comparison src/audio/alsa/SDL_alsa_audio.c @ 354:30935e76acb5

Updated ALSA audio support for ALSA 0.9
author Sam Lantinga <slouken@libsdl.org>
date Mon, 15 Apr 2002 07:38:54 +0000
parents f6ffac90895c
children a1e54d1ba16f
comparison
equal deleted inserted replaced
353:d47637068e52 354:30935e76acb5
42 #include "SDL_alsa_audio.h" 42 #include "SDL_alsa_audio.h"
43 43
44 /* The tag name used by ALSA audio */ 44 /* The tag name used by ALSA audio */
45 #define DRIVER_NAME "alsa" 45 #define DRIVER_NAME "alsa"
46 46
47 /* default card and device numbers as listed in dev/snd */ 47 /* The default ALSA audio driver */
48 static int card_no = 0; 48 #define DEFAULT_DEVICE "plughw:0,0"
49 static int device_no = 0;
50
51 /* default channel communication parameters */
52 #define DEFAULT_CPARAMS_RATE 22050
53 #define DEFAULT_CPARAMS_VOICES 1
54 #define DEFAULT_CPARAMS_FRAG_SIZE 512
55 #define DEFAULT_CPARAMS_FRAGS_MIN 1
56 #define DEFAULT_CPARAMS_FRAGS_MAX -1
57
58 /* Open the audio device for playback, and don't block if busy */
59 #define OPEN_FLAGS (SND_PCM_OPEN_PLAYBACK|SND_PCM_OPEN_NONBLOCK)
60 49
61 /* Audio driver functions */ 50 /* Audio driver functions */
62 static int PCM_OpenAudio(_THIS, SDL_AudioSpec *spec); 51 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec);
63 static void PCM_WaitAudio(_THIS); 52 static void ALSA_WaitAudio(_THIS);
64 static void PCM_PlayAudio(_THIS); 53 static void ALSA_PlayAudio(_THIS);
65 static Uint8 *PCM_GetAudioBuf(_THIS); 54 static Uint8 *ALSA_GetAudioBuf(_THIS);
66 static void PCM_CloseAudio(_THIS); 55 static void ALSA_CloseAudio(_THIS);
67 56
68 /* PCM transfer channel parameters initialize function */ 57 static const char *get_audio_device()
69 static void init_pcm_cparams(snd_pcm_channel_params_t* cparams) 58 {
70 { 59 const char *device;
71 memset(cparams,0,sizeof(snd_pcm_channel_params_t)); 60
72 61 device = getenv("AUDIODEV"); /* Is there a standard variable name? */
73 cparams->channel = SND_PCM_CHANNEL_PLAYBACK; 62 if ( device == NULL ) {
74 cparams->mode = SND_PCM_MODE_BLOCK; 63 device = DEFAULT_DEVICE;
75 cparams->start_mode = SND_PCM_START_DATA; //_FULL 64 }
76 cparams->stop_mode = SND_PCM_STOP_STOP; 65 return device;
77 cparams->format.format = SND_PCM_SFMT_S16_LE;
78 cparams->format.interleave = 1;
79 cparams->format.rate = DEFAULT_CPARAMS_RATE;
80 cparams->format.voices = DEFAULT_CPARAMS_VOICES;
81 cparams->buf.block.frag_size = DEFAULT_CPARAMS_FRAG_SIZE;
82 cparams->buf.block.frags_min = DEFAULT_CPARAMS_FRAGS_MIN;
83 cparams->buf.block.frags_max = DEFAULT_CPARAMS_FRAGS_MAX;
84 } 66 }
85 67
86 /* Audio driver bootstrap functions */ 68 /* Audio driver bootstrap functions */
87 69
88 static int Audio_Available(void) 70 static int Audio_Available(void)
89 /*
90 See if we can open a nonblocking channel.
91 Return value '1' means we can.
92 Return value '0' means we cannot.
93 */
94 { 71 {
95 int available; 72 int available;
96 int rval; 73 int status;
97 snd_pcm_t *handle; 74 snd_pcm_t *handle;
98 snd_pcm_channel_params_t cparams;
99 #ifdef DEBUG_AUDIO
100 snd_pcm_channel_status_t cstatus;
101 #endif
102 75
103 available = 0; 76 available = 0;
104 handle = NULL; 77 status = snd_pcm_open(&handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, 0);
105 78 if ( status >= 0 ) {
106 init_pcm_cparams(&cparams); 79 available = 1;
107 80 snd_pcm_close(handle);
108 rval = snd_pcm_open(&handle, card_no, device_no, OPEN_FLAGS); 81 }
109 if (rval >= 0)
110 {
111 rval = snd_pcm_plugin_params(handle, &cparams);
112
113 #ifdef DEBUG_AUDIO
114 snd_pcm_plugin_status(handle, &cstatus);
115 printf("status after snd_pcm_plugin_params call = %d\n",cstatus.status);
116 #endif
117 if (rval >= 0)
118 {
119 available = 1;
120 }
121 else
122 {
123 SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval));
124 }
125
126 if ((rval = snd_pcm_close(handle)) < 0)
127 {
128 SDL_SetError("snd_pcm_close failed: %s\n",snd_strerror(rval));
129 available = 0;
130 }
131 }
132 else
133 {
134 SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval));
135 }
136
137 return(available); 82 return(available);
138 } 83 }
139 84
140 static void Audio_DeleteDevice(SDL_AudioDevice *device) 85 static void Audio_DeleteDevice(SDL_AudioDevice *device)
141 { 86 {
160 free(this); 105 free(this);
161 } 106 }
162 return(0); 107 return(0);
163 } 108 }
164 memset(this->hidden, 0, (sizeof *this->hidden)); 109 memset(this->hidden, 0, (sizeof *this->hidden));
165 audio_handle = NULL;
166 110
167 /* Set the function pointers */ 111 /* Set the function pointers */
168 this->OpenAudio = PCM_OpenAudio; 112 this->OpenAudio = ALSA_OpenAudio;
169 this->WaitAudio = PCM_WaitAudio; 113 this->WaitAudio = ALSA_WaitAudio;
170 this->PlayAudio = PCM_PlayAudio; 114 this->PlayAudio = ALSA_PlayAudio;
171 this->GetAudioBuf = PCM_GetAudioBuf; 115 this->GetAudioBuf = ALSA_GetAudioBuf;
172 this->CloseAudio = PCM_CloseAudio; 116 this->CloseAudio = ALSA_CloseAudio;
173 117
174 this->free = Audio_DeleteDevice; 118 this->free = Audio_DeleteDevice;
175 119
176 return this; 120 return this;
177 } 121 }
178 122
179 AudioBootStrap ALSA_bootstrap = { 123 AudioBootStrap ALSA_bootstrap = {
180 DRIVER_NAME, "ALSA PCM audio", 124 DRIVER_NAME, "ALSA 0.9 PCM audio",
181 Audio_Available, Audio_CreateDevice 125 Audio_Available, Audio_CreateDevice
182 }; 126 };
183 127
184 /* This function waits until it is possible to write a full sound buffer */ 128 /* This function waits until it is possible to write a full sound buffer */
185 static void PCM_WaitAudio(_THIS) 129 static void ALSA_WaitAudio(_THIS)
186 { 130 {
187
188 /* Check to see if the thread-parent process is still alive */ 131 /* Check to see if the thread-parent process is still alive */
189 { static int cnt = 0; 132 { static int cnt = 0;
190 /* Note that this only works with thread implementations 133 /* Note that this only works with thread implementations
191 that use a different process id for each thread. 134 that use a different process id for each thread.
192 */ 135 */
194 if ( kill(parent, 0) < 0 ) { 137 if ( kill(parent, 0) < 0 ) {
195 this->enabled = 0; 138 this->enabled = 0;
196 } 139 }
197 } 140 }
198 } 141 }
199 142 }
200 /* See if we need to use timed audio synchronization */ 143
201 if ( frame_ticks ) 144 static void ALSA_PlayAudio(_THIS)
202 { 145 {
203 /* Use timer for general audio synchronization */ 146 int status;
204 Sint32 ticks; 147 int sample_len;
205 148 signed short *sample_buf;
206 ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS; 149
207 if ( ticks > 0 ) 150 sample_len = this->spec.samples;
208 { 151 sample_buf = (signed short *)mixbuf;
209 SDL_Delay(ticks); 152 while ( sample_len > 0 ) {
210 } 153 status = snd_pcm_writei(pcm_handle, sample_buf, sample_len);
211 } 154 if ( status < 0 ) {
212 else 155 if ( status == -EAGAIN ) {
213 { 156 continue;
214 /* Use select() for audio synchronization */ 157 }
215 fd_set fdset; 158 if ( status == -ESTRPIPE ) {
216 struct timeval timeout; 159 do {
217 FD_ZERO(&fdset); 160 status = snd_pcm_resume(pcm_handle);
218 FD_SET(audio_fd, &fdset); 161 } while ( status == -EAGAIN );
219 timeout.tv_sec = 10; 162 }
220 timeout.tv_usec = 0; 163 if ( status < 0 ) {
221 #ifdef DEBUG_AUDIO 164 status = snd_pcm_prepare(pcm_handle);
222 fprintf(stderr, "Waiting for audio to get ready\n"); 165 }
223 #endif 166 if ( status < 0 ) {
224 if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) 167 /* Hmm, not much we can do - abort */
225 { 168 this->enabled = 0;
226 const char *message = 169 return;
227 "Audio timeout - buggy audio driver? (disabled)"; 170 }
228 /* In general we should never print to the screen, 171 }
229 but in this case we have no other way of letting 172 sample_buf += status * this->spec.channels;
230 the user know what happened. 173 sample_len -= status;
231 */ 174 }
232 fprintf(stderr, "SDL: %s\n", message); 175 }
233 this->enabled = 0; 176
234 /* Don't try to close - may hang */ 177 static Uint8 *ALSA_GetAudioBuf(_THIS)
235 audio_fd = -1; 178 {
236 #ifdef DEBUG_AUDIO 179 return(mixbuf);
237 fprintf(stderr, "Done disabling audio\n"); 180 }
238 #endif 181
239 } 182 static void ALSA_CloseAudio(_THIS)
240 #ifdef DEBUG_AUDIO 183 {
241 fprintf(stderr, "Ready!\n"); 184 if ( mixbuf != NULL ) {
242 #endif 185 SDL_FreeAudioMem(mixbuf);
243 } 186 mixbuf = NULL;
244 } 187 }
245 188 if ( pcm_handle ) {
246 static snd_pcm_channel_status_t cstatus; 189 snd_pcm_close(pcm_handle);
247 190 pcm_handle = NULL;
248 static void PCM_PlayAudio(_THIS) 191 }
249 { 192 }
250 int written, rval; 193
251 194 static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)
252 /* Write the audio data, checking for EAGAIN (buffer full) and underrun */ 195 {
253 do { 196 int status;
254 written = snd_pcm_plugin_write(audio_handle, pcm_buf, pcm_len); 197 snd_pcm_hw_params_t *params;
255 #ifdef DEBUG_AUDIO 198 snd_pcm_format_t format;
256 fprintf(stderr, "written = %d pcm_len = %d\n",written,pcm_len); 199 snd_pcm_uframes_t frames;
257 #endif 200 Uint16 test_format;
258 if (written != pcm_len)
259 {
260 if (errno == EAGAIN)
261 {
262 SDL_Delay(1); /* Let a little CPU time go by and try to write again */
263 #ifdef DEBUG_AUDIO
264 fprintf(stderr, "errno == EAGAIN\n");
265 #endif
266 }
267 else
268 {
269 if( (rval = snd_pcm_plugin_status(audio_handle, &cstatus)) < 0 )
270 {
271 SDL_SetError("snd_pcm_plugin_status failed: %s\n", snd_strerror(rval));
272 return;
273 }
274 if ( (cstatus.status == SND_PCM_STATUS_UNDERRUN)
275 ||(cstatus.status == SND_PCM_STATUS_READY) )
276 {
277 #ifdef DEBUG_AUDIO
278 fprintf(stderr, "buffer underrun\n");
279 #endif
280 if ( (rval = snd_pcm_plugin_prepare (audio_handle,SND_PCM_CHANNEL_PLAYBACK)) < 0 )
281 {
282 SDL_SetError("snd_pcm_plugin_prepare failed: %s\n",snd_strerror(rval) );
283 return;
284 }
285 /* if we reach here, try to write again */
286 }
287 }
288 }
289 } while ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) );
290
291 /* Set the next write frame */
292 if ( frame_ticks ) {
293 next_frame += frame_ticks;
294 }
295
296 /* If we couldn't write, assume fatal error for now */
297 if ( written < 0 ) {
298 this->enabled = 0;
299 }
300 return;
301 }
302
303 static Uint8 *PCM_GetAudioBuf(_THIS)
304 {
305 return(pcm_buf);
306 }
307
308 static void PCM_CloseAudio(_THIS)
309 {
310 int rval;
311
312 if ( pcm_buf != NULL ) {
313 free(pcm_buf);
314 pcm_buf = NULL;
315 }
316 if ( audio_handle != NULL ) {
317 if ((rval = snd_pcm_plugin_flush(audio_handle,SND_PCM_CHANNEL_PLAYBACK)) < 0)
318 {
319 SDL_SetError("snd_pcm_plugin_flush failed: %s\n",snd_strerror(rval));
320 return;
321 }
322 if ((rval = snd_pcm_close(audio_handle)) < 0)
323 {
324 SDL_SetError("snd_pcm_close failed: %s\n",snd_strerror(rval));
325 return;
326 }
327 audio_handle = NULL;
328 }
329 }
330
331 static int PCM_OpenAudio(_THIS, SDL_AudioSpec *spec)
332 {
333 int rval;
334 snd_pcm_channel_params_t cparams;
335 snd_pcm_channel_setup_t csetup;
336 int format;
337 Uint16 test_format;
338 int twidth;
339
340 /* initialize channel transfer parameters to default */
341 init_pcm_cparams(&cparams);
342
343 /* Reset the timer synchronization flag */
344 frame_ticks = 0.0;
345 201
346 /* Open the audio device */ 202 /* Open the audio device */
347 203 status = snd_pcm_open(&pcm_handle, get_audio_device(), SND_PCM_STREAM_PLAYBACK, 0);
348 rval = snd_pcm_open(&audio_handle, card_no, device_no, OPEN_FLAGS); 204 if ( status < 0 ) {
349 if ( rval < 0 ) { 205 SDL_SetError("Couldn't open audio device: %s", snd_strerror(status));
350 SDL_SetError("snd_pcm_open failed: %s\n", snd_strerror(rval)); 206 return(-1);
351 return(-1); 207 }
352 } 208
353 209 /* Figure out what the hardware is capable of */
354 #ifdef PLUGIN_DISABLE_MMAP /* This is gone in newer versions of ALSA? */ 210 snd_pcm_hw_params_alloca(&params);
355 /* disable count status parameter */ 211 status = snd_pcm_hw_params_any(pcm_handle, params);
356 if ((rval = snd_plugin_set_disable(audio_handle, PLUGIN_DISABLE_MMAP))<0) 212 if ( status < 0 ) {
357 { 213 SDL_SetError("Couldn't get hardware config: %s", snd_strerror(status));
358 SDL_SetError("snd_plugin_set_disable failed: %s\n", snd_strerror(rval)); 214 ALSA_CloseAudio(this);
359 return(-1); 215 return(-1);
360 } 216 }
361 #endif 217
362 218 /* SDL only uses interleaved sample output */
363 pcm_buf = NULL; 219 status = snd_pcm_hw_params_set_access(pcm_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);
220 if ( status < 0 ) {
221 SDL_SetError("Couldn't set interleaved access: %s", snd_strerror(status));
222 ALSA_CloseAudio(this);
223 return(-1);
224 }
364 225
365 /* Try for a closest match on audio format */ 226 /* Try for a closest match on audio format */
366 format = 0; 227 status = -1;
367 for ( test_format = SDL_FirstAudioFormat(spec->format); 228 for ( test_format = SDL_FirstAudioFormat(spec->format);
368 ! format && test_format; ) 229 test_format && (status < 0); ) {
369 { 230 switch ( test_format ) {
370 #ifdef DEBUG_AUDIO
371 fprintf(stderr, "Trying format 0x%4.4x spec->samples %d\n", test_format,spec->samples);
372 #endif
373 /* if match found set format to equivalent ALSA format */
374 switch ( test_format ) {
375 case AUDIO_U8: 231 case AUDIO_U8:
376 format = SND_PCM_SFMT_U8; 232 format = SND_PCM_FORMAT_U8;
377 cparams.buf.block.frag_size = spec->samples * spec->channels;
378 break; 233 break;
379 case AUDIO_S8: 234 case AUDIO_S8:
380 format = SND_PCM_SFMT_S8; 235 format = SND_PCM_FORMAT_S8;
381 cparams.buf.block.frag_size = spec->samples * spec->channels;
382 break; 236 break;
383 case AUDIO_S16LSB: 237 case AUDIO_S16LSB:
384 format = SND_PCM_SFMT_S16_LE; 238 format = SND_PCM_FORMAT_S16_LE;
385 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
386 break; 239 break;
387 case AUDIO_S16MSB: 240 case AUDIO_S16MSB:
388 format = SND_PCM_SFMT_S16_BE; 241 format = SND_PCM_FORMAT_S16_BE;
389 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
390 break; 242 break;
391 case AUDIO_U16LSB: 243 case AUDIO_U16LSB:
392 format = SND_PCM_SFMT_U16_LE; 244 format = SND_PCM_FORMAT_U16_LE;
393 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
394 break; 245 break;
395 case AUDIO_U16MSB: 246 case AUDIO_U16MSB:
396 format = SND_PCM_SFMT_U16_BE; 247 format = SND_PCM_FORMAT_U16_BE;
397 cparams.buf.block.frag_size = spec->samples*2 * spec->channels;
398 break; 248 break;
399 default: 249 default:
400 break; 250 format = 0;
401 } 251 break;
402 if ( ! format ) { 252 }
253 if ( format != 0 ) {
254 status = snd_pcm_hw_params_set_format(pcm_handle, params, format);
255 }
256 if ( status < 0 ) {
403 test_format = SDL_NextAudioFormat(); 257 test_format = SDL_NextAudioFormat();
404 } 258 }
405 } 259 }
406 if ( format == 0 ) { 260 if ( status < 0 ) {
407 SDL_SetError("Couldn't find any hardware audio formats"); 261 SDL_SetError("Couldn't find any hardware audio formats");
262 ALSA_CloseAudio(this);
408 return(-1); 263 return(-1);
409 } 264 }
410 spec->format = test_format; 265 spec->format = test_format;
411 266
412 /* Set the audio format */ 267 /* Set the number of channels */
413 cparams.format.format = format; 268 status = snd_pcm_hw_params_set_channels(pcm_handle, params, spec->channels);
414 269 if ( status < 0 ) {
415 /* Set mono or stereo audio (currently only two channels supported) */ 270 status = snd_pcm_hw_params_get_channels(params);
416 cparams.format.voices = spec->channels; 271 if ( (status <= 0) || (status > 2) ) {
417 272 SDL_SetError("Couldn't set audio channels");
418 #ifdef DEBUG_AUDIO 273 ALSA_CloseAudio(this);
419 printf("intializing channels %d\n", cparams.format.voices); 274 return(-1);
420 #endif 275 }
421 276 spec->channels = status;
422 /* Set rate */ 277 }
423 cparams.format.rate = spec->freq ; 278
424 279 /* Set the audio rate */
425 /* Setup the transfer parameters according to cparams */ 280 status = snd_pcm_hw_params_set_rate_near(pcm_handle, params, spec->freq, NULL);
426 rval = snd_pcm_plugin_params(audio_handle, &cparams); 281 if ( status < 0 ) {
427 if (rval < 0) { 282 SDL_SetError("Couldn't set audio frequency: %s", snd_strerror(status));
428 SDL_SetError("snd_pcm_channel_params failed: %s\n", snd_strerror (rval)); 283 ALSA_CloseAudio(this);
429 return(-1); 284 return(-1);
430 } 285 }
431 286 spec->freq = status;
432 /* Make sure channel is setup right one last time */ 287
433 memset( &csetup, 0, sizeof( csetup ) ); 288 /* Set the buffer size, in samples */
434 csetup.channel = SND_PCM_CHANNEL_PLAYBACK; 289 frames = spec->samples;
435 if ( snd_pcm_plugin_setup( audio_handle, &csetup ) < 0 ) 290 frames = snd_pcm_hw_params_set_period_size_near(pcm_handle, params, frames, NULL);
436 { 291 spec->samples = frames;
437 SDL_SetError("Unable to setup playback channel\n" ); 292 snd_pcm_hw_params_set_periods_near(pcm_handle, params, 2, NULL);
438 return(-1); 293
439 } 294 /* "set" the hardware with the desired parameters */
440 295 status = snd_pcm_hw_params(pcm_handle, params);
441 #ifdef DEBUG_AUDIO 296 if ( status < 0 ) {
442 else 297 SDL_SetError("Couldn't set audio parameters: %s", snd_strerror(status));
443 { 298 ALSA_CloseAudio(this);
444 fprintf(stderr,"requested format: %d\n",cparams.format.format); 299 return(-1);
445 fprintf(stderr,"requested frag size: %d\n",cparams.buf.block.frag_size); 300 }
446 fprintf(stderr,"requested max frags: %d\n\n",cparams.buf.block.frags_max); 301
447 302 /* Calculate the final parameters for this audio specification */
448 fprintf(stderr,"real format: %d\n", csetup.format.format ); 303 SDL_CalculateAudioSpec(spec);
449 fprintf(stderr,"real frag size : %d\n", csetup.buf.block.frag_size ); 304
450 fprintf(stderr,"real max frags : %d\n", csetup.buf.block.frags_max ); 305 /* Allocate mixing buffer */
451 } 306 mixlen = spec->size;
452 #endif // DEBUG_AUDIO 307 mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
453 308 if ( mixbuf == NULL ) {
454 /* Allocate memory to the audio buffer and initialize with silence 309 ALSA_CloseAudio(this);
455 (Note that buffer size must be a multiple of fragment size, so find closest multiple) 310 return(-1);
456 */ 311 }
457 312 memset(mixbuf, spec->silence, spec->size);
458 twidth = snd_pcm_format_width(format);
459 if (twidth < 0) {
460 printf("snd_pcm_format_width failed\n");
461 twidth = 0;
462 }
463 #ifdef DEBUG_AUDIO
464 printf("format is %d bits wide\n",twidth);
465 #endif
466
467 pcm_len = csetup.buf.block.frag_size * (twidth/8) * csetup.format.voices ;
468
469 #ifdef DEBUG_AUDIO
470 printf("pcm_len set to %d\n", pcm_len);
471 #endif
472
473 if (pcm_len == 0)
474 {
475 pcm_len = csetup.buf.block.frag_size;
476 }
477
478 pcm_buf = (Uint8*)malloc(pcm_len);
479 if (pcm_buf == NULL) {
480 SDL_SetError("pcm_buf malloc failed\n");
481 return(-1);
482 }
483 memset(pcm_buf,spec->silence,pcm_len);
484
485 #ifdef DEBUG_AUDIO
486 fprintf(stderr,"pcm_buf malloced and silenced.\n");
487 #endif
488
489 /* get the file descriptor */
490 if( (audio_fd = snd_pcm_file_descriptor(audio_handle, device_no)) < 0)
491 {
492 fprintf(stderr, "snd_pcm_file_descriptor failed with error code: %d\n", audio_fd);
493 }
494
495 /* Trigger audio playback */
496 rval = snd_pcm_plugin_prepare( audio_handle, SND_PCM_CHANNEL_PLAYBACK);
497 if (rval < 0) {
498 SDL_SetError("snd_pcm_plugin_prepare failed: %s\n", snd_strerror (rval));
499 return(-1);
500 }
501 rval = snd_pcm_playback_go(audio_handle);
502 if (rval < 0) {
503 SDL_SetError("snd_pcm_playback_go failed: %s\n", snd_strerror (rval));
504 return(-1);
505 }
506
507 /* Check to see if we need to use select() workaround */
508 { char *workaround;
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;
513 }
514 }
515 313
516 /* Get the parent process id (we're the parent of the audio thread) */ 314 /* Get the parent process id (we're the parent of the audio thread) */
517 parent = getpid(); 315 parent = getpid();
518 316
519 /* We're ready to rock and roll. :-) */ 317 /* We're ready to rock and roll. :-) */