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>
date Tue, 02 Mar 2004 12:49:16 +0000
parents b8d311d90021
children c7c04f811994
rev   line source
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1 /*
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2 SDL - Simple DirectMedia Layer
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3 Copyright (C) 1997-2004 Sam Lantinga
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4
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5 This library is free software; you can redistribute it and/or
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6 modify it under the terms of the GNU Library General Public
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7 License as published by the Free Software Foundation; either
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8 version 2 of the License, or (at your option) any later version.
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9
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10 This library is distributed in the hope that it will be useful,
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11 but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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13 Library General Public License for more details.
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14
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15 You should have received a copy of the GNU Library General Public
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16 License along with this library; if not, write to the Free
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17 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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18
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19 Sam Lantinga
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20 slouken@libsdl.org
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21 */
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22
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23
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24
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25 /* Allow access to a raw mixing buffer */
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26
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27 #include <stdlib.h>
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28 #include <stdio.h>
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29 #include <string.h>
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30 #include <errno.h>
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31 #include <unistd.h>
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32 #include <fcntl.h>
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33 #include <signal.h>
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34 #include <sys/types.h>
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35 #include <sys/time.h>
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36
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37 #include "SDL_audio.h"
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38 #include "SDL_error.h"
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39 #include "SDL_audiomem.h"
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40 #include "SDL_audio_c.h"
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41 #include "SDL_timer.h"
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42 #include "SDL_alsa_audio.h"
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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
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54 /* The tag name used by ALSA audio */
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55 #define DRIVER_NAME "alsa"
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56
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57 /* The default ALSA audio driver */
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58 #define DEFAULT_DEVICE "default"
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59
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60 /* Audio driver functions */
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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);
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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
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169 }
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170
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171 /* Audio driver bootstrap functions */
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172
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173 static int Audio_Available(void)
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174 {
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175 int available;
354
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176 int status;
0
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177 snd_pcm_t *handle;
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178
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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);
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187 }
865
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188 UnloadALSALibrary();
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189 return(available);
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190 }
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191
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192 static void Audio_DeleteDevice(SDL_AudioDevice *device)
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193 {
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194 free(device->hidden);
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195 free(device);
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196 UnloadALSALibrary();
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197 }
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198
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199 static SDL_AudioDevice *Audio_CreateDevice(int devindex)
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200 {
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201 SDL_AudioDevice *this;
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202
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203 /* Initialize all variables that we clean on shutdown */
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204 LoadALSALibrary();
0
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205 this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
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206 if ( this ) {
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207 memset(this, 0, (sizeof *this));
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208 this->hidden = (struct SDL_PrivateAudioData *)
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209 malloc((sizeof *this->hidden));
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210 }
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211 if ( (this == NULL) || (this->hidden == NULL) ) {
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212 SDL_OutOfMemory();
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213 if ( this ) {
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214 free(this);
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215 }
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216 return(0);
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217 }
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218 memset(this->hidden, 0, (sizeof *this->hidden));
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219
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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
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226
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227 this->free = Audio_DeleteDevice;
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228
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229 return this;
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230 }
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231
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232 AudioBootStrap ALSA_bootstrap = {
354
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233 DRIVER_NAME, "ALSA 0.9 PCM audio",
0
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234 Audio_Available, Audio_CreateDevice
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235 };
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236
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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
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239 {
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240 /* Check to see if the thread-parent process is still alive */
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241 { static int cnt = 0;
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242 /* Note that this only works with thread implementations
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243 that use a different process id for each thread.
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244 */
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245 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
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246 if ( kill(parent, 0) < 0 ) {
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247 this->enabled = 0;
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248 }
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249 }
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250 }
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251 }
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252
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253 static void ALSA_PlayAudio(_THIS)
0
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254 {
354
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diff changeset
255 int status;
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256 int sample_len;
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257 signed short *sample_buf;
765
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258
354
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diff changeset
259 sample_len = this->spec.samples;
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260 sample_buf = (signed short *)mixbuf;
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diff changeset
261 while ( sample_len > 0 ) {
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262 status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len);
354
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diff changeset
263 if ( status < 0 ) {
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diff changeset
264 if ( status == -EAGAIN ) {
765
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diff changeset
265 SDL_Delay(1);
354
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diff changeset
266 continue;
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diff changeset
267 }
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diff changeset
268 if ( status == -ESTRPIPE ) {
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diff changeset
269 do {
765
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270 SDL_Delay(1);
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271 status = SDL_NAME(snd_pcm_resume)(pcm_handle);
354
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272 } while ( status == -EAGAIN );
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diff changeset
273 }
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diff changeset
274 if ( status < 0 ) {
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275 status = SDL_NAME(snd_pcm_prepare)(pcm_handle);
354
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diff changeset
276 }
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diff changeset
277 if ( status < 0 ) {
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diff changeset
278 /* Hmm, not much we can do - abort */
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diff changeset
279 this->enabled = 0;
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diff changeset
280 return;
0
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diff changeset
281 }
356
a1e54d1ba16f *** empty log message ***
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parents: 354
diff changeset
282 continue;
0
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diff changeset
283 }
354
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diff changeset
284 sample_buf += status * this->spec.channels;
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diff changeset
285 sample_len -= status;
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286 }
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287 }
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288
354
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diff changeset
289 static Uint8 *ALSA_GetAudioBuf(_THIS)
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290 {
354
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diff changeset
291 return(mixbuf);
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292 }
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293
354
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diff changeset
294 static void ALSA_CloseAudio(_THIS)
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295 {
354
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diff changeset
296 if ( mixbuf != NULL ) {
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diff changeset
297 SDL_FreeAudioMem(mixbuf);
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diff changeset
298 mixbuf = NULL;
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diff changeset
299 }
354
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diff changeset
300 if ( pcm_handle ) {
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diff changeset
301 SDL_NAME(snd_pcm_drain)(pcm_handle);
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302 SDL_NAME(snd_pcm_close)(pcm_handle);
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diff changeset
303 pcm_handle = NULL;
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304 }
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305 }
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306
354
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diff changeset
307 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
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308 {
354
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diff changeset
309 int status;
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diff changeset
310 snd_pcm_hw_params_t *params;
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diff changeset
311 snd_pcm_format_t format;
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diff changeset
312 snd_pcm_uframes_t frames;
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diff changeset
313 Uint16 test_format;
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314
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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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diff changeset
317 if ( status < 0 ) {
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318 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));
354
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diff changeset
319 return(-1);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
320 }
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diff changeset
321
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diff changeset
322 /* Figure out what the hardware is capable of */
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diff changeset
323 snd_pcm_hw_params_alloca(&params);
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324 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params);
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diff changeset
325 if ( status < 0 ) {
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diff changeset
326 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));
354
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diff changeset
327 ALSA_CloseAudio(this);
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diff changeset
328 return(-1);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
329 }
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
330
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
331 /* SDL only uses interleaved sample output */
865
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diff changeset
332 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
354
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diff changeset
333 if ( status < 0 ) {
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diff changeset
334 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));
354
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diff changeset
335 ALSA_CloseAudio(this);
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336 return(-1);
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337 }
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338
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diff changeset
339 /* Try for a closest match on audio format */
354
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diff changeset
340 status = -1;
0
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341 for ( test_format = SDL_FirstAudioFormat(spec->format);
354
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diff changeset
342 test_format && (status < 0); ) {
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
343 switch ( test_format ) {
0
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344 case AUDIO_U8:
354
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diff changeset
345 format = SND_PCM_FORMAT_U8;
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346 break;
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347 case AUDIO_S8:
354
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diff changeset
348 format = SND_PCM_FORMAT_S8;
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349 break;
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350 case AUDIO_S16LSB:
354
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diff changeset
351 format = SND_PCM_FORMAT_S16_LE;
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352 break;
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353 case AUDIO_S16MSB:
354
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diff changeset
354 format = SND_PCM_FORMAT_S16_BE;
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355 break;
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356 case AUDIO_U16LSB:
354
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diff changeset
357 format = SND_PCM_FORMAT_U16_LE;
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358 break;
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359 case AUDIO_U16MSB:
354
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diff changeset
360 format = SND_PCM_FORMAT_U16_BE;
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361 break;
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362 default:
354
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diff changeset
363 format = 0;
0
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diff changeset
364 break;
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diff changeset
365 }
354
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diff changeset
366 if ( format != 0 ) {
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367 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format);
354
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diff changeset
368 }
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diff changeset
369 if ( status < 0 ) {
0
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diff changeset
370 test_format = SDL_NextAudioFormat();
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diff changeset
371 }
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diff changeset
372 }
354
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diff changeset
373 if ( status < 0 ) {
0
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diff changeset
374 SDL_SetError("Couldn't find any hardware audio formats");
354
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diff changeset
375 ALSA_CloseAudio(this);
0
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diff changeset
376 return(-1);
74212992fb08 Initial revision
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parents:
diff changeset
377 }
74212992fb08 Initial revision
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diff changeset
378 spec->format = test_format;
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parents:
diff changeset
379
354
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diff changeset
380 /* Set the number of channels */
865
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diff changeset
381 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels);
354
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diff changeset
382 if ( status < 0 ) {
865
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diff changeset
383 status = SDL_NAME(snd_pcm_hw_params_get_channels)(params);
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
384 if ( (status <= 0) || (status > 2) ) {
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
385 SDL_SetError("Couldn't set audio channels");
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
386 ALSA_CloseAudio(this);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
387 return(-1);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
388 }
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
389 spec->channels = status;
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
390 }
0
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diff changeset
391
354
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diff changeset
392 /* Set the audio rate */
865
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parents: 769
diff changeset
393 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL);
354
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parents: 297
diff changeset
394 if ( status < 0 ) {
865
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diff changeset
395 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
354
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diff changeset
396 ALSA_CloseAudio(this);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
397 return(-1);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
398 }
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
399 spec->freq = status;
0
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parents:
diff changeset
400
354
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diff changeset
401 /* Set the buffer size, in samples */
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
402 frames = spec->samples;
865
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parents: 769
diff changeset
403 frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL);
354
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diff changeset
404 spec->samples = frames;
865
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diff changeset
405 SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL);
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
406
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
407 /* "set" the hardware with the desired parameters */
865
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parents: 769
diff changeset
408 status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params);
354
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parents: 297
diff changeset
409 if ( status < 0 ) {
865
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diff changeset
410 SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(snd_strerror)(status));
354
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diff changeset
411 ALSA_CloseAudio(this);
0
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parents:
diff changeset
412 return(-1);
74212992fb08 Initial revision
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parents:
diff changeset
413 }
74212992fb08 Initial revision
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parents:
diff changeset
414
354
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diff changeset
415 /* Calculate the final parameters for this audio specification */
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
416 SDL_CalculateAudioSpec(spec);
0
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parents:
diff changeset
417
354
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diff changeset
418 /* Allocate mixing buffer */
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
419 mixlen = spec->size;
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
420 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
421 if ( mixbuf == NULL ) {
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
422 ALSA_CloseAudio(this);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
423 return(-1);
0
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parents:
diff changeset
424 }
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
425 memset(mixbuf, spec->silence, spec->size);
0
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parents:
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426
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427 /* Get the parent process id (we're the parent of the audio thread) */
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428 parent = getpid();
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429
765
4c2ba6161939 Editors Note: The original patch was modified to use SDL_Delay() instead of
Sam Lantinga <slouken@libsdl.org>
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430 /* Switch to blocking mode for playback */
865
92615154bb68 Date: Sun, 29 Feb 2004 15:14:22 +0200
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parents: 769
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431 SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);
765
4c2ba6161939 Editors Note: The original patch was modified to use SDL_Delay() instead of
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parents: 547
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432
0
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433 /* We're ready to rock and roll. :-) */
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434 return(0);
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435 }