Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 4333:55717a755897 SDL-1.2
Switched from setting the period size and count to setting the buffer size directly, based on feedback from the ALSA development list:
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022267.html
This has the nice side effect of reducing latency on my SBLive! card.
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Tue, 13 Oct 2009 09:56:15 +0000 |
| parents | 67e799ffcadf |
| children | afadcd7d2319 |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
| 4159 | 3 Copyright (C) 1997-2009 Sam Lantinga |
| 0 | 4 |
| 5 This library is free software; you can redistribute it and/or | |
| 6 modify it under the terms of the GNU Library General Public | |
| 7 License as published by the Free Software Foundation; either | |
| 8 version 2 of the License, or (at your option) any later version. | |
| 9 | |
| 10 This library is distributed in the hope that it will be useful, | |
| 11 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
| 12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
| 13 Library General Public License for more details. | |
| 14 | |
| 15 You should have received a copy of the GNU Library General Public | |
| 16 License along with this library; if not, write to the Free | |
| 17 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
| 18 | |
| 19 Sam Lantinga | |
|
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20 slouken@libsdl.org |
| 0 | 21 */ |
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22 #include "SDL_config.h" |
| 0 | 23 |
| 24 /* Allow access to a raw mixing buffer */ | |
| 25 | |
| 26 #include <sys/types.h> | |
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27 #include <signal.h> /* For kill() */ |
| 0 | 28 |
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29 #include "SDL_timer.h" |
| 0 | 30 #include "SDL_audio.h" |
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31 #include "../SDL_audiomem.h" |
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32 #include "../SDL_audio_c.h" |
| 0 | 33 #include "SDL_alsa_audio.h" |
| 34 | |
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35 #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
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36 #include "SDL_name.h" |
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37 #include "SDL_loadso.h" |
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38 #else |
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39 #define SDL_NAME(X) X |
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40 #endif |
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41 |
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42 |
| 0 | 43 /* The tag name used by ALSA audio */ |
| 44 #define DRIVER_NAME "alsa" | |
| 45 | |
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46 /* The default ALSA audio driver */ |
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47 #define DEFAULT_DEVICE "default" |
| 0 | 48 |
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49 /* Whether we should set the buffer size or the period size */ |
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50 /*#define SET_PERIOD_SIZE*/ |
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51 /*#define DEBUG_PERIOD_SIZE*/ |
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55717a755897
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parents:
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52 |
| 0 | 53 /* Audio driver functions */ |
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54 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec); |
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55 static void ALSA_WaitAudio(_THIS); |
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56 static void ALSA_PlayAudio(_THIS); |
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57 static Uint8 *ALSA_GetAudioBuf(_THIS); |
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58 static void ALSA_CloseAudio(_THIS); |
| 0 | 59 |
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60 #ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC |
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61 |
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62 static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC; |
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63 static void *alsa_handle = NULL; |
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64 static int alsa_loaded = 0; |
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65 |
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66 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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67 static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm); |
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68 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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69 static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm); |
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70 static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm); |
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71 static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm); |
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72 static const char *(*SDL_NAME(snd_strerror))(int errnum); |
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73 static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void); |
| 1552 | 74 static size_t (*SDL_NAME(snd_pcm_sw_params_sizeof))(void); |
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75 static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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76 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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77 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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78 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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79 static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params, unsigned int *val); |
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80 static 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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81 static int (*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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82 static int (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir); |
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83 static 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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84 static int (*SDL_NAME(snd_pcm_hw_params_get_periods))(const snd_pcm_hw_params_t *params, unsigned int *val, int *dir); |
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Switched from setting the period size and count to setting the buffer size directly, based on feedback from the ALSA development list:
Sam Lantinga <slouken@libsdl.org>
parents:
4332
diff
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85 static int (*SDL_NAME(snd_pcm_hw_params_set_buffer_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val); |
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86 static int (*SDL_NAME(snd_pcm_hw_params_get_buffer_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val); |
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87 static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
| 1552 | 88 /* |
| 89 */ | |
| 90 static int (*SDL_NAME(snd_pcm_sw_params_current))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams); | |
| 91 static int (*SDL_NAME(snd_pcm_sw_params_set_start_threshold))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val); | |
| 92 static int (*SDL_NAME(snd_pcm_sw_params))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params); | |
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93 static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock); |
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94 #define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof) |
| 1552 | 95 #define snd_pcm_sw_params_sizeof SDL_NAME(snd_pcm_sw_params_sizeof) |
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96 |
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97 /* cast funcs to char* first, to please GCC's strict aliasing rules. */ |
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98 static struct { |
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99 const char *name; |
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100 void **func; |
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101 } alsa_functions[] = { |
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102 { "snd_pcm_open", (void**)(char*)&SDL_NAME(snd_pcm_open) }, |
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103 { "snd_pcm_close", (void**)(char*)&SDL_NAME(snd_pcm_close) }, |
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104 { "snd_pcm_writei", (void**)(char*)&SDL_NAME(snd_pcm_writei) }, |
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105 { "snd_pcm_resume", (void**)(char*)&SDL_NAME(snd_pcm_resume) }, |
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106 { "snd_pcm_prepare", (void**)(char*)&SDL_NAME(snd_pcm_prepare) }, |
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107 { "snd_pcm_drain", (void**)(char*)&SDL_NAME(snd_pcm_drain) }, |
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108 { "snd_strerror", (void**)(char*)&SDL_NAME(snd_strerror) }, |
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109 { "snd_pcm_hw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof) }, |
| 1552 | 110 { "snd_pcm_sw_params_sizeof", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_sizeof) }, |
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111 { "snd_pcm_hw_params_any", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_any) }, |
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112 { "snd_pcm_hw_params_set_access", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access) }, |
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113 { "snd_pcm_hw_params_set_format", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format) }, |
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Ryan C. Gordon <icculus@icculus.org>
parents:
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114 { "snd_pcm_hw_params_set_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels) }, |
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Ryan C. Gordon <icculus@icculus.org>
parents:
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115 { "snd_pcm_hw_params_get_channels", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels) }, |
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Ryan C. Gordon <icculus@icculus.org>
parents:
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116 { "snd_pcm_hw_params_set_rate_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near) }, |
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Ryan C. Gordon <icculus@icculus.org>
parents:
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117 { "snd_pcm_hw_params_set_period_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) }, |
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118 { "snd_pcm_hw_params_get_period_size", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_period_size) }, |
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119 { "snd_pcm_hw_params_set_periods_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near) }, |
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120 { "snd_pcm_hw_params_get_periods", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_periods) }, |
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Switched from setting the period size and count to setting the buffer size directly, based on feedback from the ALSA development list:
Sam Lantinga <slouken@libsdl.org>
parents:
4332
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121 { "snd_pcm_hw_params_set_buffer_size_near", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_buffer_size_near) }, |
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122 { "snd_pcm_hw_params_get_buffer_size", (void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_buffer_size) }, |
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1161
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Placate gcc's strict aliasing rules with an extra cast.
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123 { "snd_pcm_hw_params", (void**)(char*)&SDL_NAME(snd_pcm_hw_params) }, |
| 1552 | 124 { "snd_pcm_sw_params_current", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_current) }, |
| 125 { "snd_pcm_sw_params_set_start_threshold", (void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_start_threshold) }, | |
| 126 { "snd_pcm_sw_params", (void**)(char*)&SDL_NAME(snd_pcm_sw_params) }, | |
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127 { "snd_pcm_nonblock", (void**)(char*)&SDL_NAME(snd_pcm_nonblock) }, |
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128 }; |
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129 |
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130 static void UnloadALSALibrary(void) { |
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131 if (alsa_loaded) { |
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132 SDL_UnloadObject(alsa_handle); |
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133 alsa_handle = NULL; |
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134 alsa_loaded = 0; |
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135 } |
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136 } |
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137 |
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138 static int LoadALSALibrary(void) { |
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139 int i, retval = -1; |
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140 |
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141 alsa_handle = SDL_LoadObject(alsa_library); |
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142 if (alsa_handle) { |
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143 alsa_loaded = 1; |
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144 retval = 0; |
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145 for (i = 0; i < SDL_arraysize(alsa_functions); i++) { |
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146 *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name); |
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147 if (!*alsa_functions[i].func) { |
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148 retval = -1; |
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149 UnloadALSALibrary(); |
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150 break; |
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151 } |
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152 } |
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153 } |
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154 return retval; |
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155 } |
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156 |
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157 #else |
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158 |
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159 static void UnloadALSALibrary(void) { |
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160 return; |
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161 } |
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162 |
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163 static int LoadALSALibrary(void) { |
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164 return 0; |
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165 } |
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166 |
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167 #endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */ |
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168 |
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169 static const char *get_audio_device(int channels) |
|
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170 { |
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171 const char *device; |
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172 |
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173 device = SDL_getenv("AUDIODEV"); /* Is there a standard variable name? */ |
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174 if ( device == NULL ) { |
|
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175 if (channels == 6) device = "surround51"; |
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176 else if (channels == 4) device = "surround40"; |
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177 else device = DEFAULT_DEVICE; |
|
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178 } |
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179 return device; |
| 0 | 180 } |
| 181 | |
| 182 /* Audio driver bootstrap functions */ | |
| 183 | |
| 184 static int Audio_Available(void) | |
| 185 { | |
| 186 int available; | |
|
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187 int status; |
| 0 | 188 snd_pcm_t *handle; |
| 189 | |
| 190 available = 0; | |
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191 if (LoadALSALibrary() < 0) { |
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192 return available; |
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193 } |
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194 status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
|
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195 if ( status >= 0 ) { |
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196 available = 1; |
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197 SDL_NAME(snd_pcm_close)(handle); |
| 0 | 198 } |
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199 UnloadALSALibrary(); |
| 0 | 200 return(available); |
| 201 } | |
| 202 | |
| 203 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
| 204 { | |
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205 SDL_free(device->hidden); |
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206 SDL_free(device); |
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207 UnloadALSALibrary(); |
| 0 | 208 } |
| 209 | |
| 210 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
| 211 { | |
| 212 SDL_AudioDevice *this; | |
| 213 | |
| 214 /* Initialize all variables that we clean on shutdown */ | |
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215 LoadALSALibrary(); |
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216 this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice)); |
| 0 | 217 if ( this ) { |
|
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218 SDL_memset(this, 0, (sizeof *this)); |
| 0 | 219 this->hidden = (struct SDL_PrivateAudioData *) |
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220 SDL_malloc((sizeof *this->hidden)); |
| 0 | 221 } |
| 222 if ( (this == NULL) || (this->hidden == NULL) ) { | |
| 223 SDL_OutOfMemory(); | |
| 224 if ( this ) { | |
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225 SDL_free(this); |
| 0 | 226 } |
| 227 return(0); | |
| 228 } | |
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229 SDL_memset(this->hidden, 0, (sizeof *this->hidden)); |
| 0 | 230 |
| 231 /* Set the function pointers */ | |
|
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232 this->OpenAudio = ALSA_OpenAudio; |
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233 this->WaitAudio = ALSA_WaitAudio; |
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234 this->PlayAudio = ALSA_PlayAudio; |
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235 this->GetAudioBuf = ALSA_GetAudioBuf; |
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236 this->CloseAudio = ALSA_CloseAudio; |
| 0 | 237 |
| 238 this->free = Audio_DeleteDevice; | |
| 239 | |
| 240 return this; | |
| 241 } | |
| 242 | |
| 243 AudioBootStrap ALSA_bootstrap = { | |
|
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244 DRIVER_NAME, "ALSA 0.9 PCM audio", |
| 0 | 245 Audio_Available, Audio_CreateDevice |
| 246 }; | |
| 247 | |
| 248 /* This function waits until it is possible to write a full sound buffer */ | |
|
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249 static void ALSA_WaitAudio(_THIS) |
| 0 | 250 { |
| 251 /* Check to see if the thread-parent process is still alive */ | |
| 252 { static int cnt = 0; | |
| 253 /* Note that this only works with thread implementations | |
| 254 that use a different process id for each thread. | |
| 255 */ | |
| 256 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ | |
| 257 if ( kill(parent, 0) < 0 ) { | |
| 258 this->enabled = 0; | |
| 259 } | |
| 260 } | |
| 261 } | |
| 262 } | |
| 263 | |
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264 |
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265 /* |
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266 * http://bugzilla.libsdl.org/show_bug.cgi?id=110 |
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267 * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE |
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268 * and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR" |
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269 */ |
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270 #define SWIZ6(T) \ |
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271 T *ptr = (T *) mixbuf; \ |
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272 const Uint32 count = (this->spec.samples / 6); \ |
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273 Uint32 i; \ |
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274 for (i = 0; i < count; i++, ptr += 6) { \ |
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275 T tmp; \ |
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276 tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \ |
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277 tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \ |
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278 } |
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279 |
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280 static __inline__ void swizzle_alsa_channels_6_64bit(_THIS) { SWIZ6(Uint64); } |
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281 static __inline__ void swizzle_alsa_channels_6_32bit(_THIS) { SWIZ6(Uint32); } |
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282 static __inline__ void swizzle_alsa_channels_6_16bit(_THIS) { SWIZ6(Uint16); } |
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283 static __inline__ void swizzle_alsa_channels_6_8bit(_THIS) { SWIZ6(Uint8); } |
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284 |
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285 #undef SWIZ6 |
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286 |
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287 |
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288 /* |
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289 * Called right before feeding this->mixbuf to the hardware. Swizzle channels |
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290 * from Windows/Mac order to the format alsalib will want. |
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291 */ |
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292 static __inline__ void swizzle_alsa_channels(_THIS) |
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293 { |
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294 if (this->spec.channels == 6) { |
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295 const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */ |
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296 if (fmtsize == 16) |
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297 swizzle_alsa_channels_6_16bit(this); |
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298 else if (fmtsize == 8) |
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299 swizzle_alsa_channels_6_8bit(this); |
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300 else if (fmtsize == 32) |
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301 swizzle_alsa_channels_6_32bit(this); |
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302 else if (fmtsize == 64) |
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303 swizzle_alsa_channels_6_64bit(this); |
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304 } |
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305 |
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306 /* !!! FIXME: update this for 7.1 if needed, later. */ |
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307 } |
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308 |
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309 |
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310 static void ALSA_PlayAudio(_THIS) |
| 0 | 311 { |
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312 int status; |
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313 snd_pcm_uframes_t frames_left; |
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314 const Uint8 *sample_buf = (const Uint8 *) mixbuf; |
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315 const int sample_size = ((int) (this->spec.format & 0xFF)) / 8; |
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316 |
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317 swizzle_alsa_channels(this); |
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318 |
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319 frames_left = ((snd_pcm_uframes_t) this->spec.samples); |
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320 |
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321 while ( frames_left > 0 ) { |
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322 status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, frames_left); |
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323 if ( status < 0 ) { |
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324 if ( status == -EAGAIN ) { |
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325 SDL_Delay(1); |
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326 continue; |
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327 } |
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328 if ( status == -ESTRPIPE ) { |
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329 do { |
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330 SDL_Delay(1); |
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331 status = SDL_NAME(snd_pcm_resume)(pcm_handle); |
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332 } while ( status == -EAGAIN ); |
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333 } |
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334 if ( status < 0 ) { |
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335 status = SDL_NAME(snd_pcm_prepare)(pcm_handle); |
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336 } |
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337 if ( status < 0 ) { |
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338 /* Hmm, not much we can do - abort */ |
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339 this->enabled = 0; |
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340 return; |
| 0 | 341 } |
|
356
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342 continue; |
| 0 | 343 } |
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344 sample_buf += status * sample_size; |
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345 frames_left -= status; |
| 0 | 346 } |
| 347 } | |
| 348 | |
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349 static Uint8 *ALSA_GetAudioBuf(_THIS) |
| 0 | 350 { |
|
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351 return(mixbuf); |
| 0 | 352 } |
| 353 | |
|
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354 static void ALSA_CloseAudio(_THIS) |
| 0 | 355 { |
|
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356 if ( mixbuf != NULL ) { |
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357 SDL_FreeAudioMem(mixbuf); |
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358 mixbuf = NULL; |
| 0 | 359 } |
|
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360 if ( pcm_handle ) { |
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361 SDL_NAME(snd_pcm_drain)(pcm_handle); |
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362 SDL_NAME(snd_pcm_close)(pcm_handle); |
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363 pcm_handle = NULL; |
| 0 | 364 } |
| 365 } | |
| 366 | |
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367 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec) |
| 0 | 368 { |
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369 int status; |
| 1552 | 370 snd_pcm_hw_params_t *hwparams; |
| 371 snd_pcm_sw_params_t *swparams; | |
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372 snd_pcm_format_t format; |
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373 snd_pcm_uframes_t frames; |
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374 unsigned int rate; |
|
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375 #ifdef SET_PERIOD_SIZE |
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376 unsigned int periods; |
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377 #endif |
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378 unsigned int channels; |
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379 Uint16 test_format; |
| 0 | 380 |
| 381 /* Open the audio device */ | |
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382 /* Name of device should depend on # channels in spec */ |
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383 status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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384 |
|
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385 if ( status < 0 ) { |
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386 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); |
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387 return(-1); |
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388 } |
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389 |
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390 /* Figure out what the hardware is capable of */ |
| 1552 | 391 snd_pcm_hw_params_alloca(&hwparams); |
| 392 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams); | |
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393 if ( status < 0 ) { |
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394 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); |
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395 ALSA_CloseAudio(this); |
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396 return(-1); |
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397 } |
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398 |
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399 /* SDL only uses interleaved sample output */ |
| 1552 | 400 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); |
|
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401 if ( status < 0 ) { |
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402 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); |
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403 ALSA_CloseAudio(this); |
| 0 | 404 return(-1); |
| 405 } | |
| 406 | |
| 407 /* Try for a closest match on audio format */ | |
|
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408 status = -1; |
| 0 | 409 for ( test_format = SDL_FirstAudioFormat(spec->format); |
|
354
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410 test_format && (status < 0); ) { |
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411 switch ( test_format ) { |
| 0 | 412 case AUDIO_U8: |
|
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413 format = SND_PCM_FORMAT_U8; |
| 0 | 414 break; |
| 415 case AUDIO_S8: | |
|
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416 format = SND_PCM_FORMAT_S8; |
| 0 | 417 break; |
| 418 case AUDIO_S16LSB: | |
|
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419 format = SND_PCM_FORMAT_S16_LE; |
| 0 | 420 break; |
| 421 case AUDIO_S16MSB: | |
|
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422 format = SND_PCM_FORMAT_S16_BE; |
| 0 | 423 break; |
| 424 case AUDIO_U16LSB: | |
|
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425 format = SND_PCM_FORMAT_U16_LE; |
| 0 | 426 break; |
| 427 case AUDIO_U16MSB: | |
|
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428 format = SND_PCM_FORMAT_U16_BE; |
| 0 | 429 break; |
| 430 default: | |
|
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431 format = 0; |
| 0 | 432 break; |
| 433 } | |
|
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434 if ( format != 0 ) { |
| 1552 | 435 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format); |
|
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436 } |
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437 if ( status < 0 ) { |
| 0 | 438 test_format = SDL_NextAudioFormat(); |
| 439 } | |
| 440 } | |
|
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441 if ( status < 0 ) { |
| 0 | 442 SDL_SetError("Couldn't find any hardware audio formats"); |
|
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443 ALSA_CloseAudio(this); |
| 0 | 444 return(-1); |
| 445 } | |
| 446 spec->format = test_format; | |
| 447 | |
|
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448 /* Set the number of channels */ |
| 1552 | 449 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels); |
|
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450 channels = spec->channels; |
|
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451 if ( status < 0 ) { |
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452 status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams, &channels); |
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453 if ( status < 0 ) { |
|
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454 SDL_SetError("Couldn't set audio channels"); |
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455 ALSA_CloseAudio(this); |
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456 return(-1); |
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457 } |
|
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458 spec->channels = channels; |
|
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459 } |
| 0 | 460 |
|
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461 /* Set the audio rate */ |
|
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462 rate = spec->freq; |
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463 |
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464 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL); |
|
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465 if ( status < 0 ) { |
|
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466 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); |
|
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467 ALSA_CloseAudio(this); |
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468 return(-1); |
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469 } |
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470 spec->freq = rate; |
| 0 | 471 |
|
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472 /* Set the buffer size, in samples */ |
|
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473 #ifdef SET_PERIOD_SIZE |
|
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474 frames = spec->samples; |
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475 status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL); |
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476 if ( status < 0 ) { |
|
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477 SDL_SetError("Couldn't set period size: %s", SDL_NAME(snd_strerror)(status)); |
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478 ALSA_CloseAudio(this); |
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479 return(-1); |
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480 } |
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481 |
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482 spec->samples = frames; |
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483 |
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484 periods = 2; |
|
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485 status = SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL); |
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486 if ( status < 0 ) { |
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487 SDL_SetError("Couldn't set period count: %s", SDL_NAME(snd_strerror)(status)); |
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488 ALSA_CloseAudio(this); |
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489 return(-1); |
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490 } |
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491 #else |
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492 frames = spec->samples * 2; |
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493 status = SDL_NAME(snd_pcm_hw_params_set_buffer_size_near)(pcm_handle, hwparams, &frames); |
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494 #endif |
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495 |
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496 /* "set" the hardware with the desired parameters */ |
| 1552 | 497 status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams); |
| 498 if ( status < 0 ) { | |
| 499 SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status)); | |
| 500 ALSA_CloseAudio(this); | |
| 501 return(-1); | |
| 502 } | |
| 503 | |
|
1553
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Left the debug code more explicit
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504 /* This is useful for debugging... */ |
|
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505 #ifdef DEBUG_PERIOD_SIZE |
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506 { snd_pcm_uframes_t bufsize; snd_pcm_sframes_t persize; unsigned int periods; int dir; |
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507 SDL_NAME(snd_pcm_hw_params_get_buffer_size)(hwparams, &bufsize); |
|
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508 SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams, &persize, &dir); |
|
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509 SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams, &periods, &dir); |
| 1552 | 510 |
|
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511 fprintf(stderr, "ALSA: period size = %ld, periods = %u, buffer size = %lu\n", persize, periods, bufsize); |
| 1552 | 512 } |
|
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513 #endif |
| 1552 | 514 |
| 515 /* Set the software parameters */ | |
| 516 snd_pcm_sw_params_alloca(&swparams); | |
| 517 status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams); | |
|
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518 if ( status < 0 ) { |
| 1552 | 519 SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status)); |
| 520 ALSA_CloseAudio(this); | |
| 521 return(-1); | |
| 522 } | |
|
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523 status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 1); |
| 1552 | 524 if ( status < 0 ) { |
| 525 SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status)); | |
| 526 ALSA_CloseAudio(this); | |
| 527 return(-1); | |
| 528 } | |
| 529 status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams); | |
| 530 if ( status < 0 ) { | |
| 531 SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status)); | |
|
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532 ALSA_CloseAudio(this); |
| 0 | 533 return(-1); |
| 534 } | |
| 535 | |
|
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|
536 /* Calculate the final parameters for this audio specification */ |
|
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537 SDL_CalculateAudioSpec(spec); |
| 0 | 538 |
|
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539 /* Allocate mixing buffer */ |
|
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540 mixlen = spec->size; |
|
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541 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); |
|
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542 if ( mixbuf == NULL ) { |
|
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543 ALSA_CloseAudio(this); |
|
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544 return(-1); |
| 0 | 545 } |
|
1336
3692456e7b0f
Use SDL_ prefixed versions of C library functions.
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diff
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|
546 SDL_memset(mixbuf, spec->silence, spec->size); |
| 0 | 547 |
| 548 /* Get the parent process id (we're the parent of the audio thread) */ | |
| 549 parent = getpid(); | |
| 550 | |
|
765
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Editors Note: The original patch was modified to use SDL_Delay() instead of
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|
551 /* Switch to blocking mode for playback */ |
|
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|
552 SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); |
|
765
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Editors Note: The original patch was modified to use SDL_Delay() instead of
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|
553 |
| 0 | 554 /* We're ready to rock and roll. :-) */ |
| 555 return(0); | |
| 556 } |