Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 2270:d5a11262f067
Date: Sun, 19 Aug 2007 18:29:28 +0200
From: Christian Walther
Subject: Re: SDL 1.3 keyboard plan
> 2007-08-18 19:15:51.454 checkkeys[5795] *** _NSAutoreleaseNoPool():
> Object 0x532750 of class NSSelectionArray autoreleased with no pool
> in place - just leaking
This is fixed by the attached patch.
author | Sam Lantinga <slouken@libsdl.org> |
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date | Sun, 19 Aug 2007 16:36:51 +0000 |
parents | 866052b01ee5 |
children | e1da92da346c |
rev | line source |
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0 | 1 /* |
2 SDL - Simple DirectMedia Layer | |
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Updated copyright information for 2004 (Happy New Year!)
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3 Copyright (C) 1997-2004 Sam Lantinga |
0 | 4 |
5 This library is free software; you can redistribute it and/or | |
6 modify it under the terms of the GNU Library General Public | |
7 License as published by the Free Software Foundation; either | |
8 version 2 of the License, or (at your option) any later version. | |
9 | |
10 This library is distributed in the hope that it will be useful, | |
11 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 Library General Public License for more details. | |
14 | |
15 You should have received a copy of the GNU Library General Public | |
16 License along with this library; if not, write to the Free | |
17 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
18 | |
19 Sam Lantinga | |
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20 slouken@libsdl.org |
0 | 21 */ |
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22 #include "SDL_config.h" |
0 | 23 |
24 /* Allow access to a raw mixing buffer */ | |
25 | |
26 #include <sys/types.h> | |
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27 #include <signal.h> /* For kill() */ |
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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) |
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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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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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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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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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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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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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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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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
444 format = SND_PCM_FORMAT_U16_LE; |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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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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|
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.
Ryan C. Gordon <icculus@icculus.org>
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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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; |
c121d94672cb
SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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diff
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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diff
changeset
|
467 hwparams, format); |
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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diff
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|
468 } |
c121d94672cb
SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
469 if (status < 0) { |
c121d94672cb
SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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changeset
|
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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diff
changeset
|
475 SDL_SetError("ALSA: Couldn't find any hardware audio formats"); |
5f6550e5184f
Merged SDL-ryan-multiple-audio-device branch r2803:2871 into the trunk.
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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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diff
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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diff
changeset
|
481 status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams, |
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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: */ |