Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 2247:93994f65c74c
Removed hermes since it's LGPL and not compatible with a commercial license.
Prepping for using MMX and SSE intrinsics instead of inline assembly.
.. except for memcpy equivalents which only get faster if they can
exploit the parallelism of loading into multiple SIMD registers. :)
author | Sam Lantinga <slouken@libsdl.org> |
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date | Wed, 15 Aug 2007 08:21:10 +0000 |
parents | 866052b01ee5 |
children | e1da92da346c |
rev | line source |
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0 | 1 /* |
2 SDL - Simple DirectMedia Layer | |
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3 Copyright (C) 1997-2004 Sam Lantinga |
0 | 4 |
5 This library is free software; you can redistribute it and/or | |
6 modify it under the terms of the GNU Library General Public | |
7 License as published by the Free Software Foundation; either | |
8 version 2 of the License, or (at your option) any later version. | |
9 | |
10 This library is distributed in the hope that it will be useful, | |
11 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 Library General Public License for more details. | |
14 | |
15 You should have received a copy of the GNU Library General Public | |
16 License along with this library; if not, write to the Free | |
17 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
18 | |
19 Sam Lantinga | |
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20 slouken@libsdl.org |
0 | 21 */ |
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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 <dlfcn.h> |
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29 #include <errno.h> |
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30 #include <string.h> |
0 | 31 |
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32 #include "SDL_timer.h" |
0 | 33 #include "SDL_audio.h" |
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34 #include "../SDL_audiomem.h" |
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35 #include "../SDL_audio_c.h" |
0 | 36 #include "SDL_alsa_audio.h" |
37 | |
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38 |
0 | 39 /* The tag name used by ALSA audio */ |
40 #define DRIVER_NAME "alsa" | |
41 | |
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42 /* The default ALSA audio driver */ |
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43 #define DEFAULT_DEVICE "default" |
0 | 44 |
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45 static int (*ALSA_snd_pcm_open) |
2060 | 46 (snd_pcm_t **, const char *, snd_pcm_stream_t, int); |
47 static int (*ALSA_snd_pcm_close) (snd_pcm_t * pcm); | |
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48 static snd_pcm_sframes_t(*ALSA_snd_pcm_writei) |
2060 | 49 (snd_pcm_t *, const void *, snd_pcm_uframes_t); |
50 static int (*ALSA_snd_pcm_resume) (snd_pcm_t *); | |
51 static int (*ALSA_snd_pcm_prepare) (snd_pcm_t *); | |
52 static int (*ALSA_snd_pcm_drain) (snd_pcm_t *); | |
53 static const char *(*ALSA_snd_strerror) (int); | |
54 static size_t(*ALSA_snd_pcm_hw_params_sizeof) (void); | |
55 static size_t(*ALSA_snd_pcm_sw_params_sizeof) (void); | |
56 static int (*ALSA_snd_pcm_hw_params_any) (snd_pcm_t *, snd_pcm_hw_params_t *); | |
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57 static int (*ALSA_snd_pcm_hw_params_set_access) |
2060 | 58 (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_access_t); |
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59 static int (*ALSA_snd_pcm_hw_params_set_format) |
2060 | 60 (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_format_t); |
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61 static int (*ALSA_snd_pcm_hw_params_set_channels) |
2060 | 62 (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int); |
63 static int (*ALSA_snd_pcm_hw_params_get_channels) (const snd_pcm_hw_params_t | |
64 *); | |
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65 static unsigned int (*ALSA_snd_pcm_hw_params_set_rate_near) |
2060 | 66 (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int, int *); |
67 static snd_pcm_uframes_t(*ALSA_snd_pcm_hw_params_set_period_size_near) | |
68 (snd_pcm_t *, snd_pcm_hw_params_t *, snd_pcm_uframes_t, int *); | |
69 static snd_pcm_sframes_t(*ALSA_snd_pcm_hw_params_get_period_size) | |
70 (const snd_pcm_hw_params_t *); | |
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71 static unsigned int (*ALSA_snd_pcm_hw_params_set_periods_near) |
2060 | 72 (snd_pcm_t *, snd_pcm_hw_params_t *, unsigned int, int *); |
73 static int (*ALSA_snd_pcm_hw_params_get_periods) (snd_pcm_hw_params_t *); | |
74 static int (*ALSA_snd_pcm_hw_params) (snd_pcm_t *, snd_pcm_hw_params_t *); | |
75 static int (*ALSA_snd_pcm_sw_params_current) (snd_pcm_t *, | |
76 snd_pcm_sw_params_t *); | |
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77 static int (*ALSA_snd_pcm_sw_params_set_start_threshold) |
2060 | 78 (snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t); |
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79 static int (*ALSA_snd_pcm_sw_params_set_avail_min) |
2060 | 80 (snd_pcm_t *, snd_pcm_sw_params_t *, snd_pcm_uframes_t); |
81 static int (*ALSA_snd_pcm_sw_params) (snd_pcm_t *, snd_pcm_sw_params_t *); | |
82 static int (*ALSA_snd_pcm_nonblock) (snd_pcm_t *, int); | |
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83 #define snd_pcm_hw_params_sizeof ALSA_snd_pcm_hw_params_sizeof |
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84 #define snd_pcm_sw_params_sizeof ALSA_snd_pcm_sw_params_sizeof |
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85 |
0 | 86 |
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87 #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
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88 |
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89 static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC; |
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90 static void *alsa_handle = NULL; |
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91 |
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92 static int |
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93 load_alsa_sym(const char *fn, void **addr) |
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94 { |
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95 /* |
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96 * !!! FIXME: |
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97 * Eventually, this will deal with fallbacks, version changes, and |
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98 * missing symbols we can workaround. But for now, it doesn't. |
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99 */ |
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100 |
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101 #if HAVE_DLVSYM |
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102 *addr = dlvsym(alsa_handle, fn, "ALSA_0.9"); |
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103 if (*addr == NULL) |
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104 #endif |
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105 { |
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106 *addr = dlsym(alsa_handle, fn); |
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107 if (*addr == NULL) { |
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108 SDL_SetError("dlsym('%s') failed: %s", fn, strerror(errno)); |
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109 return 0; |
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110 } |
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111 } |
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112 |
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113 return 1; |
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114 } |
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115 |
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116 /* cast funcs to char* first, to please GCC's strict aliasing rules. */ |
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117 #define SDL_ALSA_SYM(x) \ |
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118 if (!load_alsa_sym(#x, (void **) (char *) &ALSA_##x)) return -1 |
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119 #else |
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120 #define SDL_ALSA_SYM(x) ALSA_##x = x |
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121 #endif |
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122 |
2060 | 123 static int |
124 load_alsa_syms(void) | |
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125 { |
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126 SDL_ALSA_SYM(snd_pcm_open); |
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127 SDL_ALSA_SYM(snd_pcm_close); |
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128 SDL_ALSA_SYM(snd_pcm_writei); |
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129 SDL_ALSA_SYM(snd_pcm_resume); |
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130 SDL_ALSA_SYM(snd_pcm_prepare); |
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131 SDL_ALSA_SYM(snd_pcm_drain); |
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132 SDL_ALSA_SYM(snd_strerror); |
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133 SDL_ALSA_SYM(snd_pcm_hw_params_sizeof); |
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134 SDL_ALSA_SYM(snd_pcm_sw_params_sizeof); |
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135 SDL_ALSA_SYM(snd_pcm_hw_params_any); |
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136 SDL_ALSA_SYM(snd_pcm_hw_params_set_access); |
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137 SDL_ALSA_SYM(snd_pcm_hw_params_set_format); |
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138 SDL_ALSA_SYM(snd_pcm_hw_params_set_channels); |
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139 SDL_ALSA_SYM(snd_pcm_hw_params_get_channels); |
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140 SDL_ALSA_SYM(snd_pcm_hw_params_set_rate_near); |
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141 SDL_ALSA_SYM(snd_pcm_hw_params_set_period_size_near); |
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142 SDL_ALSA_SYM(snd_pcm_hw_params_get_period_size); |
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143 SDL_ALSA_SYM(snd_pcm_hw_params_set_periods_near); |
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144 SDL_ALSA_SYM(snd_pcm_hw_params_get_periods); |
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145 SDL_ALSA_SYM(snd_pcm_hw_params); |
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146 SDL_ALSA_SYM(snd_pcm_sw_params_current); |
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147 SDL_ALSA_SYM(snd_pcm_sw_params_set_start_threshold); |
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148 SDL_ALSA_SYM(snd_pcm_sw_params_set_avail_min); |
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149 SDL_ALSA_SYM(snd_pcm_sw_params); |
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150 SDL_ALSA_SYM(snd_pcm_nonblock); |
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151 return 0; |
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152 } |
2060 | 153 |
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154 #undef SDL_ALSA_SYM |
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155 |
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156 #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
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157 |
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158 static void |
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159 UnloadALSALibrary(void) |
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160 { |
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161 if (alsa_handle != NULL) { |
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162 dlclose(alsa_handle); |
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163 alsa_handle = NULL; |
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164 } |
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165 } |
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166 |
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167 static int |
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168 LoadALSALibrary(void) |
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169 { |
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170 int retval = 0; |
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171 if (alsa_handle == NULL) { |
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172 alsa_handle = dlopen(alsa_library, RTLD_NOW); |
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173 if (alsa_handle == NULL) { |
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174 retval = -1; |
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175 SDL_SetError("ALSA: dlopen('%s') failed: %s\n", |
2060 | 176 alsa_library, strerror(errno)); |
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177 } else { |
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178 retval = load_alsa_syms(); |
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179 if (retval < 0) { |
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180 UnloadALSALibrary(); |
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181 } |
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182 } |
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183 } |
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184 return retval; |
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185 } |
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186 |
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187 #else |
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188 |
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189 static void |
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190 UnloadALSALibrary(void) |
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191 { |
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192 } |
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193 |
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194 static int |
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195 LoadALSALibrary(void) |
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196 { |
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197 load_alsa_syms(); |
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198 return 0; |
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199 } |
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200 |
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201 #endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */ |
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202 |
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203 static const char * |
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204 get_audio_device(int channels) |
354
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205 { |
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206 const char *device; |
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207 |
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208 device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */ |
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209 if (device == NULL) { |
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210 if (channels == 6) |
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211 device = "surround51"; |
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212 else if (channels == 4) |
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213 device = "surround40"; |
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214 else |
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215 device = DEFAULT_DEVICE; |
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216 } |
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217 return device; |
0 | 218 } |
219 | |
220 | |
221 /* This function waits until it is possible to write a full sound buffer */ | |
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222 static void |
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223 ALSA_WaitDevice(_THIS) |
0 | 224 { |
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225 /* Check to see if the thread-parent process is still alive */ |
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226 { |
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227 static int cnt = 0; |
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228 /* Note that this only works with thread implementations |
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229 that use a different process id for each thread. |
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230 */ |
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231 /* Check every 10 loops */ |
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232 if (this->hidden->parent && (((++cnt) % 10) == 0)) { |
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233 if (kill(this->hidden->parent, 0) < 0) { |
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234 this->enabled = 0; |
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235 } |
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236 } |
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237 } |
0 | 238 } |
239 | |
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240 |
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241 /* !!! FIXME: is there a channel swizzler in alsalib instead? */ |
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242 /* |
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243 * http://bugzilla.libsdl.org/show_bug.cgi?id=110 |
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244 * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE |
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245 * and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR" |
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246 */ |
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247 #define SWIZ6(T) \ |
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248 T *ptr = (T *) this->hidden->mixbuf; \ |
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249 const Uint32 count = (this->spec.samples / 6); \ |
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250 Uint32 i; \ |
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251 for (i = 0; i < count; i++, ptr += 6) { \ |
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252 T tmp; \ |
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253 tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \ |
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254 tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \ |
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255 } |
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256 |
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257 static __inline__ void |
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258 swizzle_alsa_channels_6_64bit(_THIS) |
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259 { |
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260 SWIZ6(Uint64); |
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261 } |
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262 static __inline__ void |
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263 swizzle_alsa_channels_6_32bit(_THIS) |
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264 { |
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265 SWIZ6(Uint32); |
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266 } |
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267 static __inline__ void |
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268 swizzle_alsa_channels_6_16bit(_THIS) |
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269 { |
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270 SWIZ6(Uint16); |
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271 } |
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272 static __inline__ void |
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273 swizzle_alsa_channels_6_8bit(_THIS) |
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274 { |
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275 SWIZ6(Uint8); |
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276 } |
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277 |
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278 #undef SWIZ6 |
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279 |
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280 |
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281 /* |
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282 * Called right before feeding this->hidden->mixbuf to the hardware. Swizzle |
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283 * channels from Windows/Mac order to the format alsalib will want. |
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284 */ |
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285 static __inline__ void |
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286 swizzle_alsa_channels(_THIS) |
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287 { |
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288 if (this->spec.channels == 6) { |
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289 const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */ |
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290 if (fmtsize == 16) |
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291 swizzle_alsa_channels_6_16bit(this); |
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292 else if (fmtsize == 8) |
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293 swizzle_alsa_channels_6_8bit(this); |
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294 else if (fmtsize == 32) |
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295 swizzle_alsa_channels_6_32bit(this); |
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296 else if (fmtsize == 64) |
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297 swizzle_alsa_channels_6_64bit(this); |
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298 } |
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299 |
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300 /* !!! FIXME: update this for 7.1 if needed, later. */ |
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301 } |
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302 |
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303 |
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304 static void |
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305 ALSA_PlayDevice(_THIS) |
0 | 306 { |
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307 int status; |
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308 int sample_len; |
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309 signed short *sample_buf; |
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310 |
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311 swizzle_alsa_channels(this); |
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312 |
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313 sample_len = this->spec.samples; |
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314 sample_buf = (signed short *) this->hidden->mixbuf; |
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315 |
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316 while (sample_len > 0) { |
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317 status = ALSA_snd_pcm_writei(this->hidden->pcm_handle, |
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318 sample_buf, sample_len); |
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319 |
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320 if (status < 0) { |
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321 if (status == -EAGAIN) { |
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322 SDL_Delay(1); |
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323 continue; |
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324 } |
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325 if (status == -ESTRPIPE) { |
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326 do { |
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327 SDL_Delay(1); |
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328 status = ALSA_snd_pcm_resume(this->hidden->pcm_handle); |
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329 } while (status == -EAGAIN); |
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330 } |
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331 if (status < 0) { |
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332 status = ALSA_snd_pcm_prepare(this->hidden->pcm_handle); |
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333 } |
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334 if (status < 0) { |
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335 /* Hmm, not much we can do - abort */ |
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336 this->enabled = 0; |
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337 return; |
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338 } |
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339 continue; |
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340 } |
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341 sample_buf += status * this->spec.channels; |
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342 sample_len -= status; |
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343 } |
0 | 344 } |
345 | |
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346 static Uint8 * |
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347 ALSA_GetDeviceBuf(_THIS) |
0 | 348 { |
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349 return (this->hidden->mixbuf); |
0 | 350 } |
351 | |
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352 static void |
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353 ALSA_CloseDevice(_THIS) |
0 | 354 { |
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355 if (this->hidden != NULL) { |
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356 if (this->hidden->mixbuf != NULL) { |
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357 SDL_FreeAudioMem(this->hidden->mixbuf); |
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358 this->hidden->mixbuf = NULL; |
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359 } |
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360 if (this->hidden->pcm_handle) { |
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361 ALSA_snd_pcm_drain(this->hidden->pcm_handle); |
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362 ALSA_snd_pcm_close(this->hidden->pcm_handle); |
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363 this->hidden->pcm_handle = NULL; |
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364 } |
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365 SDL_free(this->hidden); |
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366 this->hidden = NULL; |
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367 } |
0 | 368 } |
369 | |
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370 static int |
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371 ALSA_OpenDevice(_THIS, const char *devname, int iscapture) |
0 | 372 { |
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373 int status = 0; |
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374 snd_pcm_t *pcm_handle = NULL; |
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|
375 snd_pcm_hw_params_t *hwparams = NULL; |
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|
376 snd_pcm_sw_params_t *swparams = NULL; |
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|
377 snd_pcm_format_t format = 0; |
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|
378 snd_pcm_uframes_t frames = 0; |
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|
379 SDL_AudioFormat test_format = 0; |
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|
380 |
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|
381 /* Initialize all variables that we clean on shutdown */ |
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|
382 this->hidden = (struct SDL_PrivateAudioData *) |
2060 | 383 SDL_malloc((sizeof *this->hidden)); |
2049
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|
384 if (this->hidden == NULL) { |
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|
385 SDL_OutOfMemory(); |
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|
386 return 0; |
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|
387 } |
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388 SDL_memset(this->hidden, 0, (sizeof *this->hidden)); |
0 | 389 |
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390 /* Open the audio device */ |
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|
391 /* Name of device should depend on # channels in spec */ |
2049
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392 status = ALSA_snd_pcm_open(&pcm_handle, |
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|
393 get_audio_device(this->spec.channels), |
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|
394 SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
942
41a59de7f2ed
Here are patches for SDL12 and SDL_mixer for 4 or 6 channel
Sam Lantinga <slouken@libsdl.org>
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395 |
1895
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396 if (status < 0) { |
2049
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397 ALSA_CloseDevice(this); |
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398 SDL_SetError("ALSA: Couldn't open audio device: %s", |
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399 ALSA_snd_strerror(status)); |
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400 return 0; |
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401 } |
354
30935e76acb5
Updated ALSA audio support for ALSA 0.9
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402 |
2049
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403 this->hidden->pcm_handle = pcm_handle; |
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404 |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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405 /* Figure out what the hardware is capable of */ |
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406 snd_pcm_hw_params_alloca(&hwparams); |
2049
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407 status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams); |
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408 if (status < 0) { |
2049
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|
409 ALSA_CloseDevice(this); |
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410 SDL_SetError("ALSA: Couldn't get hardware config: %s", |
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|
411 ALSA_snd_strerror(status)); |
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412 return 0; |
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413 } |
354
30935e76acb5
Updated ALSA audio support for ALSA 0.9
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|
414 |
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|
415 /* SDL only uses interleaved sample output */ |
2049
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416 status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams, |
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|
417 SND_PCM_ACCESS_RW_INTERLEAVED); |
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418 if (status < 0) { |
2049
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419 ALSA_CloseDevice(this); |
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420 SDL_SetError("ALSA: Couldn't set interleaved access: %s", |
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421 ALSA_snd_strerror(status)); |
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422 return 0; |
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423 } |
0 | 424 |
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425 /* Try for a closest match on audio format */ |
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426 status = -1; |
2049
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427 for (test_format = SDL_FirstAudioFormat(this->spec.format); |
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428 test_format && (status < 0);) { |
2060 | 429 status = 0; /* if we can't support a format, it'll become -1. */ |
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430 switch (test_format) { |
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431 case AUDIO_U8: |
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432 format = SND_PCM_FORMAT_U8; |
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433 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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434 case AUDIO_S8: |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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435 format = SND_PCM_FORMAT_S8; |
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|
436 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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437 case AUDIO_S16LSB: |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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438 format = SND_PCM_FORMAT_S16_LE; |
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|
439 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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440 case AUDIO_S16MSB: |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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441 format = SND_PCM_FORMAT_S16_BE; |
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442 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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443 case AUDIO_U16LSB: |
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|
444 format = SND_PCM_FORMAT_U16_LE; |
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|
445 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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|
446 case AUDIO_U16MSB: |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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447 format = SND_PCM_FORMAT_U16_BE; |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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changeset
|
448 break; |
1995
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
Ryan C. Gordon <icculus@icculus.org>
parents:
1982
diff
changeset
|
449 case AUDIO_S32LSB: |
2010
39897da56f63
Whoops, wrong tokens for int32 support in ALSA driver (specified unsigned
Ryan C. Gordon <icculus@icculus.org>
parents:
2009
diff
changeset
|
450 format = SND_PCM_FORMAT_S32_LE; |
1995
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
Ryan C. Gordon <icculus@icculus.org>
parents:
1982
diff
changeset
|
451 break; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
Ryan C. Gordon <icculus@icculus.org>
parents:
1982
diff
changeset
|
452 case AUDIO_S32MSB: |
2010
39897da56f63
Whoops, wrong tokens for int32 support in ALSA driver (specified unsigned
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parents:
2009
diff
changeset
|
453 format = SND_PCM_FORMAT_S32_BE; |
1995
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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parents:
1982
diff
changeset
|
454 break; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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parents:
1982
diff
changeset
|
455 case AUDIO_F32LSB: |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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parents:
1982
diff
changeset
|
456 format = SND_PCM_FORMAT_FLOAT_LE; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
Ryan C. Gordon <icculus@icculus.org>
parents:
1982
diff
changeset
|
457 break; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
Ryan C. Gordon <icculus@icculus.org>
parents:
1982
diff
changeset
|
458 case AUDIO_F32MSB: |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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parents:
1982
diff
changeset
|
459 format = SND_PCM_FORMAT_FLOAT_BE; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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parents:
1982
diff
changeset
|
460 break; |
1895
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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changeset
|
461 default: |
2009
f2058fb367e4
ALSA was testing if (format) was set to zero as an error condition, but
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parents:
1995
diff
changeset
|
462 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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changeset
|
463 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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changeset
|
464 } |
2009
f2058fb367e4
ALSA was testing if (format) was set to zero as an error condition, but
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parents:
1995
diff
changeset
|
465 if (status >= 0) { |
2049
5f6550e5184f
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diff
changeset
|
466 status = ALSA_snd_pcm_hw_params_set_format(pcm_handle, |
5f6550e5184f
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changeset
|
467 hwparams, format); |
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|
468 } |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
469 if (status < 0) { |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
470 test_format = SDL_NextAudioFormat(); |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
471 } |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
472 } |
c121d94672cb
SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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changeset
|
473 if (status < 0) { |
2049
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changeset
|
474 ALSA_CloseDevice(this); |
5f6550e5184f
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changeset
|
475 SDL_SetError("ALSA: Couldn't find any hardware audio formats"); |
5f6550e5184f
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changeset
|
476 return 0; |
1895
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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changeset
|
477 } |
2049
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changeset
|
478 this->spec.format = test_format; |
0 | 479 |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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diff
changeset
|
480 /* Set the number of channels */ |
2049
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changeset
|
481 status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams, |
5f6550e5184f
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changeset
|
482 this->spec.channels); |
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483 if (status < 0) { |
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484 status = ALSA_snd_pcm_hw_params_get_channels(hwparams); |
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485 if ((status <= 0) || (status > 2)) { |
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486 ALSA_CloseDevice(this); |
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487 SDL_SetError("ALSA: Couldn't set audio channels"); |
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488 return 0; |
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489 } |
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490 this->spec.channels = status; |
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491 } |
0 | 492 |
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493 /* Set the audio rate */ |
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494 status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, |
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495 this->spec.freq, NULL); |
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496 if (status < 0) { |
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497 ALSA_CloseDevice(this); |
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498 SDL_SetError("ALSA: Couldn't set audio frequency: %s", |
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499 ALSA_snd_strerror(status)); |
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500 return 0; |
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501 } |
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502 this->spec.freq = status; |
0 | 503 |
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504 /* Set the buffer size, in samples */ |
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505 frames = this->spec.samples; |
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506 frames = ALSA_snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, |
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507 frames, NULL); |
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508 this->spec.samples = frames; |
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509 ALSA_snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, 2, NULL); |
354
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510 |
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511 /* "set" the hardware with the desired parameters */ |
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512 status = ALSA_snd_pcm_hw_params(pcm_handle, hwparams); |
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513 if (status < 0) { |
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514 ALSA_CloseDevice(this); |
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515 SDL_SetError("ALSA: Couldn't set hardware audio parameters: %s", |
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516 ALSA_snd_strerror(status)); |
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517 return 0; |
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518 } |
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519 #if AUDIO_DEBUG |
2060 | 520 { |
521 snd_pcm_sframes_t bufsize; | |
522 int fragments; | |
523 bufsize = ALSA_snd_pcm_hw_params_get_period_size(hwparams); | |
524 fragments = ALSA_snd_pcm_hw_params_get_periods(hwparams); | |
525 fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, | |
526 fragments); | |
527 } | |
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528 #endif |
1552 | 529 |
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530 /* Set the software parameters */ |
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531 snd_pcm_sw_params_alloca(&swparams); |
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532 status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams); |
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533 if (status < 0) { |
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534 ALSA_CloseDevice(this); |
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535 SDL_SetError("ALSA: Couldn't get software config: %s", |
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536 ALSA_snd_strerror(status)); |
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537 return 0; |
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538 } |
2060 | 539 status = |
540 ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 0); | |
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541 if (status < 0) { |
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542 ALSA_CloseDevice(this); |
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543 SDL_SetError("ALSA: Couldn't set start threshold: %s", |
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544 ALSA_snd_strerror(status)); |
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545 return 0; |
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546 } |
2060 | 547 status = |
548 ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, frames); | |
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549 if (status < 0) { |
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550 ALSA_CloseDevice(this); |
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551 SDL_SetError("Couldn't set avail min: %s", ALSA_snd_strerror(status)); |
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552 return 0; |
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553 } |
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554 status = ALSA_snd_pcm_sw_params(pcm_handle, swparams); |
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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 software audio parameters: %s", |
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558 ALSA_snd_strerror(status)); |
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559 return 0; |
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560 } |
0 | 561 |
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562 /* Calculate the final parameters for this audio specification */ |
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563 SDL_CalculateAudioSpec(&this->spec); |
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564 |
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565 /* Allocate mixing buffer */ |
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566 this->hidden->mixlen = this->spec.size; |
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567 this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen); |
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568 if (this->hidden->mixbuf == NULL) { |
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|
569 ALSA_CloseDevice(this); |
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570 SDL_OutOfMemory(); |
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571 return 0; |
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572 } |
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573 SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size); |
0 | 574 |
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575 /* Get the parent process id (we're the parent of the audio thread) */ |
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576 this->hidden->parent = getpid(); |
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577 |
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578 /* Switch to blocking mode for playback */ |
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579 ALSA_snd_pcm_nonblock(pcm_handle, 0); |
0 | 580 |
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581 /* We're ready to rock and roll. :-) */ |
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582 return 1; |
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583 } |
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584 |
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585 static void |
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586 ALSA_Deinitialize(void) |
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587 { |
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588 UnloadALSALibrary(); |
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589 } |
0 | 590 |
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591 static int |
2060 | 592 ALSA_Init(SDL_AudioDriverImpl * impl) |
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593 { |
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594 if (LoadALSALibrary() < 0) { |
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595 return 0; |
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596 } |
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597 |
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598 /* Set the function pointers */ |
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599 impl->OpenDevice = ALSA_OpenDevice; |
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600 impl->WaitDevice = ALSA_WaitDevice; |
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601 impl->GetDeviceBuf = ALSA_GetDeviceBuf; |
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602 impl->PlayDevice = ALSA_PlayDevice; |
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603 impl->CloseDevice = ALSA_CloseDevice; |
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604 impl->Deinitialize = ALSA_Deinitialize; |
2060 | 605 impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: Add device enum! */ |
2049
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606 |
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607 return 1; |
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608 } |
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609 |
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610 |
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611 AudioBootStrap ALSA_bootstrap = { |
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612 DRIVER_NAME, "ALSA 0.9 PCM audio", ALSA_Init, 0 |
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613 }; |
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614 |
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615 /* vi: set ts=4 sw=4 expandtab: */ |