Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 3099:82e60908fab1
Date: Mon, 23 Mar 2009 09:17:24 +0200
From: "Mike Gorchak"
Subject: New QNX patches
Please apply patch qnx4.diff, which is attached. What has been done:
1)Added back OpenGL ES renderer for QNX target. Added few corrections to
OpenGL ES renderer to let it work under QNX. OpenGL ES renderer do not
support textures under QNX, so I think some additional work must be done.
2) Added GL_OES_query_matrix extension to SDL_opengles.h header file, which
required by OpenGL ES 1.1 specification.
3) Added attribute clearing at the entrance of function
SDL_GL_GetAttribure(). Added error checking into the function
SDL_GL_GetAttribure(), because some attributes can't be obtained in OpenGL
ES 1.0.
4) Porting testdyngles to OpenGL ES 1.0 (1.1 has glColor4ub() and
glColor4f() functions, but 1.0 has glColor4f() only).
5) Added error checking after obtaining attributes using
SDL_GL_GetAttribute() function to the testgl2 and testgles.
6) Small correction to testmultiaudio with printing errors.
7) Added software and accelerated OpenGL ES 1.0 support into the QNX GF
driver.
Please remove ./src/audio/nto directory - it will not be used anymore.
Please create ./src/audio/qsa directory and add content of the archive
qsa.tar.gz into this directory. I rewrote some sound code, added support for
multiple audio cards, enumeration, etc. Added initial support for capture.
As far as I can understand SDL 1.3 is not supporting audio capture right now
? Sam, Am I right ? Or audio capture must be supported through the
PlayDevice routine ?
And last, please put file SDL_gf_opengles.c to the ./src/video/qnxgf
directory. It is OpenGL ES 1.1 emulation layer for some functions, which are
not supported by OpenGL ES 1.0.
author | Sam Lantinga <slouken@libsdl.org> |
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date | Tue, 24 Mar 2009 10:33:12 +0000 |
parents | b21348d47cab |
children | 4e83cdb58134 |
rev | line source |
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0 | 1 /* |
2 SDL - Simple DirectMedia Layer | |
2859 | 3 Copyright (C) 1997-2009 Sam Lantinga |
0 | 4 |
5 This library is free software; you can redistribute it and/or | |
6 modify it under the terms of the GNU Library General Public | |
7 License as published by the Free Software Foundation; either | |
8 version 2 of the License, or (at your option) any later version. | |
9 | |
10 This library is distributed in the hope that it will be useful, | |
11 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 Library General Public License for more details. | |
14 | |
15 You should have received a copy of the GNU Library General Public | |
16 License along with this library; if not, write to the Free | |
17 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
18 | |
19 Sam Lantinga | |
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20 slouken@libsdl.org |
0 | 21 */ |
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22 #include "SDL_config.h" |
0 | 23 |
24 /* Allow access to a raw mixing buffer */ | |
25 | |
26 #include <sys/types.h> | |
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27 #include <signal.h> /* For kill() */ |
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28 #include <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)) { |
3068 | 233 if (kill(this->hidden->parent, 0) < 0 && errno == ESRCH) { |
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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 |
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263 static __inline__ void |
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264 swizzle_alsa_channels_6_32bit(_THIS) |
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265 { |
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266 SWIZ6(Uint32); |
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267 } |
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268 |
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269 static __inline__ void |
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270 swizzle_alsa_channels_6_16bit(_THIS) |
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271 { |
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272 SWIZ6(Uint16); |
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273 } |
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274 |
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275 static __inline__ void |
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276 swizzle_alsa_channels_6_8bit(_THIS) |
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277 { |
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278 SWIZ6(Uint8); |
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279 } |
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280 |
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281 #undef SWIZ6 |
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282 |
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283 |
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284 /* |
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285 * Called right before feeding this->hidden->mixbuf to the hardware. Swizzle |
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286 * channels from Windows/Mac order to the format alsalib will want. |
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287 */ |
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288 static __inline__ void |
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289 swizzle_alsa_channels(_THIS) |
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290 { |
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291 if (this->spec.channels == 6) { |
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292 const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */ |
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293 if (fmtsize == 16) |
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294 swizzle_alsa_channels_6_16bit(this); |
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295 else if (fmtsize == 8) |
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296 swizzle_alsa_channels_6_8bit(this); |
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297 else if (fmtsize == 32) |
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298 swizzle_alsa_channels_6_32bit(this); |
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299 else if (fmtsize == 64) |
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300 swizzle_alsa_channels_6_64bit(this); |
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301 } |
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302 |
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303 /* !!! FIXME: update this for 7.1 if needed, later. */ |
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304 } |
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305 |
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306 |
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307 static void |
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308 ALSA_PlayDevice(_THIS) |
0 | 309 { |
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310 int status; |
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311 int sample_len; |
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312 signed short *sample_buf; |
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313 |
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314 swizzle_alsa_channels(this); |
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315 |
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316 sample_len = this->spec.samples; |
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317 sample_buf = (signed short *) this->hidden->mixbuf; |
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318 |
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319 while (sample_len > 0) { |
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320 status = ALSA_snd_pcm_writei(this->hidden->pcm_handle, |
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321 sample_buf, sample_len); |
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322 |
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323 if (status < 0) { |
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324 if (status == -EAGAIN) { |
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325 SDL_Delay(1); |
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326 continue; |
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327 } |
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328 if (status == -ESTRPIPE) { |
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329 do { |
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330 SDL_Delay(1); |
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331 status = ALSA_snd_pcm_resume(this->hidden->pcm_handle); |
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332 } while (status == -EAGAIN); |
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333 } |
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334 if (status < 0) { |
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335 status = ALSA_snd_pcm_prepare(this->hidden->pcm_handle); |
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336 } |
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337 if (status < 0) { |
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338 /* Hmm, not much we can do - abort */ |
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339 this->enabled = 0; |
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340 return; |
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341 } |
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342 continue; |
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343 } |
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344 sample_buf += status * this->spec.channels; |
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345 sample_len -= status; |
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346 } |
0 | 347 } |
348 | |
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349 static Uint8 * |
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350 ALSA_GetDeviceBuf(_THIS) |
0 | 351 { |
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352 return (this->hidden->mixbuf); |
0 | 353 } |
354 | |
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355 static void |
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356 ALSA_CloseDevice(_THIS) |
0 | 357 { |
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358 if (this->hidden != NULL) { |
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359 if (this->hidden->mixbuf != NULL) { |
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360 SDL_FreeAudioMem(this->hidden->mixbuf); |
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361 this->hidden->mixbuf = NULL; |
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362 } |
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363 if (this->hidden->pcm_handle) { |
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364 ALSA_snd_pcm_drain(this->hidden->pcm_handle); |
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365 ALSA_snd_pcm_close(this->hidden->pcm_handle); |
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366 this->hidden->pcm_handle = NULL; |
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367 } |
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368 SDL_free(this->hidden); |
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369 this->hidden = NULL; |
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370 } |
0 | 371 } |
372 | |
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373 static int |
2049
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374 ALSA_OpenDevice(_THIS, const char *devname, int iscapture) |
0 | 375 { |
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376 int status = 0; |
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377 snd_pcm_t *pcm_handle = NULL; |
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378 snd_pcm_hw_params_t *hwparams = NULL; |
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379 snd_pcm_sw_params_t *swparams = NULL; |
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380 snd_pcm_format_t format = 0; |
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381 snd_pcm_uframes_t frames = 0; |
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382 SDL_AudioFormat test_format = 0; |
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383 |
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384 /* Initialize all variables that we clean on shutdown */ |
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385 this->hidden = (struct SDL_PrivateAudioData *) |
2060 | 386 SDL_malloc((sizeof *this->hidden)); |
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387 if (this->hidden == NULL) { |
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388 SDL_OutOfMemory(); |
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389 return 0; |
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390 } |
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391 SDL_memset(this->hidden, 0, (sizeof *this->hidden)); |
0 | 392 |
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393 /* Open the audio device */ |
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394 /* Name of device should depend on # channels in spec */ |
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395 status = ALSA_snd_pcm_open(&pcm_handle, |
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396 get_audio_device(this->spec.channels), |
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397 SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
942
41a59de7f2ed
Here are patches for SDL12 and SDL_mixer for 4 or 6 channel
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398 |
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399 if (status < 0) { |
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400 ALSA_CloseDevice(this); |
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401 SDL_SetError("ALSA: Couldn't open audio device: %s", |
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402 ALSA_snd_strerror(status)); |
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403 return 0; |
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404 } |
354
30935e76acb5
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405 |
2049
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406 this->hidden->pcm_handle = pcm_handle; |
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407 |
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408 /* Figure out what the hardware is capable of */ |
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409 snd_pcm_hw_params_alloca(&hwparams); |
2049
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410 status = ALSA_snd_pcm_hw_params_any(pcm_handle, hwparams); |
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411 if (status < 0) { |
2049
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412 ALSA_CloseDevice(this); |
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413 SDL_SetError("ALSA: Couldn't get hardware config: %s", |
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414 ALSA_snd_strerror(status)); |
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415 return 0; |
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416 } |
354
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Updated ALSA audio support for ALSA 0.9
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417 |
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418 /* SDL only uses interleaved sample output */ |
2049
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419 status = ALSA_snd_pcm_hw_params_set_access(pcm_handle, hwparams, |
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420 SND_PCM_ACCESS_RW_INTERLEAVED); |
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421 if (status < 0) { |
2049
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422 ALSA_CloseDevice(this); |
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423 SDL_SetError("ALSA: Couldn't set interleaved access: %s", |
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424 ALSA_snd_strerror(status)); |
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425 return 0; |
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426 } |
0 | 427 |
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428 /* Try for a closest match on audio format */ |
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429 status = -1; |
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430 for (test_format = SDL_FirstAudioFormat(this->spec.format); |
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431 test_format && (status < 0);) { |
2060 | 432 status = 0; /* if we can't support a format, it'll become -1. */ |
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433 switch (test_format) { |
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434 case AUDIO_U8: |
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435 format = SND_PCM_FORMAT_U8; |
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436 break; |
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437 case AUDIO_S8: |
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438 format = SND_PCM_FORMAT_S8; |
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439 break; |
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440 case AUDIO_S16LSB: |
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441 format = SND_PCM_FORMAT_S16_LE; |
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442 break; |
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443 case AUDIO_S16MSB: |
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444 format = SND_PCM_FORMAT_S16_BE; |
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445 break; |
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446 case AUDIO_U16LSB: |
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447 format = SND_PCM_FORMAT_U16_LE; |
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448 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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449 case AUDIO_U16MSB: |
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450 format = SND_PCM_FORMAT_U16_BE; |
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451 break; |
1995
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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parents:
1982
diff
changeset
|
452 case AUDIO_S32LSB: |
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_LE; |
1995
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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diff
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|
454 break; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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diff
changeset
|
455 case AUDIO_S32MSB: |
2010
39897da56f63
Whoops, wrong tokens for int32 support in ALSA driver (specified unsigned
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parents:
2009
diff
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|
456 format = SND_PCM_FORMAT_S32_BE; |
1995
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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parents:
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diff
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|
457 break; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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diff
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|
458 case AUDIO_F32LSB: |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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459 format = SND_PCM_FORMAT_FLOAT_LE; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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diff
changeset
|
460 break; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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1982
diff
changeset
|
461 case AUDIO_F32MSB: |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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diff
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462 format = SND_PCM_FORMAT_FLOAT_BE; |
0ca6ba107642
ALSA backend can handle int32 and float32 data directly.
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diff
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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|
464 default: |
2009
f2058fb367e4
ALSA was testing if (format) was set to zero as an error condition, but
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diff
changeset
|
465 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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|
466 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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|
467 } |
2009
f2058fb367e4
ALSA was testing if (format) was set to zero as an error condition, but
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1995
diff
changeset
|
468 if (status >= 0) { |
2049
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|
469 status = ALSA_snd_pcm_hw_params_set_format(pcm_handle, |
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changeset
|
470 hwparams, format); |
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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 if (status < 0) { |
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|
473 test_format = SDL_NextAudioFormat(); |
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474 } |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
475 } |
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SDL 1.2 is moving to a branch, and SDL 1.3 is becoming the head.
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|
476 if (status < 0) { |
2049
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changeset
|
477 ALSA_CloseDevice(this); |
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478 SDL_SetError("ALSA: Couldn't find any hardware audio formats"); |
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479 return 0; |
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480 } |
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481 this->spec.format = test_format; |
0 | 482 |
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483 /* Set the number of channels */ |
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484 status = ALSA_snd_pcm_hw_params_set_channels(pcm_handle, hwparams, |
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485 this->spec.channels); |
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486 if (status < 0) { |
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487 status = ALSA_snd_pcm_hw_params_get_channels(hwparams); |
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488 if ((status <= 0) || (status > 2)) { |
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489 ALSA_CloseDevice(this); |
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490 SDL_SetError("ALSA: Couldn't set audio channels"); |
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491 return 0; |
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492 } |
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493 this->spec.channels = status; |
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494 } |
0 | 495 |
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496 /* Set the audio rate */ |
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497 status = ALSA_snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, |
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498 this->spec.freq, NULL); |
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499 if (status < 0) { |
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500 ALSA_CloseDevice(this); |
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501 SDL_SetError("ALSA: Couldn't set audio frequency: %s", |
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502 ALSA_snd_strerror(status)); |
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503 return 0; |
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504 } |
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505 this->spec.freq = status; |
0 | 506 |
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507 /* Set the buffer size, in samples */ |
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508 frames = this->spec.samples; |
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509 frames = ALSA_snd_pcm_hw_params_set_period_size_near(pcm_handle, hwparams, |
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510 frames, NULL); |
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511 this->spec.samples = frames; |
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512 ALSA_snd_pcm_hw_params_set_periods_near(pcm_handle, hwparams, 2, NULL); |
354
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513 |
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514 /* "set" the hardware with the desired parameters */ |
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515 status = ALSA_snd_pcm_hw_params(pcm_handle, hwparams); |
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516 if (status < 0) { |
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517 ALSA_CloseDevice(this); |
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518 SDL_SetError("ALSA: Couldn't set hardware audio parameters: %s", |
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519 ALSA_snd_strerror(status)); |
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520 return 0; |
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521 } |
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522 #if AUDIO_DEBUG |
2060 | 523 { |
524 snd_pcm_sframes_t bufsize; | |
525 int fragments; | |
526 bufsize = ALSA_snd_pcm_hw_params_get_period_size(hwparams); | |
527 fragments = ALSA_snd_pcm_hw_params_get_periods(hwparams); | |
528 fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, | |
529 fragments); | |
530 } | |
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531 #endif |
1552 | 532 |
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533 /* Set the software parameters */ |
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534 snd_pcm_sw_params_alloca(&swparams); |
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535 status = ALSA_snd_pcm_sw_params_current(pcm_handle, swparams); |
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536 if (status < 0) { |
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537 ALSA_CloseDevice(this); |
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538 SDL_SetError("ALSA: Couldn't get software config: %s", |
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539 ALSA_snd_strerror(status)); |
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540 return 0; |
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541 } |
2060 | 542 status = |
543 ALSA_snd_pcm_sw_params_set_start_threshold(pcm_handle, swparams, 0); | |
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544 if (status < 0) { |
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545 ALSA_CloseDevice(this); |
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546 SDL_SetError("ALSA: Couldn't set start threshold: %s", |
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547 ALSA_snd_strerror(status)); |
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548 return 0; |
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549 } |
2060 | 550 status = |
551 ALSA_snd_pcm_sw_params_set_avail_min(pcm_handle, swparams, frames); | |
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552 if (status < 0) { |
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553 ALSA_CloseDevice(this); |
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554 SDL_SetError("Couldn't set avail min: %s", ALSA_snd_strerror(status)); |
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555 return 0; |
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556 } |
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557 status = ALSA_snd_pcm_sw_params(pcm_handle, swparams); |
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558 if (status < 0) { |
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559 ALSA_CloseDevice(this); |
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560 SDL_SetError("Couldn't set software audio parameters: %s", |
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561 ALSA_snd_strerror(status)); |
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562 return 0; |
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563 } |
0 | 564 |
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565 /* Calculate the final parameters for this audio specification */ |
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566 SDL_CalculateAudioSpec(&this->spec); |
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567 |
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568 /* Allocate mixing buffer */ |
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569 this->hidden->mixlen = this->spec.size; |
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570 this->hidden->mixbuf = (Uint8 *) SDL_AllocAudioMem(this->hidden->mixlen); |
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571 if (this->hidden->mixbuf == NULL) { |
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572 ALSA_CloseDevice(this); |
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573 SDL_OutOfMemory(); |
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574 return 0; |
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575 } |
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576 SDL_memset(this->hidden->mixbuf, this->spec.silence, this->spec.size); |
0 | 577 |
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578 /* Get the parent process id (we're the parent of the audio thread) */ |
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579 this->hidden->parent = getpid(); |
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580 |
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581 /* Switch to blocking mode for playback */ |
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582 ALSA_snd_pcm_nonblock(pcm_handle, 0); |
0 | 583 |
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584 /* We're ready to rock and roll. :-) */ |
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585 return 1; |
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586 } |
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587 |
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588 static void |
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589 ALSA_Deinitialize(void) |
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590 { |
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591 UnloadALSALibrary(); |
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592 } |
0 | 593 |
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594 static int |
2060 | 595 ALSA_Init(SDL_AudioDriverImpl * impl) |
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596 { |
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597 if (LoadALSALibrary() < 0) { |
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598 return 0; |
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599 } |
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600 |
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601 /* Set the function pointers */ |
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602 impl->OpenDevice = ALSA_OpenDevice; |
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603 impl->WaitDevice = ALSA_WaitDevice; |
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604 impl->GetDeviceBuf = ALSA_GetDeviceBuf; |
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605 impl->PlayDevice = ALSA_PlayDevice; |
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606 impl->CloseDevice = ALSA_CloseDevice; |
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607 impl->Deinitialize = ALSA_Deinitialize; |
2060 | 608 impl->OnlyHasDefaultOutputDevice = 1; /* !!! FIXME: Add device enum! */ |
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609 |
2942 | 610 return 1; /* !!! FIXME: return 2 once device enum is implemented. */ |
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611 } |
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612 |
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613 |
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614 AudioBootStrap ALSA_bootstrap = { |
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615 DRIVER_NAME, "ALSA 0.9 PCM audio", ALSA_Init, 0 |
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616 }; |
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617 |
1895
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618 /* vi: set ts=4 sw=4 expandtab: */ |