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0
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1 /*
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2 SDL - Simple DirectMedia Layer
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3 Copyright (C) 1997, 1998, 1999, 2000, 2001 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@devolution.com
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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 /* The tag name used by ALSA audio */
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45 #define DRIVER_NAME "alsa"
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46
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47 /* default card and device numbers as listed in dev/snd */
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48 static int card_no = 0;
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49 static int device_no = 0;
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50
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51 /* default channel communication parameters */
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52 #define DEFAULT_CPARAMS_RATE 22050
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53 #define DEFAULT_CPARAMS_VOICES 1
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54 #define DEFAULT_CPARAMS_FRAG_SIZE 512
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55 #define DEFAULT_CPARAMS_FRAGS_MIN 1
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56 #define DEFAULT_CPARAMS_FRAGS_MAX -1
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57
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58 /* Open the audio device for playback, and don't block if busy */
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59 #define OPEN_FLAGS (SND_PCM_OPEN_PLAYBACK|SND_PCM_OPEN_NONBLOCK)
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60
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61 /* Audio driver functions */
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62 static int PCM_OpenAudio(_THIS, SDL_AudioSpec *spec);
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63 static void PCM_WaitAudio(_THIS);
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64 static void PCM_PlayAudio(_THIS);
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65 static Uint8 *PCM_GetAudioBuf(_THIS);
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66 static void PCM_CloseAudio(_THIS);
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67
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68 /* PCM transfer channel parameters initialize function */
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69 static void init_pcm_cparams(snd_pcm_channel_params_t* cparams)
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70 {
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71 memset(cparams,0,sizeof(snd_pcm_channel_params_t));
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72
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73 cparams->channel = SND_PCM_CHANNEL_PLAYBACK;
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74 cparams->mode = SND_PCM_MODE_BLOCK;
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75 cparams->start_mode = SND_PCM_START_DATA; //_FULL
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76 cparams->stop_mode = SND_PCM_STOP_STOP;
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77 cparams->format.format = SND_PCM_SFMT_S16_LE;
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78 cparams->format.interleave = 1;
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79 cparams->format.rate = DEFAULT_CPARAMS_RATE;
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80 cparams->format.voices = DEFAULT_CPARAMS_VOICES;
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81 cparams->buf.block.frag_size = DEFAULT_CPARAMS_FRAG_SIZE;
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82 cparams->buf.block.frags_min = DEFAULT_CPARAMS_FRAGS_MIN;
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83 cparams->buf.block.frags_max = DEFAULT_CPARAMS_FRAGS_MAX;
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84 }
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85
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86 /* Audio driver bootstrap functions */
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87
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88 static int Audio_Available(void)
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89 /*
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90 See if we can open a nonblocking channel.
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91 Return value '1' means we can.
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92 Return value '0' means we cannot.
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93 */
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94 {
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95 int available;
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96 int rval;
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97 snd_pcm_t *handle;
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98 snd_pcm_channel_params_t cparams;
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99 #ifdef DEBUG_AUDIO
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100 snd_pcm_channel_status_t cstatus;
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101 #endif
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102
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103 available = 0;
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104 handle = NULL;
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105
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106 init_pcm_cparams(&cparams);
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107
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108 rval = snd_pcm_open(&handle, card_no, device_no, OPEN_FLAGS);
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109 if (rval >= 0)
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110 {
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111 rval = snd_pcm_plugin_params(handle, &cparams);
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112
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113 #ifdef DEBUG_AUDIO
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114 snd_pcm_plugin_status(handle, &cstatus);
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115 printf("status after snd_pcm_plugin_params call = %d\n",cstatus.status);
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116 #endif
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117 if (rval >= 0)
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118 {
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119 available = 1;
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120 }
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121 else
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122 {
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123 SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
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124 }
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125
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126 if ((rval = snd_pcm_close(handle)) < 0)
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127 {
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128 SDL_SetError("snd_pcm_close failed: %s\n",snd_strerror(rval));
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129 available = 0;
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130 }
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131 }
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132 else
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133 {
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134 SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
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135 }
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136
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137 return(available);
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138 }
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139
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140 static void Audio_DeleteDevice(SDL_AudioDevice *device)
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141 {
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142 free(device->hidden);
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143 free(device);
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144 }
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145
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146 static SDL_AudioDevice *Audio_CreateDevice(int devindex)
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147 {
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148 SDL_AudioDevice *this;
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149
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150 /* Initialize all variables that we clean on shutdown */
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151 this = (SDL_AudioDevice *)malloc(sizeof(SDL_AudioDevice));
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152 if ( this ) {
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153 memset(this, 0, (sizeof *this));
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154 this->hidden = (struct SDL_PrivateAudioData *)
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155 malloc((sizeof *this->hidden));
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156 }
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157 if ( (this == NULL) || (this->hidden == NULL) ) {
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158 SDL_OutOfMemory();
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159 if ( this ) {
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160 free(this);
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161 }
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162 return(0);
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163 }
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164 memset(this->hidden, 0, (sizeof *this->hidden));
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165 audio_handle = NULL;
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166
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167 /* Set the function pointers */
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168 this->OpenAudio = PCM_OpenAudio;
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169 this->WaitAudio = PCM_WaitAudio;
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170 this->PlayAudio = PCM_PlayAudio;
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171 this->GetAudioBuf = PCM_GetAudioBuf;
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172 this->CloseAudio = PCM_CloseAudio;
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173
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174 this->free = Audio_DeleteDevice;
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175
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176 return this;
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177 }
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178
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179 AudioBootStrap ALSA_bootstrap = {
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180 DRIVER_NAME, "ALSA PCM audio",
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181 Audio_Available, Audio_CreateDevice
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182 };
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183
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184 /* This function waits until it is possible to write a full sound buffer */
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185 static void PCM_WaitAudio(_THIS)
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186 {
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187
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188 /* Check to see if the thread-parent process is still alive */
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189 { static int cnt = 0;
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190 /* Note that this only works with thread implementations
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191 that use a different process id for each thread.
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192 */
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193 if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */
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194 if ( kill(parent, 0) < 0 ) {
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195 this->enabled = 0;
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196 }
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197 }
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198 }
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199
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200 /* See if we need to use timed audio synchronization */
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201 if ( frame_ticks )
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202 {
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203 /* Use timer for general audio synchronization */
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204 Sint32 ticks;
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205
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206 ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;
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207 if ( ticks > 0 )
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208 {
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209 SDL_Delay(ticks);
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210 }
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211 }
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212 else
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213 {
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214 /* Use select() for audio synchronization */
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215 fd_set fdset;
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216 struct timeval timeout;
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217 FD_ZERO(&fdset);
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218 FD_SET(audio_fd, &fdset);
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219 timeout.tv_sec = 10;
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220 timeout.tv_usec = 0;
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221 #ifdef DEBUG_AUDIO
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222 fprintf(stderr, "Waiting for audio to get ready\n");
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223 #endif
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224 if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 )
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225 {
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226 const char *message =
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227 "Audio timeout - buggy audio driver? (disabled)";
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228 /* In general we should never print to the screen,
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229 but in this case we have no other way of letting
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230 the user know what happened.
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231 */
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232 fprintf(stderr, "SDL: %s\n", message);
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233 this->enabled = 0;
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234 /* Don't try to close - may hang */
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235 audio_fd = -1;
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236 #ifdef DEBUG_AUDIO
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237 fprintf(stderr, "Done disabling audio\n");
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238 #endif
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239 }
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240 #ifdef DEBUG_AUDIO
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241 fprintf(stderr, "Ready!\n");
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242 #endif
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243 }
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244 }
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245
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246 static snd_pcm_channel_status_t cstatus;
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247
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248 static void PCM_PlayAudio(_THIS)
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249 {
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250 int written, rval;
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251
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252 /* Write the audio data, checking for EAGAIN (buffer full) and underrun */
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253 do {
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254 written = snd_pcm_plugin_write(audio_handle, pcm_buf, pcm_len);
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255 #ifdef DEBUG_AUDIO
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256 fprintf(stderr, "written = %d pcm_len = %d\n",written,pcm_len);
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257 #endif
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258 if (written != pcm_len)
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259 {
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260 if (errno == EAGAIN)
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261 {
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262 SDL_Delay(1); /* Let a little CPU time go by and try to write again */
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263 #ifdef DEBUG_AUDIO
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264 fprintf(stderr, "errno == EAGAIN\n");
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265 #endif
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266 }
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267 else
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268 {
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269 if( (rval = snd_pcm_plugin_status(audio_handle, &cstatus)) < 0 )
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270 {
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271 SDL_SetError("snd_pcm_plugin_status failed: %s\n", snd_strerror(rval));
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272 return;
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273 }
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274 if ( (cstatus.status == SND_PCM_STATUS_UNDERRUN)
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275 ||(cstatus.status == SND_PCM_STATUS_READY) )
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276 {
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277 #ifdef DEBUG_AUDIO
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278 fprintf(stderr, "buffer underrun\n");
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279 #endif
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280 if ( (rval = snd_pcm_plugin_prepare (audio_handle,SND_PCM_CHANNEL_PLAYBACK)) < 0 )
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281 {
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282 SDL_SetError("snd_pcm_plugin_prepare failed: %s\n",snd_strerror(rval) );
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283 return;
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284 }
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285 /* if we reach here, try to write again */
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286 }
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287 }
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288 }
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289 } while ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) );
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290
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291 /* Set the next write frame */
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292 if ( frame_ticks ) {
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293 next_frame += frame_ticks;
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294 }
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295
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296 /* If we couldn't write, assume fatal error for now */
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297 if ( written < 0 ) {
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298 this->enabled = 0;
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299 }
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300 return;
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301 }
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302
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303 static Uint8 *PCM_GetAudioBuf(_THIS)
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304 {
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305 return(pcm_buf);
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306 }
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307
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308 static void PCM_CloseAudio(_THIS)
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309 {
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310 int rval;
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311
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312 if ( pcm_buf != NULL ) {
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313 free(pcm_buf);
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314 pcm_buf = NULL;
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315 }
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316 if ( audio_handle != NULL ) {
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317 if ((rval = snd_pcm_plugin_flush(audio_handle,SND_PCM_CHANNEL_PLAYBACK)) < 0)
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318 {
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319 SDL_SetError("snd_pcm_plugin_flush failed: %s\n",snd_strerror(rval));
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320 return;
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321 }
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322 if ((rval = snd_pcm_close(audio_handle)) < 0)
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323 {
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324 SDL_SetError("snd_pcm_close failed: %s\n",snd_strerror(rval));
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325 return;
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326 }
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327 audio_handle = NULL;
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328 }
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329 }
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330
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331 static int PCM_OpenAudio(_THIS, SDL_AudioSpec *spec)
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332 {
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333 int rval;
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334 snd_pcm_channel_params_t cparams;
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335 snd_pcm_channel_setup_t csetup;
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336 int format;
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337 Uint16 test_format;
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338 int twidth;
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339
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340 /* initialize channel transfer parameters to default */
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341 init_pcm_cparams(&cparams);
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342
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343 /* Reset the timer synchronization flag */
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344 frame_ticks = 0.0;
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345
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346 /* Open the audio device */
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347
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348 rval = snd_pcm_open(&audio_handle, card_no, device_no, OPEN_FLAGS);
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349 if ( rval < 0 ) {
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350 SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
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351 return(-1);
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352 }
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353
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354 #ifdef PLUGIN_DISABLE_MMAP /* This is gone in newer versions of ALSA? */
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355 /* disable count status parameter */
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356 if ((rval = snd_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP))<0)
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357 {
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358 SDL_SetError("snd_plugin_set_disable failed: %s\n", snd_strerror(rval));
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359 return(-1);
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360 }
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361 #endif
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362
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363 pcm_buf = NULL;
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364
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365 /* Try for a closest match on audio format */
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366 format = 0;
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367 for ( test_format = SDL_FirstAudioFormat(spec->format);
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368 ! format && test_format; )
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369 {
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370 #ifdef DEBUG_AUDIO
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371 fprintf(stderr, "Trying format 0x%4.4x spec->samples %d\n", test_format,spec->samples);
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372 #endif
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373 /* if match found set format to equivalent ALSA format */
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374 switch ( test_format ) {
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375 case AUDIO_U8:
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376 format = SND_PCM_SFMT_U8;
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377 cparams.buf.block.frag_size = spec->samples * spec->channels;
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378 break;
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379 case AUDIO_S8:
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380 format = SND_PCM_SFMT_S8;
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381 cparams.buf.block.frag_size = spec->samples * spec->channels;
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382 break;
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383 case AUDIO_S16LSB:
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384 format = SND_PCM_SFMT_S16_LE;
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385 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
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386 break;
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387 case AUDIO_S16MSB:
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388 format = SND_PCM_SFMT_S16_BE;
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389 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
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390 break;
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391 case AUDIO_U16LSB:
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392 format = SND_PCM_SFMT_U16_LE;
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393 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
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394 break;
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395 case AUDIO_U16MSB:
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396 format = SND_PCM_SFMT_U16_BE;
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397 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
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398 break;
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399 default:
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400 break;
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401 }
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402 if ( ! format ) {
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403 test_format = SDL_NextAudioFormat();
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404 }
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405 }
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406 if ( format == 0 ) {
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407 SDL_SetError("Couldn't find any hardware audio formats");
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408 return(-1);
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409 }
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410 spec->format = test_format;
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411
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412 /* Set the audio format */
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413 cparams.format.format = format;
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414
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415 /* Set mono or stereo audio (currently only two channels supported) */
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416 cparams.format.voices = spec->channels;
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417
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418 #ifdef DEBUG_AUDIO
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419 printf("intializing channels %d\n", cparams.format.voices);
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420 #endif
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421
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422 /* Set rate */
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423 cparams.format.rate = spec->freq ;
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424
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425 /* Setup the transfer parameters according to cparams */
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426 rval = snd_pcm_plugin_params(audio_handle, &cparams);
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427 if (rval < 0) {
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428 SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
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429 return(-1);
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430 }
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431
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432 /* Make sure channel is setup right one last time */
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433 memset( &csetup, 0, sizeof( csetup ) );
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434 csetup.channel = SND_PCM_CHANNEL_PLAYBACK;
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435 if ( snd_pcm_plugin_setup( audio_handle, &csetup ) < 0 )
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436 {
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437 SDL_SetError("Unable to setup playback channel\n" );
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438 return(-1);
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439 }
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440
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441 #ifdef DEBUG_AUDIO
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442 else
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443 {
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444 fprintf(stderr,"requested format: %d\n",cparams.format.format);
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445 fprintf(stderr,"requested frag size: %d\n",cparams.buf.block.frag_size);
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446 fprintf(stderr,"requested max frags: %d\n\n",cparams.buf.block.frags_max);
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447
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448 fprintf(stderr,"real format: %d\n", csetup.format.format );
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449 fprintf(stderr,"real frag size : %d\n", csetup.buf.block.frag_size );
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450 fprintf(stderr,"real max frags : %d\n", csetup.buf.block.frags_max );
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451 }
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452 #endif // DEBUG_AUDIO
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453
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454 /* Allocate memory to the audio buffer and initialize with silence
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455 (Note that buffer size must be a multiple of fragment size, so find closest multiple)
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456 */
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457
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458 twidth = snd_pcm_format_width(format);
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459 if (twidth < 0) {
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460 printf("snd_pcm_format_width failed\n");
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461 twidth = 0;
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462 }
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463 #ifdef DEBUG_AUDIO
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464 printf("format is %d bits wide\n",twidth);
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465 #endif
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466
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467 pcm_len = csetup.buf.block.frag_size * (twidth/8) * csetup.format.voices ;
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468
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469 #ifdef DEBUG_AUDIO
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470 printf("pcm_len set to %d\n", pcm_len);
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471 #endif
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472
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473 if (pcm_len == 0)
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474 {
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475 pcm_len = csetup.buf.block.frag_size;
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476 }
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477
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478 pcm_buf = (Uint8*)malloc(pcm_len);
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479 if (pcm_buf == NULL) {
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480 SDL_SetError("pcm_buf malloc failed\n");
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481 return(-1);
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482 }
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483 memset(pcm_buf,spec->silence,pcm_len);
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484
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485 #ifdef DEBUG_AUDIO
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486 fprintf(stderr,"pcm_buf malloced and silenced.\n");
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|
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487 #endif
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488
|
|
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489 /* get the file descriptor */
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|
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490 if( (audio_fd = snd_pcm_file_descriptor(audio_handle, device_no)) < 0)
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|
|
491 {
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|
|
492 fprintf(stderr, "snd_pcm_file_descriptor failed with error code: %d\n", audio_fd);
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|
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493 }
|
|
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494
|
|
|
495 /* Trigger audio playback */
|
|
|
496 rval = snd_pcm_plugin_prepare( audio_handle, SND_PCM_CHANNEL_PLAYBACK);
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|
|
497 if (rval < 0) {
|
|
|
498 SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror (rval));
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|
|
499 return(-1);
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|
|
500 }
|
|
|
501 rval = snd_pcm_playback_go(audio_handle);
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|
|
502 if (rval < 0) {
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|
|
503 SDL_SetError("snd_pcm_playback_go failed: %s\n", snd_strerror (rval));
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|
|
504 return(-1);
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|
|
505 }
|
|
|
506
|
|
|
507 /* Check to see if we need to use select() workaround */
|
|
|
508 { char *workaround;
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|
|
509 workaround = getenv("SDL_DSP_NOSELECT");
|
|
|
510 if ( workaround ) {
|
|
|
511 frame_ticks = (float)(spec->samples*1000)/spec->freq;
|
|
|
512 next_frame = SDL_GetTicks()+frame_ticks;
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|
|
513 }
|
|
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514 }
|
|
|
515
|
|
|
516 /* Get the parent process id (we're the parent of the audio thread) */
|
|
|
517 parent = getpid();
|
|
|
518
|
|
|
519 /* We're ready to rock and roll. :-) */
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|
|
520 return(0);
|
|
|
521 }
|
