Mercurial > sdl-ios-xcode
annotate src/audio/alsa/SDL_alsa_audio.c @ 942:41a59de7f2ed
Here are patches for SDL12 and SDL_mixer for 4 or 6 channel
surround sound on Linux using the Alsa driver. To use them, naturally
you need a sound card that will do 4 or 6 channels and probably also a
recent version of the Alsa drivers and library. Since the only SDL
output driver that knows about surround sound is the Alsa driver,
you���ll want to choose it, using:
export SDL_AUDIODRIVER=alsa
There are no syntactic changes to the programming API. No new
library calls, no differences in arguments.
There are two semantic changes:
(1) For library calls with number of channels as an argument, formerly
you could use only 1 or 2 for the number of channels. Now you
can also use 4 or 6.
(2) The two "left" and "right" arguments to Mix_SetPanning, for the
case of 4 or 6 channels, no longer simply control the volumes of
the left and right channels. Now the "left" argument is converted
to an angle and Mix_SetPosition is called, and the "right" argu-
ment is ignored.
With two exceptions, so far as I know, the modified SDL12 and
SDL_mixer work the same way as the original versions, when opened for
1 or 2 channel output. The two exceptions are bugs which I fixed.
Well, the first, anyway, is a bug for sure. When rate conversions up
or down by a factor of two are applied (in src/audio/SDL_audiocvt.c),
streams with different numbers of channels (that is, mono and stereo)
are treated the same way: either each sample is copied or every other
sample is omitted. This is ok for mono, but for stereo, it is frames
that should be copied or omitted, where by "frame" I mean a portion of
the stream containing one sample for each channel. (In the SDL source,
confusingly, sometimes frames are called "samples".) So for these
rate conversions, stereo streams have to be treated differently, and
they are, in my modified version.
The other problem that might be characterized as a bug arises
when SDL_mixer is passed a multichannel chunk which does not have an
integral number of frames. Due to the way the effect_position code
loops over frames, when the chunk ends with a partial frame, memory
outside the chunk buffer will be accessed. In the case of stereo,
it���s possible that because malloc may give more memory than requested,
this potential problem never actually causes a segment fault. I don���t
know. For 6 channel chunks, I do know, and it does cause segment
faults.
If SDL_mixer is passed defective chunks and this causes a segment
fault, arguably, that���s not a bug in SDL_mixer. Still, whether or not
it counts as a bug, it���s easy to protect against, so why not? I added
code in mixer.c to discard any partial frame at the end of a chunk.
Then what about when SDL or SDL_mixer is opened for 4 or 6 chan-
nel output? What happens with the parts of the current library
designed for stereo? I don���t know whether I���ve covered all the bases,
but I���ve tried:
(1) For playing 2 channel waves, or other cases where SDL knows it has
to match up a 2 channel source with a 4 or 6 channel output, I���ve
added code in SDL_audiocvt.c to make the necessary conversions.
(2) For playing midis using timidity, I���ve converted timidity to do 4
or 6 channel output, upon request.
(3) For playing mods using mikmod, I put ad hoc code in music.c to
convert the stereo output that mikmod produces to 4 or 6 chan-
nels. Obviously it would be better to change the mikmod code to
mix down into 4 or 6 channels, but I have a hard time following
the code in mikmod, so I didn���t do that.
(4) For playing mp3s, I put ad hoc code in smpeg to copy channels in
the case when 4 or 6 channel output is needed.
(5) There seems to be no problem with .ogg files - stereo .oggs can be
up converted as .wavs are.
(6) The effect_position code in SDL_mixer is now generalized to in-
clude the cases of 4 and 6 channel streams.
I���ve done a very limited amount of compatibility testing for some
of the games using SDL I happen to have. For details, see the file
TESTS.
I���ve put into a separate archive, Surround-SDL-testfiles.tgz, a
couple of 6 channel wave files for testing and a 6 channel ogg file.
If you have the right hardware and version of Alsa, you should be able
to play the wave files with the Alsa utility aplay (and hear all
channels, except maybe lfe, for chan-id.wav, since it���s rather faint).
Don���t expect aplay to give good sound, though. There���s something
wrong with the current version of aplay.
The canyon.ogg file is to test loading of 6 channel oggs. After
patching and compiling, you can play it with playmus. (My version of
ogg123 will not play it, and I had to patch mplayer to get it to play
6 channel oggs.)
Greg Lee <greg@ling.lll.hawaii.edu>
Thus, July 1, 2004
author | Sam Lantinga <slouken@libsdl.org> |
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date | Sat, 21 Aug 2004 12:27:02 +0000 |
parents | c7c04f811994 |
children | 05d4b93b911e |
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 #ifdef USE_DLVSYM |
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46 #define __USE_GNU |
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47 #endif |
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48 #include <dlfcn.h> |
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49 #include "SDL_name.h" |
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50 #include "SDL_loadso.h" |
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51 #else |
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52 #define SDL_NAME(X) X |
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53 #endif |
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54 |
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55 |
0 | 56 /* The tag name used by ALSA audio */ |
57 #define DRIVER_NAME "alsa" | |
58 | |
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59 /* The default ALSA audio driver */ |
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60 #define DEFAULT_DEVICE "default" |
0 | 61 |
62 /* Audio driver functions */ | |
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63 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec); |
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64 static void ALSA_WaitAudio(_THIS); |
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65 static void ALSA_PlayAudio(_THIS); |
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66 static Uint8 *ALSA_GetAudioBuf(_THIS); |
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67 static void ALSA_CloseAudio(_THIS); |
0 | 68 |
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69 #ifdef ALSA_DYNAMIC |
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70 |
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71 static const char *alsa_library = ALSA_DYNAMIC; |
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72 static void *alsa_handle = NULL; |
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73 static int alsa_loaded = 0; |
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74 |
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75 static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); |
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76 static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode); |
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77 static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm); |
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78 static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size); |
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79 static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm); |
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80 static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm); |
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81 static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm); |
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82 static const char *(*SDL_NAME(snd_strerror))(int errnum); |
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83 static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void); |
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84 static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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85 static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access); |
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86 static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val); |
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87 static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val); |
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88 static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params); |
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89 static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir); |
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90 static snd_pcm_uframes_t (*SDL_NAME(snd_pcm_hw_params_set_period_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val, int *dir); |
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91 static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir); |
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92 static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params); |
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93 static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock); |
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94 #define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof) |
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95 |
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96 static struct { |
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97 const char *name; |
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98 void **func; |
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99 } alsa_functions[] = { |
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100 { "snd_pcm_open", (void**)&SDL_NAME(snd_pcm_open) }, |
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101 { "snd_pcm_close", (void**)&SDL_NAME(snd_pcm_close) }, |
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102 { "snd_pcm_writei", (void**)&SDL_NAME(snd_pcm_writei) }, |
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103 { "snd_pcm_resume", (void**)&SDL_NAME(snd_pcm_resume) }, |
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104 { "snd_pcm_prepare", (void**)&SDL_NAME(snd_pcm_prepare) }, |
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105 { "snd_pcm_drain", (void**)&SDL_NAME(snd_pcm_drain) }, |
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106 { "snd_strerror", (void**)&SDL_NAME(snd_strerror) }, |
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107 { "snd_pcm_hw_params_sizeof", (void**)&SDL_NAME(snd_pcm_hw_params_sizeof) }, |
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108 { "snd_pcm_hw_params_any", (void**)&SDL_NAME(snd_pcm_hw_params_any) }, |
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109 { "snd_pcm_hw_params_set_access", (void**)&SDL_NAME(snd_pcm_hw_params_set_access) }, |
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110 { "snd_pcm_hw_params_set_format", (void**)&SDL_NAME(snd_pcm_hw_params_set_format) }, |
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111 { "snd_pcm_hw_params_set_channels", (void**)&SDL_NAME(snd_pcm_hw_params_set_channels) }, |
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112 { "snd_pcm_hw_params_get_channels", (void**)&SDL_NAME(snd_pcm_hw_params_get_channels) }, |
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113 { "snd_pcm_hw_params_set_rate_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_rate_near) }, |
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114 { "snd_pcm_hw_params_set_period_size_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_period_size_near) }, |
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115 { "snd_pcm_hw_params_set_periods_near", (void**)&SDL_NAME(snd_pcm_hw_params_set_periods_near) }, |
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116 { "snd_pcm_hw_params", (void**)&SDL_NAME(snd_pcm_hw_params) }, |
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117 { "snd_pcm_nonblock", (void**)&SDL_NAME(snd_pcm_nonblock) }, |
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118 }; |
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119 |
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120 static void UnloadALSALibrary(void) { |
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121 if (alsa_loaded) { |
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122 /* SDL_UnloadObject(alsa_handle);*/ |
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123 dlclose(alsa_handle); |
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124 alsa_handle = NULL; |
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125 alsa_loaded = 0; |
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126 } |
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127 } |
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128 |
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129 static int LoadALSALibrary(void) { |
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130 int i, retval = -1; |
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131 |
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132 /* alsa_handle = SDL_LoadObject(alsa_library);*/ |
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133 alsa_handle = dlopen(alsa_library,RTLD_NOW); |
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134 if (alsa_handle) { |
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135 alsa_loaded = 1; |
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136 retval = 0; |
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137 for (i = 0; i < SDL_TABLESIZE(alsa_functions); i++) { |
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138 /* *alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/ |
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139 #ifdef USE_DLVSYM |
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140 *alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9"); |
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141 if (!*alsa_functions[i].func) |
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142 #endif |
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143 *alsa_functions[i].func = dlsym(alsa_handle,alsa_functions[i].name); |
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144 if (!*alsa_functions[i].func) { |
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145 retval = -1; |
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146 UnloadALSALibrary(); |
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147 break; |
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148 } |
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149 } |
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150 } |
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151 return retval; |
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152 } |
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153 |
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154 #else |
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155 |
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156 static void UnloadALSALibrary(void) { |
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157 return; |
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158 } |
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159 |
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160 static int LoadALSALibrary(void) { |
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161 return 0; |
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162 } |
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163 |
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164 #endif /* ALSA_DYNAMIC */ |
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165 |
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166 static const char *get_audio_device(int channels) |
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167 { |
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168 const char *device; |
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169 |
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170 device = getenv("AUDIODEV"); /* Is there a standard variable name? */ |
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171 if ( device == NULL ) { |
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172 if (channels == 6) device = "surround51"; |
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173 else if (channels == 4) device = "surround40"; |
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174 else device = DEFAULT_DEVICE; |
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175 } |
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176 return device; |
0 | 177 } |
178 | |
179 /* Audio driver bootstrap functions */ | |
180 | |
181 static int Audio_Available(void) | |
182 { | |
183 int available; | |
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184 int status; |
0 | 185 snd_pcm_t *handle; |
186 | |
187 available = 0; | |
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188 if (LoadALSALibrary() < 0) { |
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189 return available; |
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190 } |
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191 status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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192 if ( status >= 0 ) { |
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193 available = 1; |
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194 SDL_NAME(snd_pcm_close)(handle); |
0 | 195 } |
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196 UnloadALSALibrary(); |
0 | 197 return(available); |
198 } | |
199 | |
200 static void Audio_DeleteDevice(SDL_AudioDevice *device) | |
201 { | |
202 free(device->hidden); | |
203 free(device); | |
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204 UnloadALSALibrary(); |
0 | 205 } |
206 | |
207 static SDL_AudioDevice *Audio_CreateDevice(int devindex) | |
208 { | |
209 SDL_AudioDevice *this; | |
210 | |
211 /* Initialize all variables that we clean on shutdown */ | |
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212 LoadALSALibrary(); |
0 | 213 this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice)); |
214 if ( this ) { | |
215 memset(this, 0, (sizeof *this)); | |
216 this->hidden = (struct SDL_PrivateAudioData *) | |
217 malloc((sizeof *this->hidden)); | |
218 } | |
219 if ( (this == NULL) || (this->hidden == NULL) ) { | |
220 SDL_OutOfMemory(); | |
221 if ( this ) { | |
222 free(this); | |
223 } | |
224 return(0); | |
225 } | |
226 memset(this->hidden, 0, (sizeof *this->hidden)); | |
227 | |
228 /* Set the function pointers */ | |
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229 this->OpenAudio = ALSA_OpenAudio; |
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230 this->WaitAudio = ALSA_WaitAudio; |
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231 this->PlayAudio = ALSA_PlayAudio; |
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232 this->GetAudioBuf = ALSA_GetAudioBuf; |
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233 this->CloseAudio = ALSA_CloseAudio; |
0 | 234 |
235 this->free = Audio_DeleteDevice; | |
236 | |
237 return this; | |
238 } | |
239 | |
240 AudioBootStrap ALSA_bootstrap = { | |
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241 DRIVER_NAME, "ALSA 0.9 PCM audio", |
0 | 242 Audio_Available, Audio_CreateDevice |
243 }; | |
244 | |
245 /* This function waits until it is possible to write a full sound buffer */ | |
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246 static void ALSA_WaitAudio(_THIS) |
0 | 247 { |
248 /* Check to see if the thread-parent process is still alive */ | |
249 { static int cnt = 0; | |
250 /* Note that this only works with thread implementations | |
251 that use a different process id for each thread. | |
252 */ | |
253 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */ | |
254 if ( kill(parent, 0) < 0 ) { | |
255 this->enabled = 0; | |
256 } | |
257 } | |
258 } | |
259 } | |
260 | |
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261 static void ALSA_PlayAudio(_THIS) |
0 | 262 { |
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263 int status; |
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264 int sample_len; |
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265 signed short *sample_buf; |
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266 |
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267 sample_len = this->spec.samples; |
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268 sample_buf = (signed short *)mixbuf; |
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269 while ( sample_len > 0 ) { |
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270 status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len); |
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271 if ( status < 0 ) { |
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272 if ( status == -EAGAIN ) { |
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273 SDL_Delay(1); |
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274 continue; |
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275 } |
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276 if ( status == -ESTRPIPE ) { |
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277 do { |
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278 SDL_Delay(1); |
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279 status = SDL_NAME(snd_pcm_resume)(pcm_handle); |
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280 } while ( status == -EAGAIN ); |
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281 } |
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282 if ( status < 0 ) { |
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283 status = SDL_NAME(snd_pcm_prepare)(pcm_handle); |
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284 } |
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285 if ( status < 0 ) { |
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286 /* Hmm, not much we can do - abort */ |
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287 this->enabled = 0; |
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288 return; |
0 | 289 } |
356
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290 continue; |
0 | 291 } |
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292 sample_buf += status * this->spec.channels; |
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293 sample_len -= status; |
0 | 294 } |
295 } | |
296 | |
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297 static Uint8 *ALSA_GetAudioBuf(_THIS) |
0 | 298 { |
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299 return(mixbuf); |
0 | 300 } |
301 | |
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302 static void ALSA_CloseAudio(_THIS) |
0 | 303 { |
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304 if ( mixbuf != NULL ) { |
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305 SDL_FreeAudioMem(mixbuf); |
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306 mixbuf = NULL; |
0 | 307 } |
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308 if ( pcm_handle ) { |
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309 SDL_NAME(snd_pcm_drain)(pcm_handle); |
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310 SDL_NAME(snd_pcm_close)(pcm_handle); |
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311 pcm_handle = NULL; |
0 | 312 } |
313 } | |
314 | |
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315 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec) |
0 | 316 { |
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317 int status; |
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318 snd_pcm_hw_params_t *params; |
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319 snd_pcm_format_t format; |
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320 snd_pcm_uframes_t frames; |
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321 Uint16 test_format; |
0 | 322 |
323 /* Open the audio device */ | |
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324 /* Name of device should depend on # channels in spec */ |
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325 status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); |
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326 |
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327 if ( status < 0 ) { |
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328 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status)); |
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329 return(-1); |
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330 } |
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331 |
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332 /* Figure out what the hardware is capable of */ |
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333 snd_pcm_hw_params_alloca(¶ms); |
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334 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params); |
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335 if ( status < 0 ) { |
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336 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status)); |
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337 ALSA_CloseAudio(this); |
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338 return(-1); |
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339 } |
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340 |
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341 /* SDL only uses interleaved sample output */ |
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342 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); |
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343 if ( status < 0 ) { |
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344 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status)); |
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345 ALSA_CloseAudio(this); |
0 | 346 return(-1); |
347 } | |
348 | |
349 /* Try for a closest match on audio format */ | |
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350 status = -1; |
0 | 351 for ( test_format = SDL_FirstAudioFormat(spec->format); |
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352 test_format && (status < 0); ) { |
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353 switch ( test_format ) { |
0 | 354 case AUDIO_U8: |
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355 format = SND_PCM_FORMAT_U8; |
0 | 356 break; |
357 case AUDIO_S8: | |
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358 format = SND_PCM_FORMAT_S8; |
0 | 359 break; |
360 case AUDIO_S16LSB: | |
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361 format = SND_PCM_FORMAT_S16_LE; |
0 | 362 break; |
363 case AUDIO_S16MSB: | |
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364 format = SND_PCM_FORMAT_S16_BE; |
0 | 365 break; |
366 case AUDIO_U16LSB: | |
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367 format = SND_PCM_FORMAT_U16_LE; |
0 | 368 break; |
369 case AUDIO_U16MSB: | |
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370 format = SND_PCM_FORMAT_U16_BE; |
0 | 371 break; |
372 default: | |
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373 format = 0; |
0 | 374 break; |
375 } | |
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376 if ( format != 0 ) { |
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377 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format); |
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378 } |
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379 if ( status < 0 ) { |
0 | 380 test_format = SDL_NextAudioFormat(); |
381 } | |
382 } | |
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383 if ( status < 0 ) { |
0 | 384 SDL_SetError("Couldn't find any hardware audio formats"); |
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385 ALSA_CloseAudio(this); |
0 | 386 return(-1); |
387 } | |
388 spec->format = test_format; | |
389 | |
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390 /* Set the number of channels */ |
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391 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels); |
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392 if ( status < 0 ) { |
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393 status = SDL_NAME(snd_pcm_hw_params_get_channels)(params); |
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394 if ( (status <= 0) || (status > 2) ) { |
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395 SDL_SetError("Couldn't set audio channels"); |
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396 ALSA_CloseAudio(this); |
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397 return(-1); |
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398 } |
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399 spec->channels = status; |
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400 } |
0 | 401 |
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402 /* Set the audio rate */ |
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403 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL); |
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404 if ( status < 0 ) { |
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405 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status)); |
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406 ALSA_CloseAudio(this); |
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407 return(-1); |
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408 } |
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409 spec->freq = status; |
0 | 410 |
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411 /* Set the buffer size, in samples */ |
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412 frames = spec->samples; |
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413 frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, params, frames, NULL); |
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414 spec->samples = frames; |
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415 SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, params, 2, NULL); |
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416 |
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417 /* "set" the hardware with the desired parameters */ |
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418 status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, params); |
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419 if ( status < 0 ) { |
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420 SDL_SetError("Couldn't set audio parameters: %s", SDL_NAME(snd_strerror)(status)); |
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421 ALSA_CloseAudio(this); |
0 | 422 return(-1); |
423 } | |
424 | |
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425 /* Calculate the final parameters for this audio specification */ |
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426 SDL_CalculateAudioSpec(spec); |
0 | 427 |
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428 /* Allocate mixing buffer */ |
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429 mixlen = spec->size; |
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430 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen); |
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431 if ( mixbuf == NULL ) { |
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432 ALSA_CloseAudio(this); |
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433 return(-1); |
0 | 434 } |
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435 memset(mixbuf, spec->silence, spec->size); |
0 | 436 |
437 /* Get the parent process id (we're the parent of the audio thread) */ | |
438 parent = getpid(); | |
439 | |
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440 /* Switch to blocking mode for playback */ |
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441 SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0); |
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442 |
0 | 443 /* We're ready to rock and roll. :-) */ |
444 return(0); | |
445 } |