NAME(snd_pcm_sw_params_current))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams); | |
86 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); | |
87 static int (*SDL_NAME(snd_pcm_sw_params))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params); | |
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88 static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock); |
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89 #define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof) |
1552 | 90 #define snd_pcm_sw_params_sizeof SDL_NAME(snd_pcm_sw_params_sizeof) |
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91 |
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92 /* cast funcs to char* first, to please GCC's strict aliasing rules. */ |
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93 static struct { |
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94 const char *name; |
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95 void **func; |
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96 } alsa_functions[] = { |
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97 { "snd_pcm_open", (void**)(char*)&SDL_NAME(snd_pcm_open) }, |
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98 { "snd_pcm_close", (void**)(char*)&SDL_NAME(snd_pcm_close) }, |
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99 { "snd_pcm_writei", (void**)(char*)&SDL_NAME(snd_pcm_writei) }, |
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100 { "snd_pcm_resume", (void**)(char*)&SDL_NAME(snd_pcm_resume) }, |
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101 { "snd_pcm_prepare", (void**)(char*)&SDL_NAME(snd_pcm_prepare) }, |
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102 { "snd_pcm_drain", (void**)(char*)&SDL_NAME(snd_pcm_drain) }, |
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103 { "snd_strerror", (void**)(char*)&SDL_NAME(snd_strerror) }, |
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104 { "snd_pcm_hw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof) }, |
1552 | 105 { "snd_pcm_sw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_sizeof) }, |
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106 { "snd_pcm_hw_params_any", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_any) }, |
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107 { "snd_pcm_hw_params_set_access", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access) }, |
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108 { "snd_pcm_hw_params_set_format", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format) }, |
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109 { "snd_pcm_hw_params_set_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels) }, |
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110 { "snd_pcm_hw_params_get_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels) }, |
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111 { "snd_pcm_hw_params_set_rate_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near) }, |
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112 { "snd_pcm_hw_params_set_period_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) }, |
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113 { "snd_pcm_hw_params_get_period_size", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_period_size) }, |
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114 { "snd_pcm_hw_params_set_periods_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near) }, |
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115 { "snd_pcm_hw_params_get_periods", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_periods) }, |
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116 { "snd_pcm_hw_params_set_buffer_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_buffer_size_near) }, |
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117 { "snd_pcm_hw_params_get_buffer_size", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_buffer_size) }, |
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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", (void**)(char*)&SDL_NAME(snd_pcm_sw_params) }, | |
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122 { "snd_pcm_nonblock", (void**)(char*)&SDL_NAME(snd_pcm_nonblock) }, |
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123 }; |
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124 |
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125 static void UnloadALSALibrary(void) { |
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126 if (alsa_loaded) { |
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127 SDL_UnloadObject(alsa_handle); |
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128 alsa_handle = NULL; |
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129 alsa_loaded = 0; |
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130 } |
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131 } |
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132 |
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133 static int LoadALSALibrary(void) { |
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134 int i, retval = -1; |
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135 |
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136 alsa_handle = SDL_LoadObject(alsa_library); |
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137 if (alsa_handle) { |
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138 alsa_loaded = 1; |
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139 retval = 0; |
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140 for (i = 0; i < SDL_arraysize(alsa_functions); i++) { |
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141 *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name); |
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142 if (!*alsa_functions[i].func) { |
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143 retval = -1; |
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144 UnloadALSALibrary(); |
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145 break; |
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146 } |
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147 } |
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148 } |
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149 return retval; |
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150 } |
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151 |
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152 #else |
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153 |
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154 static void UnloadALSALibrary(void) { |
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155 return; |
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156 } |
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157 |
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158 static int LoadALSALibrary(void) { |
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159 return 0; |
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160 } |
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161 |
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162 #endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */ |
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163 |
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164 static const char *get_audio_device(int channels) |
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165 { |
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166 const char *device; |
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167 |
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168 device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */ |
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169 if ( device == NULL ) { |
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170 switch (channels) { |
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171 case 6: |
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172 device = "plug:surround51"; |
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173 break; |
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174 case 4: |
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175 device = "plug:surround40"; |
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176 break; |
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177 default: |
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178 device = "default"; |
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179 break; |
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180 } |
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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 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 |
354
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313 static void ALSA_PlayAudio(_THIS) |
0 | 314 { |
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315 int status; |
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316 snd_pcm_uframes_t frames_left; |
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317 const Uint8 *sample_buf = (const Uint8 *) mixbuf; |
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318 const int frame_size = (((int) (this->spec.format & 0xFF)) / 8) * this->spec.channels; |
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319 |
1878
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320 swizzle_alsa_channels(this); |
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321 |
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322 frames_left = ((snd_pcm_uframes_t) this->spec.samples); |
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323 |
4337 | 324 while ( frames_left > 0 && this->enabled ) { |
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325 status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, frames_left); |
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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 } |
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347 sample_buf += status * frame_size; |
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348 frames_left -= 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 unsigned int rate; |
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378 unsigned int periods; |
4333
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379 unsigned int channels; |
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380 Uint16 test_format; |
0 | 381 |
382 /* Open the audio device */ | |
942
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383 /* Name of device should depend on # channels in spec */ |
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384 status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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385 |
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386 if ( status < 0 ) { |
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387 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); |
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388 return(-1); |
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389 } |
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390 |
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391 /* Figure out what the hardware is capable of */ |
1552 | 392 snd_pcm_hw_params_alloca(&hwparams); |
393 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams); | |
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394 if ( status < 0 ) { |
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395 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); |
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396 ALSA_CloseAudio(this); |
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397 return(-1); |
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398 } |
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399 |
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400 /* SDL only uses interleaved sample output */ |
1552 | 401 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); |
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402 if ( status < 0 ) { |
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403 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); |
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404 ALSA_CloseAudio(this); |
0 | 405 return(-1); |
406 } | |
407 | |
408 /* Try for a closest match on audio format */ | |
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409 status = -1; |
0 | 410 for ( test_format = SDL_FirstAudioFormat(spec->format); |
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411 test_format && (status < 0); ) { |
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412 switch ( test_format ) { |
0 | 413 case AUDIO_U8: |
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414 format = SND_PCM_FORMAT_U8; |
0 | 415 break; |
416 case AUDIO_S8: | |
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417 format = SND_PCM_FORMAT_S8; |
0 | 418 break; |
419 case AUDIO_S16LSB: | |
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420 format = SND_PCM_FORMAT_S16_LE; |
0 | 421 break; |
422 case AUDIO_S16MSB: | |
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423 format = SND_PCM_FORMAT_S16_BE; |
0 | 424 break; |
425 case AUDIO_U16LSB: | |
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426 format = SND_PCM_FORMAT_U16_LE; |
0 | 427 break; |
428 case AUDIO_U16MSB: | |
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429 format = SND_PCM_FORMAT_U16_BE; |
0 | 430 break; |
431 default: | |
354
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432 format = 0; |
0 | 433 break; |
434 } | |
354
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435 if ( format != 0 ) { |
1552 | 436 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format); |
354
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437 } |
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438 if ( status < 0 ) { |
0 | 439 test_format = SDL_NextAudioFormat(); |
440 } | |
441 } | |
354
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442 if ( status < 0 ) { |
0 | 443 SDL_SetError("Couldn't find any hardware audio formats"); |
354
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444 ALSA_CloseAudio(this); |
0 | 445 return(-1); |
446 } | |
447 spec->format = test_format; | |
448 | |
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449 /* Set the number of channels */ |
1552 | 450 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels); |
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451 channels = spec->channels; |
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452 if ( status < 0 ) { |
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453 status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams, &channels); |
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454 if ( status < 0 ) { |
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455 SDL_SetError("Couldn't set audio channels"); |
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456 ALSA_CloseAudio(this); |
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457 return(-1); |
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458 } |
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459 spec->channels = channels; |
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460 } |
0 | 461 |
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462 /* Set the audio rate */ |
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463 rate = spec->freq; |
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464 |
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465 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL); |
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466 if ( status < 0 ) { |
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467 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); |
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468 ALSA_CloseAudio(this); |
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469 return(-1); |
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470 } |
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471 spec->freq = rate; |
0 | 472 |
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473 /* Set the buffer size, in samples */ |
4347 | 474 if (getenv("SDL_AUDIO_ALSA_SET_PERIOD_SIZE")) { |
475 frames = spec->samples; | |
476 status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL); | |
477 if ( status < 0 ) { | |
478 SDL_SetError("Couldn't set period size: %s", SDL_NAME(snd_strerror)(status)); | |
479 ALSA_CloseAudio(this); | |
480 return(-1); | |
481 } | |
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482 |
4347 | 483 spec->samples = frames; |
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484 |
4347 | 485 periods = 2; |
486 status = SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL); | |
487 if ( status < 0 ) { | |
488 SDL_SetError("Couldn't set period count: %s", SDL_NAME(snd_strerror)(status)); | |
489 ALSA_CloseAudio(this); | |
490 return(-1); | |
491 } | |
492 } else { | |
493 frames = spec->samples * 2; | |
494 status = SDL_NAME(snd_pcm_hw_params_set_buffer_size_near)(pcm_handle, hwparams, &frames); | |
495 if ( status < 0 ) { | |
496 SDL_SetError("Couldn't set buffer size: %s", SDL_NAME(snd_strerror)(status)); | |
497 ALSA_CloseAudio(this); | |
498 return(-1); | |
499 } | |
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500 } |
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501 |
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502 /* "set" the hardware with the desired parameters */ |
1552 | 503 status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams); |
504 if ( status < 0 ) { | |
505 SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status)); | |
506 ALSA_CloseAudio(this); | |
507 return(-1); | |
508 } | |
509 | |
4348
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510 /* This is useful for debugging */ |
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511 if (getenv("SDL_AUDIO_ALSA_DEBUG_PERIOD_SIZE")) { |
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512 snd_pcm_uframes_t bufsize; |
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513 snd_pcm_sframes_t persize; |
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514 unsigned int periods; int dir; |
1552 | 515 |
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516 SDL_NAME(snd_pcm_hw_params_get_buffer_size)(hwparams, &bufsize); |
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517 SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams, &persize, &dir); |
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518 SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams, &periods, &dir); |
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519 |
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520 fprintf(stderr, "ALSA: period size = %ld, periods = %u, buffer size = %lu\n", persize, periods, bufsize); |
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521 } |
1552 | 522 |
523 /* Set the software parameters */ | |
524 snd_pcm_sw_params_alloca(&swparams); | |
525 status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams); | |
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526 if ( status < 0 ) { |
1552 | 527 SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status)); |
528 ALSA_CloseAudio(this); | |
529 return(-1); | |
530 } | |
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531 status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 1); |
1552 | 532 if ( status < 0 ) { |
533 SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status)); | |
534 ALSA_CloseAudio(this); | |
535 return(-1); | |
536 } | |
537 status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams); | |
538 if ( status < 0 ) { | |
539 SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status)); | |
354
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540 ALSA_CloseAudio(this); |
0 | 541 return(-1); |
542 } | |
543 | |
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544 /* Calculate the final parameters for this audio specification */ |
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545 SDL_CalculateAudioSpec(spec); |
0 | 546 |
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547 /* Allocate mixing buffer */ |
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548 mixlen = spec->size; |
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549 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); |
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550 if ( mixbuf == NULL ) { |
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551 ALSA_CloseAudio(this); |
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552 return(-1); |
0 | 553 } |
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554 SDL_memset(mixbuf, spec->silence, spec->size); |
0 | 555 |
556 /* Get the parent process id (we're the parent of the audio thread) */ | |
557 parent = getpid(); | |
558 | |
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559 /* Switch to blocking mode for playback */ |
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560 SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); |
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561 |
0 | 562 /* We're ready to rock and roll. :-) */ |
563 return(0); | |
564 } |