Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 865:92615154bb68
Date: Sun, 29 Feb 2004 15:14:22 +0200
From: Martin_Storsj
Subject: Dynamic loading of ALSA
I recently discovered that SDL can dynamically load ESD and aRts, and
made a patch which adds this same functionality to ALSA.
The update for configure.in isn't too good (it should e.g. look for
libasound.so in other directories than /usr/lib), because I'm not too
good at shellscripting and autoconf.
The reason for using dlfcn.h and dlopen instead of SDL_LoadLibrary and
SDL_LoadFunction is that libasound uses versioned symbols, and it is
necessary to load the correct version using dlvsym. This isn't probably
any real portability issue, because ALSA is linux-only.
author | Sam Lantinga <slouken@libsdl.org> |
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date | Tue, 02 Mar 2004 12:49:16 +0000 |
parents | b8d311d90021 |
children | c7c04f811994 |
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 #define __USE_GNU |
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46 #include <dlfcn.h> |
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47 #include "SDL_name.h" |
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48 #include "SDL_loadso.h" |
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49 #else |
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50 #define SDL_NAME(X) X |
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51 #endif |
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52 |
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53 |
0 | 54 /* The tag name used by ALSA audio */ |
55 #define DRIVER_NAME "alsa" | |
56 | |
354
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57 /* The default ALSA audio driver */ |
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58 #define DEFAULT_DEVICE "default" |
0 | 59 |
60 /* Audio driver functions */ | |
354
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61 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec); |
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62 static void ALSA_WaitAudio(_THIS); |
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63 static void ALSA_PlayAudio(_THIS); |
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64 static Uint8 *ALSA_GetAudioBuf(_THIS); |
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65 static void ALSA_CloseAudio(_THIS); |
0 | 66 |
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67 #ifdef ALSA_DYNAMIC |
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68 |
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69 static const char *alsa_library = ALSA_DYNAMIC; |
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70 static void *alsa_handle = NULL; |
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71 static int alsa_loaded = 0; |
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72 |
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73 static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); |
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74 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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75 static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm); |
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76 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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77 static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm); |
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78 static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm); |
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79 static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm); |
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80 static const char *(*SDL_NAME(snd_strerror))(int errnum); |
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81 static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void); |
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82 static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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83 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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84 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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85 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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86 static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params); |
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87 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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88 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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89 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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90 static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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91 static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock); |
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92 #define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof) |
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93 |
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94 static struct { |
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95 const char *name; |
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96 void **func; |
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97 } alsa_functions[] = { |
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98 { "snd_pcm_open", (void**)&SDL_NAME(snd_pcm_open) }, |
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99 { "snd_pcm_close", (void**)&SDL_NAME(snd_pcm_close) }, |
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100 { "snd_pcm_writei", (void**)&SDL_NAME(snd_pcm_writei) }, |
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101 { "snd_pcm_resume", (void**)&SDL_NAME(snd_pcm_resume) }, |
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102 { "snd_pcm_prepare", (void**)&SDL_NAME(snd_pcm_prepare) }, |
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103 { "snd_pcm_drain", (void**)&SDL_NAME(snd_pcm_drain) }, |
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104 { "snd_strerror", (void**)&SDL_NAME(snd_strerror) }, |
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105 { "snd_pcm_hw_params_sizeof", (void**)&SDL_NAME(snd_pcm_hw_params_sizeof) }, |
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106 { "snd_pcm_hw_params_any", (void**)&SDL_NAME(snd_pcm_hw_params_any) }, |
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107 { "snd_pcm_hw_params_set_access", (void**)&SDL_NAME(snd_pcm_hw_params_set_access) }, |
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108 { "snd_pcm_hw_params_set_format", (void**)&SDL_NAME(snd_pcm_hw_params_set_format) }, |
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109 { "snd_pcm_hw_params_set_channels", (void**)&SDL_NAME(snd_pcm_hw_params_set_channels) }, |
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110 { "snd_pcm_hw_params_get_channels", (void**)&SDL_NAME(snd_pcm_hw_params_get_channels) }, |
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111 { "snd_pcm_hw_params_set_rate_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_rate_near) }, |
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112 { "snd_pcm_hw_params_set_period_size_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) }, |
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113 { "snd_pcm_hw_params_set_periods_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_periods_near) }, |
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114 { "snd_pcm_hw_params", (void**)&SDL_NAME(snd_pcm_hw_params) }, |
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115 { "snd_pcm_nonblock", (void**)&SDL_NAME(snd_pcm_nonblock) }, |
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116 }; |
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117 |
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118 static void UnloadALSALibrary(void) { |
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119 if (alsa_loaded) { |
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120 /* SDL_UnloadObject(alsa_handle);*/ |
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121 dlclose(alsa_handle); |
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122 alsa_handle = NULL; |
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123 alsa_loaded = 0; |
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124 } |
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125 } |
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126 |
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127 static int LoadALSALibrary(void) { |
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128 int i, retval = -1; |
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129 |
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130 /* alsa_handle = SDL_LoadObject(alsa_library);*/ |
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131 alsa_handle = dlopen(alsa_library,RTLD_NOW); |
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132 if (alsa_handle) { |
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133 alsa_loaded = 1; |
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134 retval = 0; |
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135 for (i = 0; i < SDL_TABLESIZE(alsa_functions); i++) { |
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136 /* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/ |
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137 *alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9"); |
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138 if (!*alsa_functions[i].func) { |
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139 retval = -1; |
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140 UnloadALSALibrary(); |
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141 break; |
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142 } |
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143 } |
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144 } |
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145 return retval; |
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146 } |
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147 |
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148 #else |
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149 |
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150 static void UnloadALSALibrary(void) { |
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151 return; |
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152 } |
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153 |
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154 static int LoadALSALibrary(void) { |
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155 return 0; |
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156 } |
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157 |
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158 #endif /* ALSA_DYNAMIC */ |
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159 |
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160 static const char *get_audio_device() |
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161 { |
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162 const char *device; |
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163 |
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164 device = getenv("AUDIODEV"); /* Is there a standard variable name? */ |
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165 if ( device == NULL ) { |
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166 device = DEFAULT_DEVICE; |
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167 } |
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168 return device; |
0 | 169 } |
170 | |
171 /* Audio driver bootstrap functions */ | |
172 | |
173 static int Audio_Available(void) | |
174 { | |
175 int available; | |
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176 int status; |
0 | 177 snd_pcm_t *handle; |
178 | |
179 available = 0; | |
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180 if (LoadALSALibrary() < 0) { |
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181 return available; |
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182 } |
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183 status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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184 if ( status >= 0 ) { |
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185 available = 1; |
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186 SDL_NAME(snd_pcm_close)(handle); |
0 | 187 } |
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188 UnloadALSALibrary(); |
0 | 189 return(available); |
190 } | |
191 | |
192 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
193 { | |
194 free(device->hidden); | |
195 free(device); | |
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196 UnloadALSALibrary(); |
0 | 197 } |
198 | |
199 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
200 { | |
201 SDL_AudioDevice *this; | |
202 | |
203 /* Initialize all variables that we clean on shutdown */ | |
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204 LoadALSALibrary(); |
0 | 205 this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice)); |
206 if ( this ) { | |
207 memset(this, 0, (sizeof *this)); | |
208 this->hidden = (struct SDL_PrivateAudioData *) | |
209 malloc((sizeof *this->hidden)); | |
210 } | |
211 if ( (this == NULL) || (this->hidden == NULL) ) { | |
212 SDL_OutOfMemory(); | |
213 if ( this ) { | |
214 free(this); | |
215 } | |
216 return(0); | |
217 } | |
218 memset(this->hidden, 0, (sizeof *this->hidden)); | |
219 | |
220 /* Set the function pointers */ | |
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221 this->OpenAudio = ALSA_OpenAudio; |
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222 this->WaitAudio = ALSA_WaitAudio; |
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223 this->PlayAudio = ALSA_PlayAudio; |
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224 this->GetAudioBuf = ALSA_GetAudioBuf; |
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225 this->CloseAudio = ALSA_CloseAudio; |
0 | 226 |
227 this->free = Audio_DeleteDevice; | |
228 | |
229 return this; | |
230 } | |
231 | |
232 AudioBootStrap ALSA_bootstrap = { | |
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233 DRIVER_NAME, "ALSA 0.9 PCM audio", |
0 | 234 Audio_Available, Audio_CreateDevice |
235 }; | |
236 | |
237 /* This function waits until it is possible to write a full sound buffer */ | |
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238 static void ALSA_WaitAudio(_THIS) |
0 | 239 { |
240 /* Check to see if the thread-parent process is still alive */ | |
241 { static int cnt = 0; | |
242 /* Note that this only works with thread implementations | |
243 that use a different process id for each thread. | |
244 */ | |
245 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ | |
246 if ( kill(parent, 0) < 0 ) { | |
247 this->enabled = 0; | |
248 } | |
249 } | |
250 } | |
251 } | |
252 | |
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253 static void ALSA_PlayAudio(_THIS) |
0 | 254 { |
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255 int status; |
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256 int sample_len; |
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257 signed short *sample_buf; |
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258 |
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259 sample_len = this->spec.samples; |
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260 sample_buf = (signed short *)mixbuf; |
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261 while ( sample_len > 0 ) { |
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262 status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len); |
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263 if ( status < 0 ) { |
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264 if ( status == -EAGAIN ) { |
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265 SDL_Delay(1); |
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266 continue; |
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267 } |
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268 if ( status == -ESTRPIPE ) { |
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269 do { |
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270 SDL_Delay(1); |
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271 status = SDL_NAME(snd_pcm_resume)(pcm_handle); |
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272 } while ( status == -EAGAIN ); |
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273 } |
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274 if ( status < 0 ) { |
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275 status = SDL_NAME(snd_pcm_prepare)(pcm_handle); |
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276 } |
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277 if ( status < 0 ) { |
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278 /* Hmm, not much we can do - abort */ |
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279 this->enabled = 0; |
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280 return; |
0 | 281 } |
356
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282 continue; |
0 | 283 } |
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284 sample_buf += status * this->spec.channels; |
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285 sample_len -= status; |
0 | 286 } |
287 } | |
288 | |
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289 static Uint8 *ALSA_GetAudioBuf(_THIS) |
0 | 290 { |
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291 return(mixbuf); |
0 | 292 } |
293 | |
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294 static void ALSA_CloseAudio(_THIS) |
0 | 295 { |
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296 if ( mixbuf != NULL ) { |
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297 SDL_FreeAudioMem(mixbuf); |
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298 mixbuf = NULL; |
0 | 299 } |
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300 if ( pcm_handle ) { |
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301 SDL_NAME(snd_pcm_drain)(pcm_handle); |
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302 SDL_NAME(snd_pcm_close)(pcm_handle); |
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303 pcm_handle = NULL; |
0 | 304 } |
305 } | |
306 | |
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307 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec) |
0 | 308 { |
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309 int status; |
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310 snd_pcm_hw_params_t *params; |
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311 snd_pcm_format_t format; |
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312 snd_pcm_uframes_t frames; |
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313 Uint16 test_format; |
0 | 314 |
315 /* Open the audio device */ | |
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316 status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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317 if ( status < 0 ) { |
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318 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); |
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319 return(-1); |
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320 } |
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321 |
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322 /* Figure out what the hardware is capable of */ |
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323 snd_pcm_hw_params_alloca(¶ms); |
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324 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params); |
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325 if ( status < 0 ) { |
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326 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); |
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327 ALSA_CloseAudio(this); |
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328 return(-1); |
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329 } |
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330 |
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331 /* SDL only uses interleaved sample output */ |
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332 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); |
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333 if ( status < 0 ) { |
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334 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); |
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335 ALSA_CloseAudio(this); |
0 | 336 return(-1); |
337 } | |
338 | |
339 /* Try for a closest match on audio format */ | |
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340 status = -1; |
0 | 341 for ( test_format = SDL_FirstAudioFormat(spec->format); |
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342 test_format && (status < 0); ) { |
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343 switch ( test_format ) { |
0 | 344 case AUDIO_U8: |
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345 format = SND_PCM_FORMAT_U8; |
0 | 346 break; |
347 case AUDIO_S8: | |
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348 format = SND_PCM_FORMAT_S8; |
0 | 349 break; |
350 case AUDIO_S16LSB: | |
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351 format = SND_PCM_FORMAT_S16_LE; |
0 | 352 break; |
353 case AUDIO_S16MSB: | |
354
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354 format = SND_PCM_FORMAT_S16_BE; |
0 | 355 break; |
356 case AUDIO_U16LSB: | |
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357 format = SND_PCM_FORMAT_U16_LE; |
0 | 358 break; |
359 case AUDIO_U16MSB: | |
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360 format = SND_PCM_FORMAT_U16_BE; |
0 | 361 break; |
362 default: | |
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363 format = 0; |
0 | 364 break; |
365 } | |
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366 if ( format != 0 ) { |
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367 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format); |
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368 } |
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369 if ( status < 0 ) { |
0 | 370 test_format = SDL_NextAudioFormat(); |
371 } | |
372 } | |
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373 if ( status < 0 ) { |
0 | 374 SDL_SetError("Couldn't find any hardware audio formats"); |
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375 ALSA_CloseAudio(this); |
0 | 376 return(-1); |
377 } | |
378 spec->format = test_format; | |
379 | |
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380 /* Set the number of channels */ |
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381 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels); |
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382 if ( status < 0 ) { |
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383 status = SDL_NAME(snd_pcm_hw_params_get_channels)(params); |
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384 if ( (status <= 0) || (status > 2) ) { |
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385 SDL_SetError("Couldn't set audio channels"); |
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386 ALSA_CloseAudio(this); |
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387 return(-1); |
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388 } |
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389 spec->channels = status; |
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390 } |
0 | 391 |
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392 /* Set the audio rate */ |
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393 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL); |
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394 if ( status < 0 ) { |
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395 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); |
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396 ALSA_CloseAudio(this); |
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397 return(-1); |
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398 } |
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399 spec->freq = status; |
0 | 400 |
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401 /* Set the buffer size, in samples */ |
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402 frames = spec->samples; |
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403 frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL); |
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404 spec->samples = frames; |
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405 SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL); |
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406 |
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407 /* "set" the hardware with the desired parameters */ |
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408 status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params); |
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409 if ( status < 0 ) { |
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410 SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(snd_strerror)(status)); |
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411 ALSA_CloseAudio(this); |
0 | 412 return(-1); |
413 } | |
414 | |
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415 /* Calculate the final parameters for this audio specification */ |
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416 SDL_CalculateAudioSpec(spec); |
0 | 417 |
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418 /* Allocate mixing buffer */ |
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419 mixlen = spec->size; |
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420 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); |
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421 if ( mixbuf == NULL ) { |
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422 ALSA_CloseAudio(this); |
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423 return(-1); |
0 | 424 } |
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425 memset(mixbuf, spec->silence, spec->size); |
0 | 426 |
427 /* Get the parent process id (we're the parent of the audio thread) */ | |
428 parent = getpid(); | |
429 | |
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430 /* Switch to blocking mode for playback */ |
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431 SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); |
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432 |
0 | 433 /* We're ready to rock and roll. :-) */ |
434 return(0); | |
435 } |