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>
date Sat, 21 Aug 2004 12:27:02 +0000
parents c7c04f811994
children 05d4b93b911e
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 #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
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56 /* The tag name used by ALSA audio */
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57 #define DRIVER_NAME "alsa"
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58
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59 /* The default ALSA audio driver */
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60 #define DEFAULT_DEVICE "default"
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61
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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);
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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);
865
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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
942
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166 static const char *get_audio_device(int channels)
354
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167 {
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168 const char *device;
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169
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diff changeset
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
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177 }
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178
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179 /* Audio driver bootstrap functions */
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180
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181 static int Audio_Available(void)
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182 {
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183 int available;
354
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184 int status;
0
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185 snd_pcm_t *handle;
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186
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187 available = 0;
865
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188 if (LoadALSALibrary() < 0) {
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189 return available;
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190 }
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41a59de7f2ed Here are patches for SDL12 and SDL_mixer for 4 or 6 channel
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diff changeset
191 status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
354
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diff changeset
192 if ( status >= 0 ) {
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diff changeset
193 available = 1;
865
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194 SDL_NAME(snd_pcm_close)(handle);
0
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195 }
865
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196 UnloadALSALibrary();
0
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197 return(available);
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198 }
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199
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200 static void Audio_DeleteDevice(SDL_AudioDevice *device)
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201 {
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202 free(device->hidden);
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diff changeset
203 free(device);
865
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204 UnloadALSALibrary();
0
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diff changeset
205 }
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diff changeset
206
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207 static SDL_AudioDevice *Audio_CreateDevice(int devindex)
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diff changeset
208 {
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diff changeset
209 SDL_AudioDevice *this;
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diff changeset
210
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211 /* Initialize all variables that we clean on shutdown */
865
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212 LoadALSALibrary();
0
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diff changeset
213 this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
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parents:
diff changeset
214 if ( this ) {
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diff changeset
215 memset(this, 0, (sizeof *this));
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diff changeset
216 this->hidden = (struct SDL_PrivateAudioData *)
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diff changeset
217 malloc((sizeof *this->hidden));
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parents:
diff changeset
218 }
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parents:
diff changeset
219 if ( (this == NULL) || (this->hidden == NULL) ) {
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diff changeset
220 SDL_OutOfMemory();
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parents:
diff changeset
221 if ( this ) {
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parents:
diff changeset
222 free(this);
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parents:
diff changeset
223 }
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diff changeset
224 return(0);
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parents:
diff changeset
225 }
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diff changeset
226 memset(this->hidden, 0, (sizeof *this->hidden));
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parents:
diff changeset
227
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diff changeset
228 /* Set the function pointers */
354
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diff changeset
229 this->OpenAudio = ALSA_OpenAudio;
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diff changeset
230 this->WaitAudio = ALSA_WaitAudio;
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diff changeset
231 this->PlayAudio = ALSA_PlayAudio;
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diff changeset
232 this->GetAudioBuf = ALSA_GetAudioBuf;
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diff changeset
233 this->CloseAudio = ALSA_CloseAudio;
0
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diff changeset
234
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diff changeset
235 this->free = Audio_DeleteDevice;
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parents:
diff changeset
236
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diff changeset
237 return this;
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parents:
diff changeset
238 }
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diff changeset
239
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diff changeset
240 AudioBootStrap ALSA_bootstrap = {
354
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diff changeset
241 DRIVER_NAME, "ALSA 0.9 PCM audio",
0
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parents:
diff changeset
242 Audio_Available, Audio_CreateDevice
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parents:
diff changeset
243 };
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diff changeset
244
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diff changeset
245 /* This function waits until it is possible to write a full sound buffer */
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
246 static void ALSA_WaitAudio(_THIS)
0
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parents:
diff changeset
247 {
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parents:
diff changeset
248 /* Check to see if the thread-parent process is still alive */
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parents:
diff changeset
249 { static int cnt = 0;
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diff changeset
250 /* Note that this only works with thread implementations
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parents:
diff changeset
251 that use a different process id for each thread.
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diff changeset
252 */
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parents:
diff changeset
253 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
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parents:
diff changeset
254 if ( kill(parent, 0) < 0 ) {
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parents:
diff changeset
255 this->enabled = 0;
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parents:
diff changeset
256 }
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diff changeset
257 }
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diff changeset
258 }
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diff changeset
259 }
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diff changeset
260
354
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diff changeset
261 static void ALSA_PlayAudio(_THIS)
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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);
354
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diff changeset
271 if ( status < 0 ) {
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diff changeset
272 if ( status == -EAGAIN ) {
765
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273 SDL_Delay(1);
354
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diff changeset
274 continue;
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diff changeset
275 }
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diff changeset
276 if ( status == -ESTRPIPE ) {
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diff changeset
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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diff changeset
280 } while ( status == -EAGAIN );
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diff changeset
281 }
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diff changeset
282 if ( status < 0 ) {
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283 status = SDL_NAME(snd_pcm_prepare)(pcm_handle);
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diff changeset
284 }
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diff changeset
285 if ( status < 0 ) {
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diff changeset
286 /* Hmm, not much we can do - abort */
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diff changeset
287 this->enabled = 0;
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288 return;
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289 }
356
a1e54d1ba16f *** empty log message ***
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diff changeset
290 continue;
0
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291 }
354
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292 sample_buf += status * this->spec.channels;
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293 sample_len -= status;
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294 }
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295 }
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296
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297 static Uint8 *ALSA_GetAudioBuf(_THIS)
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298 {
354
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299 return(mixbuf);
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300 }
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301
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diff changeset
302 static void ALSA_CloseAudio(_THIS)
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303 {
354
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diff changeset
304 if ( mixbuf != NULL ) {
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diff changeset
305 SDL_FreeAudioMem(mixbuf);
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diff changeset
306 mixbuf = NULL;
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307 }
354
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diff changeset
308 if ( pcm_handle ) {
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309 SDL_NAME(snd_pcm_drain)(pcm_handle);
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diff changeset
310 SDL_NAME(snd_pcm_close)(pcm_handle);
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diff changeset
311 pcm_handle = NULL;
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312 }
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313 }
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314
354
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diff changeset
315 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
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316 {
354
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diff changeset
317 int status;
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diff changeset
318 snd_pcm_hw_params_t *params;
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diff changeset
319 snd_pcm_format_t format;
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diff changeset
320 snd_pcm_uframes_t frames;
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diff changeset
321 Uint16 test_format;
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322
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323 /* Open the audio device */
942
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diff changeset
324 /* Name of device should depend on # channels in spec */
41a59de7f2ed Here are patches for SDL12 and SDL_mixer for 4 or 6 channel
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diff changeset
325 status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
41a59de7f2ed Here are patches for SDL12 and SDL_mixer for 4 or 6 channel
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diff changeset
326
354
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diff changeset
327 if ( status < 0 ) {
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328 SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));
354
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diff changeset
329 return(-1);
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diff changeset
330 }
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diff changeset
331
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diff changeset
332 /* Figure out what the hardware is capable of */
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diff changeset
333 snd_pcm_hw_params_alloca(&params);
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334 status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, params);
354
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diff changeset
335 if ( status < 0 ) {
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diff changeset
336 SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));
354
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diff changeset
337 ALSA_CloseAudio(this);
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diff changeset
338 return(-1);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
339 }
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diff changeset
340
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diff changeset
341 /* SDL only uses interleaved sample output */
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diff changeset
342 status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
354
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diff changeset
343 if ( status < 0 ) {
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diff changeset
344 SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));
354
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diff changeset
345 ALSA_CloseAudio(this);
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346 return(-1);
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347 }
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348
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349 /* Try for a closest match on audio format */
354
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diff changeset
350 status = -1;
0
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351 for ( test_format = SDL_FirstAudioFormat(spec->format);
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
352 test_format && (status < 0); ) {
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
353 switch ( test_format ) {
0
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diff changeset
354 case AUDIO_U8:
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
355 format = SND_PCM_FORMAT_U8;
0
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diff changeset
356 break;
74212992fb08 Initial revision
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diff changeset
357 case AUDIO_S8:
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
358 format = SND_PCM_FORMAT_S8;
0
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diff changeset
359 break;
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diff changeset
360 case AUDIO_S16LSB:
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
361 format = SND_PCM_FORMAT_S16_LE;
0
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diff changeset
362 break;
74212992fb08 Initial revision
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diff changeset
363 case AUDIO_S16MSB:
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
364 format = SND_PCM_FORMAT_S16_BE;
0
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parents:
diff changeset
365 break;
74212992fb08 Initial revision
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diff changeset
366 case AUDIO_U16LSB:
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
367 format = SND_PCM_FORMAT_U16_LE;
0
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parents:
diff changeset
368 break;
74212992fb08 Initial revision
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diff changeset
369 case AUDIO_U16MSB:
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
370 format = SND_PCM_FORMAT_U16_BE;
0
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parents:
diff changeset
371 break;
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diff changeset
372 default:
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
373 format = 0;
0
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diff changeset
374 break;
74212992fb08 Initial revision
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parents:
diff changeset
375 }
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
376 if ( format != 0 ) {
865
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diff changeset
377 status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, params, format);
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
378 }
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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diff changeset
379 if ( status < 0 ) {
0
74212992fb08 Initial revision
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diff changeset
380 test_format = SDL_NextAudioFormat();
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parents:
diff changeset
381 }
74212992fb08 Initial revision
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parents:
diff changeset
382 }
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
383 if ( status < 0 ) {
0
74212992fb08 Initial revision
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parents:
diff changeset
384 SDL_SetError("Couldn't find any hardware audio formats");
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
385 ALSA_CloseAudio(this);
0
74212992fb08 Initial revision
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parents:
diff changeset
386 return(-1);
74212992fb08 Initial revision
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parents:
diff changeset
387 }
74212992fb08 Initial revision
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diff changeset
388 spec->format = test_format;
74212992fb08 Initial revision
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parents:
diff changeset
389
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
390 /* Set the number of channels */
865
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diff changeset
391 status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, params, spec->channels);
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
392 if ( status < 0 ) {
865
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diff changeset
393 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
394 if ( (status <= 0) || (status > 2) ) {
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
395 SDL_SetError("Couldn't set audio channels");
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
396 ALSA_CloseAudio(this);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
397 return(-1);
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
398 }
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
399 spec->channels = status;
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
400 }
0
74212992fb08 Initial revision
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parents:
diff changeset
401
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
402 /* Set the audio rate */
865
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parents: 769
diff changeset
403 status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, params, spec->freq, NULL);
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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parents: 297
diff changeset
404 if ( status < 0 ) {
865
92615154bb68 Date: Sun, 29 Feb 2004 15:14:22 +0200
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parents: 769
diff changeset
405 SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
354
30935e76acb5 Updated ALSA audio support for ALSA 0.9
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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;
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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
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422 return(-1);
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423 }
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424
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425 /* Calculate the final parameters for this audio specification */
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426 SDL_CalculateAudioSpec(spec);
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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);
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434 }
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435 memset(mixbuf, spec->silence, spec->size);
0
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436
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437 /* Get the parent process id (we're the parent of the audio thread) */
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438 parent = getpid();
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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
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443 /* We're ready to rock and roll. :-) */
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444 return(0);
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445 }