Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 914:bbf8dcc8aed6
Date: Wed, 23 Jun 2004 17:05:33 -0400
From: Chris Nelson
Subject: [SDL] [Patch] WiseGroup MP-8800 / MP-8866 (PS2 Joystick)
In the current cvs version, SDL doesn't handle these Playstation2
controller => USB adapters correctly, in linux.
It will always assume that the maximum number of joysticks (2 in the
case of the MP-8866, 4 in the case of the 8800) are plugged in. This is
bad not only because it allows SDL to exaggerate the number of logical
joysticks, but primarily because the joystick axes are mapped
incorrectly, all over the place, such that the devices are effectively
unusable unless you have the maximum number of joysticks plugged in.
My changes to src/joystick/linux/SDL_sysjoystick.c build on another's
previous work (which was a special case for this very joystick,
actually), and fix both of these problems, as well as making the current
code a little more general, to allow for others to more easily drop in
code for quirky joysticks such as these.
I've tested this code under 2.6.7 as well as 2.4.24... Both work as
advertised (provided you load the JOYDEV linux code as a module,
otherwise they won't work at all, new code or old, but that's another
issue entirely).
Though this sounds horribly formal, you have my permission to distribute
all of my work on this issue under the LGPL. So there.
author | Sam Lantinga <slouken@libsdl.org> |
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date | Sun, 25 Jul 2004 18:31:50 +0000 |
parents | 92615154bb68 |
children | c7c04f811994 |
rev | line source |
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0 | 1 /* |
2 SDL - Simple DirectMedia Layer | |
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3 Copyright (C) 1997-2004 Sam Lantinga |
0 | 4 |
5 This library is free software; you can redistribute it and/or | |
6 modify it under the terms of the GNU Library General Public | |
7 License as published by the Free Software Foundation; either | |
8 version 2 of the License, or (at your option) any later version. | |
9 | |
10 This library is distributed in the hope that it will be useful, | |
11 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 Library General Public License for more details. | |
14 | |
15 You should have received a copy of the GNU Library General Public | |
16 License along with this library; if not, write to the Free | |
17 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
18 | |
19 Sam Lantinga | |
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20 slouken@libsdl.org |
0 | 21 */ |
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: | |
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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 } |