Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 955:d74fbf56f2f6
Date: Fri, 25 Jun 2004 13:29:15 +0100
From: "alan buckley"
Subject: Modification for RISC OS version of SDL
Ive attached a zip file with the changes to this email, it contains the
following:
The file sdldiff.txt is the output from cvs diff u. .
The directory thread/riscos contains all the new files to support threading.
Readme.riscos is a new readme file to add.
author | Sam Lantinga <slouken@libsdl.org> |
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date | Fri, 17 Sep 2004 13:20:10 +0000 |
parents | 41a59de7f2ed |
children | 05d4b93b911e |
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 */ |
22 | |
23 | |
24 | |
25 /* Allow access to a raw mixing buffer */ | |
26 | |
27 #include <stdlib.h> | |
28 #include <stdio.h> | |
29 #include <string.h> | |
30 #include <errno.h> | |
31 #include <unistd.h> | |
32 #include <fcntl.h> | |
33 #include <signal.h> | |
34 #include <sys/types.h> | |
35 #include <sys/time.h> | |
36 | |
37 #include "SDL_audio.h" | |
38 #include "SDL_error.h" | |
39 #include "SDL_audiomem.h" | |
40 #include "SDL_audio_c.h" | |
41 #include "SDL_timer.h" | |
42 #include "SDL_alsa_audio.h" | |
43 | |
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44 #ifdef ALSA_DYNAMIC |
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45 #ifdef USE_DLVSYM |
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46 #define __USE_GNU |
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47 #endif |
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48 #include <dlfcn.h> |
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49 #include "SDL_name.h" |
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50 #include "SDL_loadso.h" |
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51 #else |
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52 #define SDL_NAME(X) X |
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53 #endif |
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54 |
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55 |
0 | 56 /* The tag name used by ALSA audio */ |
57 #define DRIVER_NAME "alsa" | |
58 | |
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59 /* The default ALSA audio driver */ |
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60 #define DEFAULT_DEVICE "default" |
0 | 61 |
62 /* Audio driver functions */ | |
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63 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec); |
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64 static void ALSA_WaitAudio(_THIS); |
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65 static void ALSA_PlayAudio(_THIS); |
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66 static Uint8 *ALSA_GetAudioBuf(_THIS); |
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67 static void ALSA_CloseAudio(_THIS); |
0 | 68 |
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69 #ifdef ALSA_DYNAMIC |
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70 |
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71 static const char *alsa_library = ALSA_DYNAMIC; |
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72 static void *alsa_handle = NULL; |
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73 static int alsa_loaded = 0; |
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74 |
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75 static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); |
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76 static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); |
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77 static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm); |
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78 static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size); |
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79 static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm); |
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80 static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm); |
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81 static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm); |
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82 static const char *(*SDL_NAME(snd_strerror))(int errnum); |
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83 static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void); |
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84 static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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85 static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access); |
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86 static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val); |
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87 static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val); |
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88 static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params); |
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89 static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir); |
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90 static snd_pcm_uframes_t (*SDL_NAME(snd_pcm_hw_params_set_period_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val, int *dir); |
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91 static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir); |
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92 static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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93 static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock); |
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94 #define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof) |
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95 |
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96 static struct { |
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97 const char *name; |
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98 void **func; |
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99 } alsa_functions[] = { |
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100 { "snd_pcm_open", (void**)&SDL_NAME(snd_pcm_open) }, |
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101 { "snd_pcm_close", (void**)&SDL_NAME(snd_pcm_close) }, |
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102 { "snd_pcm_writei", (void**)&SDL_NAME(snd_pcm_writei) }, |
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103 { "snd_pcm_resume", (void**)&SDL_NAME(snd_pcm_resume) }, |
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104 { "snd_pcm_prepare", (void**)&SDL_NAME(snd_pcm_prepare) }, |
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105 { "snd_pcm_drain", (void**)&SDL_NAME(snd_pcm_drain) }, |
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106 { "snd_strerror", (void**)&SDL_NAME(snd_strerror) }, |
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107 { "snd_pcm_hw_params_sizeof", (void**)&SDL_NAME(snd_pcm_hw_params_sizeof) }, |
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108 { "snd_pcm_hw_params_any", (void**)&SDL_NAME(snd_pcm_hw_params_any) }, |
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109 { "snd_pcm_hw_params_set_access", (void**)&SDL_NAME(snd_pcm_hw_params_set_access) }, |
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110 { "snd_pcm_hw_params_set_format", (void**)&SDL_NAME(snd_pcm_hw_params_set_format) }, |
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111 { "snd_pcm_hw_params_set_channels", (void**)&SDL_NAME(snd_pcm_hw_params_set_channels) }, |
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112 { "snd_pcm_hw_params_get_channels", (void**)&SDL_NAME(snd_pcm_hw_params_get_channels) }, |
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113 { "snd_pcm_hw_params_set_rate_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_rate_near) }, |
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114 { "snd_pcm_hw_params_set_period_size_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) }, |
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115 { "snd_pcm_hw_params_set_periods_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_periods_near) }, |
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116 { "snd_pcm_hw_params", (void**)&SDL_NAME(snd_pcm_hw_params) }, |
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117 { "snd_pcm_nonblock", (void**)&SDL_NAME(snd_pcm_nonblock) }, |
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118 }; |
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119 |
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120 static void UnloadALSALibrary(void) { |
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121 if (alsa_loaded) { |
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122 /* SDL_UnloadObject(alsa_handle);*/ |
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123 dlclose(alsa_handle); |
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124 alsa_handle = NULL; |
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125 alsa_loaded = 0; |
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126 } |
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127 } |
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128 |
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129 static int LoadALSALibrary(void) { |
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130 int i, retval = -1; |
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131 |
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132 /* alsa_handle = SDL_LoadObject(alsa_library);*/ |
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133 alsa_handle = dlopen(alsa_library,RTLD_NOW); |
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134 if (alsa_handle) { |
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135 alsa_loaded = 1; |
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136 retval = 0; |
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137 for (i = 0; i < SDL_TABLESIZE(alsa_functions); i++) { |
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138 /* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/ |
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139 #ifdef USE_DLVSYM |
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140 *alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9"); |
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141 if (!*alsa_functions[i].func) |
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142 #endif |
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143 *alsa_functions[i].func = dlsym(alsa_handle,alsa_functions[i].name); |
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144 if (!*alsa_functions[i].func) { |
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145 retval = -1; |
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146 UnloadALSALibrary(); |
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147 break; |
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148 } |
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149 } |
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150 } |
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151 return retval; |
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152 } |
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153 |
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154 #else |
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155 |
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156 static void UnloadALSALibrary(void) { |
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157 return; |
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158 } |
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159 |
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160 static int LoadALSALibrary(void) { |
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161 return 0; |
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162 } |
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163 |
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164 #endif /* ALSA_DYNAMIC */ |
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165 |
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166 static const char *get_audio_device(int channels) |
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167 { |
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168 const char *device; |
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169 |
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170 device = getenv("AUDIODEV"); /* Is there a standard variable name? */ |
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171 if ( device == NULL ) { |
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172 if (channels == 6) device = "surround51"; |
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173 else if (channels == 4) device = "surround40"; |
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174 else device = DEFAULT_DEVICE; |
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175 } |
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176 return device; |
0 | 177 } |
178 | |
179 /* Audio driver bootstrap functions */ | |
180 | |
181 static int Audio_Available(void) | |
182 { | |
183 int available; | |
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184 int status; |
0 | 185 snd_pcm_t *handle; |
186 | |
187 available = 0; | |
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188 if (LoadALSALibrary() < 0) { |
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189 return available; |
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190 } |
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191 status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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192 if ( status >= 0 ) { |
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193 available = 1; |
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194 SDL_NAME(snd_pcm_close)(handle); |
0 | 195 } |
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196 UnloadALSALibrary(); |
0 | 197 return(available); |
198 } | |
199 | |
200 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
201 { | |
202 free(device->hidden); | |
203 free(device); | |
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204 UnloadALSALibrary(); |
0 | 205 } |
206 | |
207 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
208 { | |
209 SDL_AudioDevice *this; | |
210 | |
211 /* Initialize all variables that we clean on shutdown */ | |
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212 LoadALSALibrary(); |
0 | 213 this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice)); |
214 if ( this ) { | |
215 memset(this, 0, (sizeof *this)); | |
216 this->hidden = (struct SDL_PrivateAudioData *) | |
217 malloc((sizeof *this->hidden)); | |
218 } | |
219 if ( (this == NULL) || (this->hidden == NULL) ) { | |
220 SDL_OutOfMemory(); | |
221 if ( this ) { | |
222 free(this); | |
223 } | |
224 return(0); | |
225 } | |
226 memset(this->hidden, 0, (sizeof *this->hidden)); | |
227 | |
228 /* Set the function pointers */ | |
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229 this->OpenAudio = ALSA_OpenAudio; |
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230 this->WaitAudio = ALSA_WaitAudio; |
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231 this->PlayAudio = ALSA_PlayAudio; |
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232 this->GetAudioBuf = ALSA_GetAudioBuf; |
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233 this->CloseAudio = ALSA_CloseAudio; |
0 | 234 |
235 this->free = Audio_DeleteDevice; | |
236 | |
237 return this; | |
238 } | |
239 | |
240 AudioBootStrap ALSA_bootstrap = { | |
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241 DRIVER_NAME, "ALSA 0.9 PCM audio", |
0 | 242 Audio_Available, Audio_CreateDevice |
243 }; | |
244 | |
245 /* This function waits until it is possible to write a full sound buffer */ | |
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246 static void ALSA_WaitAudio(_THIS) |
0 | 247 { |
248 /* Check to see if the thread-parent process is still alive */ | |
249 { static int cnt = 0; | |
250 /* Note that this only works with thread implementations | |
251 that use a different process id for each thread. | |
252 */ | |
253 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ | |
254 if ( kill(parent, 0) < 0 ) { | |
255 this->enabled = 0; | |
256 } | |
257 } | |
258 } | |
259 } | |
260 | |
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261 static void ALSA_PlayAudio(_THIS) |
0 | 262 { |
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263 int status; |
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264 int sample_len; |
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265 signed short *sample_buf; |
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266 |
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267 sample_len = this->spec.samples; |
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268 sample_buf = (signed short *)mixbuf; |
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269 while ( sample_len > 0 ) { |
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270 status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len); |
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271 if ( status < 0 ) { |
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272 if ( status == -EAGAIN ) { |
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273 SDL_Delay(1); |
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274 continue; |
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275 } |
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276 if ( status == -ESTRPIPE ) { |
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277 do { |
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278 SDL_Delay(1); |
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279 status = SDL_NAME(snd_pcm_resume)(pcm_handle); |
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280 } while ( status == -EAGAIN ); |
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281 } |
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282 if ( status < 0 ) { |
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283 status = SDL_NAME(snd_pcm_prepare)(pcm_handle); |
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284 } |
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285 if ( status < 0 ) { |
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286 /* Hmm, not much we can do - abort */ |
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287 this->enabled = 0; |
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288 return; |
0 | 289 } |
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290 continue; |
0 | 291 } |
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292 sample_buf += status * this->spec.channels; |
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293 sample_len -= status; |
0 | 294 } |
295 } | |
296 | |
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297 static Uint8 *ALSA_GetAudioBuf(_THIS) |
0 | 298 { |
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299 return(mixbuf); |
0 | 300 } |
301 | |
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302 static void ALSA_CloseAudio(_THIS) |
0 | 303 { |
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304 if ( mixbuf != NULL ) { |
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305 SDL_FreeAudioMem(mixbuf); |
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306 mixbuf = NULL; |
0 | 307 } |
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308 if ( pcm_handle ) { |
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309 SDL_NAME(snd_pcm_drain)(pcm_handle); |
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310 SDL_NAME(snd_pcm_close)(pcm_handle); |
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311 pcm_handle = NULL; |
0 | 312 } |
313 } | |
314 | |
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315 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec) |
0 | 316 { |
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317 int status; |
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318 snd_pcm_hw_params_t *params; |
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319 snd_pcm_format_t format; |
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320 snd_pcm_uframes_t frames; |
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321 Uint16 test_format; |
0 | 322 |
323 /* Open the audio device */ | |
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324 /* Name of device should depend on # channels in spec */ |
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325 status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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326 |
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327 if ( status < 0 ) { |
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328 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); |
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329 return(-1); |
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330 } |
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331 |
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332 /* Figure out what the hardware is capable of */ |
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333 snd_pcm_hw_params_alloca(¶ms); |
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334 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params); |
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335 if ( status < 0 ) { |
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336 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); |
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337 ALSA_CloseAudio(this); |
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338 return(-1); |
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339 } |
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340 |
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341 /* SDL only uses interleaved sample output */ |
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342 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); |
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343 if ( status < 0 ) { |
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344 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); |
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345 ALSA_CloseAudio(this); |
0 | 346 return(-1); |
347 } | |
348 | |
349 /* Try for a closest match on audio format */ | |
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350 status = -1; |
0 | 351 for ( test_format = SDL_FirstAudioFormat(spec->format); |
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352 test_format && (status < 0); ) { |
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353 switch ( test_format ) { |
0 | 354 case AUDIO_U8: |
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355 format = SND_PCM_FORMAT_U8; |
0 | 356 break; |
357 case AUDIO_S8: | |
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358 format = SND_PCM_FORMAT_S8; |
0 | 359 break; |
360 case AUDIO_S16LSB: | |
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361 format = SND_PCM_FORMAT_S16_LE; |
0 | 362 break; |
363 case AUDIO_S16MSB: | |
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364 format = SND_PCM_FORMAT_S16_BE; |
0 | 365 break; |
366 case AUDIO_U16LSB: | |
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367 format = SND_PCM_FORMAT_U16_LE; |
0 | 368 break; |
369 case AUDIO_U16MSB: | |
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370 format = SND_PCM_FORMAT_U16_BE; |
0 | 371 break; |
372 default: | |
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373 format = 0; |
0 | 374 break; |
375 } | |
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376 if ( format != 0 ) { |
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377 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format); |
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378 } |
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379 if ( status < 0 ) { |
0 | 380 test_format = SDL_NextAudioFormat(); |
381 } | |
382 } | |
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383 if ( status < 0 ) { |
0 | 384 SDL_SetError("Couldn't find any hardware audio formats"); |
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385 ALSA_CloseAudio(this); |
0 | 386 return(-1); |
387 } | |
388 spec->format = test_format; | |
389 | |
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390 /* Set the number of channels */ |
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391 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels); |
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392 if ( status < 0 ) { |
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393 status = SDL_NAME(snd_pcm_hw_params_get_channels)(params); |
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394 if ( (status <= 0) || (status > 2) ) { |
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395 SDL_SetError("Couldn't set audio channels"); |
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396 ALSA_CloseAudio(this); |
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397 return(-1); |
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398 } |
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399 spec->channels = status; |
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400 } |
0 | 401 |
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402 /* Set the audio rate */ |
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403 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL); |
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404 if ( status < 0 ) { |
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405 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); |
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406 ALSA_CloseAudio(this); |
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407 return(-1); |
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408 } |
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409 spec->freq = status; |
0 | 410 |
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411 /* Set the buffer size, in samples */ |
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412 frames = spec->samples; |
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413 frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL); |
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414 spec->samples = frames; |
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415 SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL); |
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416 |
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417 /* "set" the hardware with the desired parameters */ |
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418 status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params); |
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419 if ( status < 0 ) { |
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420 SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(snd_strerror)(status)); |
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421 ALSA_CloseAudio(this); |
0 | 422 return(-1); |
423 } | |
424 | |
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425 /* Calculate the final parameters for this audio specification */ |
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426 SDL_CalculateAudioSpec(spec); |
0 | 427 |
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428 /* Allocate mixing buffer */ |
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429 mixlen = spec->size; |
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430 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); |
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431 if ( mixbuf == NULL ) { |
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432 ALSA_CloseAudio(this); |
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433 return(-1); |
0 | 434 } |
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435 memset(mixbuf, spec->silence, spec->size); |
0 | 436 |
437 /* Get the parent process id (we're the parent of the audio thread) */ | |
438 parent = getpid(); | |
439 | |
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440 /* Switch to blocking mode for playback */ |
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441 SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); |
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442 |
0 | 443 /* We're ready to rock and roll. :-) */ |
444 return(0); | |
445 } |