cinelerra-5.1/cinelerra/audioalsa.C

   1
   2 /*
   3  * CINELERRA
   4  * Copyright (C) 2008 Adam Williams <broadcast at earthling dot net>
   5  *
   6  * This program is free software; you can redistribute it and/or modify
   7  * it under the terms of the GNU General Public License as published by
   8  * the Free Software Foundation; either version 2 of the License, or
   9  * (at your option) any later version.
  10  *
  11  * This program is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  14  * GNU General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU General Public License
  17  * along with this program; if not, write to the Free Software
  18  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  19  *
  20  */
  21
  22 #include "audiodevice.h"
  23 #include "audioalsa.h"
  24 #include "bcsignals.h"
  25 #include "language.h"
  26 #include "mutex.h"
  27 #include "playbackconfig.h"
  28 #include "preferences.h"
  29 #include "recordconfig.h"
  30
  31 #include <errno.h>
  32
  33 #ifdef HAVE_ALSA
  34
  35 AudioALSA::AudioALSA(AudioDevice *device)
  36  : AudioLowLevel(device)
  37 {
  38         buffer_position = 0;
  39         samples_written = 0;
  40         timer = new Timer;
  41         delay = 0;
  42         period_size = 0;
  43         timer_lock = new Mutex("AudioALSA::timer_lock");
  44         interrupted = 0;
  45         dsp_in = 0;
  46         dsp_out = 0;
  47 }
  48
  49 AudioALSA::~AudioALSA()
  50 {
  51         delete timer_lock;
  52         delete timer;
  53 }
  54
  55 // leak checking
  56 static class alsa_leaks
  57 {
  58 public:
  59 // This is required in the top thread for Alsa to work
  60         alsa_leaks() {
  61                 ArrayList<char*> *alsa_titles = new ArrayList<char*>;
  62                 alsa_titles->set_array_delete();
  63                 AudioALSA::list_devices(alsa_titles, 0, MODEPLAY);
  64                 alsa_titles->remove_all_objects();
  65                 delete alsa_titles;
  66         }
  67         ~alsa_leaks() { snd_config_update_free_global(); }
  68 } alsa_leak;
  69
  70 void AudioALSA::list_devices(ArrayList<char*> *devices, int pcm_title, int mode)
  71 {
  72         snd_ctl_t *handle;
  73         int card, err, dev;
  74         snd_ctl_card_info_t *info;
  75         snd_pcm_info_t *pcminfo;
  76         char string[BCTEXTLEN];
  77         snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
  78
  79         switch(mode)
  80         {
  81                 case MODERECORD:
  82                         stream = SND_PCM_STREAM_CAPTURE;
  83                         break;
  84                 case MODEPLAY:
  85                         stream = SND_PCM_STREAM_PLAYBACK;
  86                         break;
  87         }
  88
  89
  90         snd_ctl_card_info_alloca(&info);
  91         snd_pcm_info_alloca(&pcminfo);
  92
  93         card = -1;
  94 #define DEFAULT_DEVICE "default"
  95         char *result = new char[strlen(DEFAULT_DEVICE) + 1];
  96         devices->append(result);
  97         strcpy(result, DEFAULT_DEVICE);
  98
  99         while(snd_card_next(&card) >= 0)
 100         {
 101                 char name[BCTEXTLEN];
 102                 if(card < 0) break;
 103                 sprintf(name, "hw:%i", card);
 104
 105                 if((err = snd_ctl_open(&handle, name, 0)) < 0)
 106                 {
 107                         printf("AudioALSA::list_devices card=%i: %s\n", card, snd_strerror(err));
 108                         continue;
 109                 }
 110
 111                 if((err = snd_ctl_card_info(handle, info)) < 0)
 112                 {
 113                         printf("AudioALSA::list_devices card=%i: %s\n", card, snd_strerror(err));
 114                         snd_ctl_close(handle);
 115                         continue;
 116                 }
 117
 118                 dev = -1;
 119
 120                 while(1)
 121                 {
 122                         if(snd_ctl_pcm_next_device(handle, &dev) < 0)
 123                                 printf("AudioALSA::list_devices: snd_ctl_pcm_next_device\n");
 124
 125                         if (dev < 0)
 126                                 break;
 127
 128                         snd_pcm_info_set_device(pcminfo, dev);
 129                         snd_pcm_info_set_subdevice(pcminfo, 0);
 130                         snd_pcm_info_set_stream(pcminfo, stream);
 131
 132                         if((err = snd_ctl_pcm_info(handle, pcminfo)) < 0)
 133                         {
 134                                 if(err != -ENOENT)
 135                                         printf("AudioALSA::list_devices card=%i: %s\n", card, snd_strerror(err));
 136                                 continue;
 137                         }
 138
 139                         if(pcm_title)
 140                         {
 141                                 sprintf(string, "plughw:%d,%d", card, dev);
 142 //                              strcpy(string, "cards.pcm.front");
 143                         }
 144                         else
 145                         {
 146                                 sprintf(string, "%s #%d",
 147                                         snd_ctl_card_info_get_name(info),
 148                                         dev);
 149                         }
 150
 151                         char *result = devices->append(new char[strlen(string) + 1]);
 152                         strcpy(result, string);
 153                 }
 154
 155
 156
 157                 snd_ctl_close(handle);
 158         }
 159
 160 #ifdef HAVE_PACTL
 161 // attempt to add pulseaudio "monitor" devices
 162 //  run: pactl list <sources>|<sinks>
 163 //   scan output for <Source/Sink> #n,  Name: <device>
 164 //   build alsa device config and add to alsa snd_config
 165
 166         const char *arg = 0;
 167         switch( mode ) {
 168                 case MODERECORD:
 169                         arg = "source";
 170                         break;
 171                 case MODEPLAY:
 172                         arg = "sink";
 173                         break;
 174         }
 175         FILE *pactl = 0;
 176         char line[BCTEXTLEN];
 177         if( arg ) {
 178                 sprintf(line, "pactl list %ss", arg);
 179                 pactl = popen(line,"r");
 180         }
 181         if( pactl ) {
 182                 if( pcm_title ) snd_config_update();
 183                 char name[BCTEXTLEN], pa_name[BCTEXTLEN], device[BCTEXTLEN];
 184                 name[0] = pa_name[0] = device[0] = 0;
 185                 int arg_len = strlen(arg);
 186                 while( fgets(line, sizeof(line), pactl) ) {
 187                         if( !strncasecmp(line, arg, arg_len) ) {
 188                                 char *sp = name, *id = pa_name;
 189                                 for( char *cp=line; *cp && *cp!='\n'; *sp++=*cp++ )
 190                                         *id++ = (*cp>='A' && *cp<='Z') ||
 191                                                 (*cp>='a' && *cp<='z') ||
 192                                                 (*cp>='0' && *cp<='9') ? *cp : '_';
 193                                 *sp++ = 0;  *id = 0;
 194                                 if( !pcm_title )
 195                                         devices->append(strcpy(new char[sp-name], name));
 196                                 continue;
 197                         }
 198                         if( !pcm_title ) continue;
 199                         if( sscanf(line, " Name: %s", device) != 1 ) continue;
 200                         int len = strlen(pa_name);
 201                         devices->append(strcpy(new char[len+1], pa_name));
 202                         char alsa_config[BCTEXTLEN];
 203                         len = snprintf(alsa_config, sizeof(alsa_config),
 204                                 "pcm.!%s {\n type pulse\n device %s\n}\n"
 205                                 "ctl.!%s {\n type pulse\n device %s\n}\n",
 206                                 pa_name, device, pa_name, device);
 207                         FILE *fp = fmemopen(alsa_config,len,"r");
 208                         snd_input_t *inp;
 209                         snd_input_stdio_attach(&inp, fp, 1);
 210                         snd_config_load(snd_config, inp);
 211                         name[0] = pa_name[0] = device[0] = 0;
 212                         snd_input_close(inp);
 213                 }
 214                 pclose(pactl);
 215         }
 216 #endif
 217 }
 218
 219 void AudioALSA::translate_name(char *output, char *input, int mode)
 220 {
 221         ArrayList<char*> titles;
 222         titles.set_array_delete();
 223
 224         ArrayList<char*> pcm_titles;
 225         pcm_titles.set_array_delete();
 226
 227         list_devices(&titles, 0, mode);
 228         list_devices(&pcm_titles, 1, mode);
 229
 230         sprintf(output, "default");
 231         for(int i = 0; i < titles.total; i++)
 232         {
 233 //printf("AudioALSA::translate_name %s %s\n", titles.values[i], pcm_titles.values[i]);
 234                 if(!strcasecmp(titles.values[i], input))
 235                 {
 236                         strcpy(output, pcm_titles.values[i]);
 237                         break;
 238                 }
 239         }
 240
 241         titles.remove_all_objects();
 242         pcm_titles.remove_all_objects();
 243 }
 244
 245 snd_pcm_format_t AudioALSA::translate_format(int format)
 246 {
 247         switch(format)
 248         {
 249         case 8:
 250                 return SND_PCM_FORMAT_S8;
 251                 break;
 252         case 16:
 253                 return SND_PCM_FORMAT_S16_LE;
 254                 break;
 255         case 24:
 256                 return SND_PCM_FORMAT_S24_LE;
 257                 break;
 258         case 32:
 259                 return SND_PCM_FORMAT_S32_LE;
 260                 break;
 261         }
 262         return SND_PCM_FORMAT_UNKNOWN;
 263 }
 264
 265 int AudioALSA::set_params(snd_pcm_t *dsp, int mode,
 266         int channels, int bits, int samplerate, int samples)
 267 {
 268         snd_pcm_hw_params_t *params;
 269         snd_pcm_sw_params_t *swparams;
 270         int err;
 271
 272         snd_pcm_hw_params_alloca(&params);
 273         snd_pcm_sw_params_alloca(&swparams);
 274         err = snd_pcm_hw_params_any(dsp, params);
 275
 276         if (err < 0) {
 277                 fprintf(stderr, "AudioALSA::set_params: ");
 278                 fprintf(stderr, _("no PCM configurations available\n"));
 279                 return 1;
 280         }
 281
 282         err=snd_pcm_hw_params_set_access(dsp,
 283                 params,
 284                 SND_PCM_ACCESS_RW_INTERLEAVED);
 285         if(err) {
 286                 fprintf(stderr, "AudioALSA::set_params: ");
 287                 fprintf(stderr, _("failed to set up interleaved device access.\n"));
 288                 return 1;
 289         }
 290
 291         err=snd_pcm_hw_params_set_format(dsp,
 292                 params,
 293                 translate_format(bits));
 294         if(err) {
 295                 fprintf(stderr, "AudioALSA::set_params: ");
 296                 fprintf(stderr, _("failed to set output format.\n"));
 297                 return 1;
 298         }
 299
 300         err=snd_pcm_hw_params_set_channels(dsp,
 301                 params,
 302                 channels);
 303         if(err) {
 304                 fprintf(stderr, "AudioALSA::set_params: ");
 305                 fprintf(stderr, _("Configured ALSA device does not support %d channel operation.\n"),
 306                         channels);
 307                 return 1;
 308         }
 309
 310         err=snd_pcm_hw_params_set_rate_near(dsp,
 311                 params,
 312                 (unsigned int*)&samplerate,
 313                 (int*)0);
 314         if(err) {
 315                 fprintf(stderr, "AudioALSA::set_params: ");
 316                 fprintf(stderr, _(" Configured ALSA device does not support %u Hz playback.\n"),
 317                         (unsigned int)samplerate);
 318                 return 1;
 319         }
 320
 321 // Buffers written must be equal to period_time
 322         int buffer_time = 0;
 323         int period_time = (int)(1000000 * (double)samples / samplerate);
 324         switch( mode ) {
 325         case MODERECORD:
 326                 buffer_time = 10000000;
 327                 break;
 328         case MODEPLAY:
 329                 buffer_time = 2 * period_time;
 330                 break;
 331         }
 332
 333 //printf("AudioALSA::set_params 1 %d %d %d\n", samples, buffer_time, period_time);
 334         snd_pcm_hw_params_set_buffer_time_near(dsp,
 335                 params,
 336                 (unsigned int*)&buffer_time,
 337                 (int*)0);
 338         snd_pcm_hw_params_set_period_time_near(dsp,
 339                 params,
 340                 (unsigned int*)&period_time,
 341                 (int*)0);
 342 //printf("AudioALSA::set_params 5 %d %d\n", buffer_time, period_time);
 343         err = snd_pcm_hw_params(dsp, params);
 344         if(err < 0) {
 345                 fprintf(stderr, "AudioALSA::set_params: hw_params failed\n");
 346                 return 1;
 347         }
 348
 349         snd_pcm_uframes_t chunk_size = 1024;
 350         snd_pcm_uframes_t buffer_size = 262144;
 351         snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
 352         snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
 353 //printf("AudioALSA::set_params 10 %d %d\n", chunk_size, buffer_size);
 354
 355         snd_pcm_sw_params_current(dsp, swparams);
 356         //snd_pcm_uframes_t xfer_align = 1;
 357         //snd_pcm_sw_params_get_xfer_align(swparams, &xfer_align);
 358         //unsigned int sleep_min = 0;
 359         //err = snd_pcm_sw_params_set_sleep_min(dsp, swparams, sleep_min);
 360         period_size = chunk_size;
 361         err = snd_pcm_sw_params_set_avail_min(dsp, swparams, period_size);
 362         //err = snd_pcm_sw_params_set_xfer_align(dsp, swparams, xfer_align);
 363         if(snd_pcm_sw_params(dsp, swparams) < 0) {
 364                 /* we can continue staggering along even if this fails */
 365                 fprintf(stderr, "AudioALSA::set_params: snd_pcm_sw_params failed\n");
 366         }
 367
 368         device->device_buffer = samples * bits / 8 * channels;
 369         period_size /= 2;
 370 //printf("AudioALSA::set_params 100 %d %d\n", samples,  device->device_buffer);
 371
 372 //      snd_pcm_hw_params_free(params);
 373 //      snd_pcm_sw_params_free(swparams);
 374         return 0;
 375 }
 376
 377 int AudioALSA::open_input()
 378 {
 379         char pcm_name[BCTEXTLEN];
 380         snd_pcm_stream_t stream = SND_PCM_STREAM_CAPTURE;
 381         int open_mode = 0;
 382         int err;
 383
 384         device->in_channels = device->get_ichannels();
 385         device->in_bits = device->in_config->alsa_in_bits;
 386
 387         translate_name(pcm_name, device->in_config->alsa_in_device,MODERECORD);
 388 //printf("AudioALSA::open_input %s\n", pcm_name);
 389
 390         err = snd_pcm_open(&dsp_in, pcm_name, stream, open_mode);
 391
 392         if(err < 0) {
 393                 dsp_in = 0;
 394                 printf("AudioALSA::open_input: %s\n", snd_strerror(err));
 395                 return 1;
 396         }
 397
 398         err = set_params(dsp_in, MODERECORD,
 399                 device->get_ichannels(),
 400                 device->in_config->alsa_in_bits,
 401                 device->in_samplerate,
 402                 device->in_samples);
 403         if(err) {
 404                 fprintf(stderr, "AudioALSA::open_input: set_params failed.  Aborting sampling.\n");
 405                 close_input();
 406                 return 1;
 407         }
 408
 409         return 0;
 410 }
 411
 412 int AudioALSA::open_output()
 413 {
 414         char pcm_name[BCTEXTLEN];
 415         snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
 416         int open_mode = SND_PCM_NONBLOCK;
 417         int err;
 418
 419         device->out_channels = device->get_ochannels();
 420         device->out_bits = device->out_config->alsa_out_bits;
 421
 422 //printf("AudioALSA::open_output out_device %s\n", device->out_config->alsa_out_device);
 423         translate_name(pcm_name, device->out_config->alsa_out_device,MODEPLAY);
 424 //printf("AudioALSA::open_output pcm_name %s\n", pcm_name);
 425
 426         err = snd_pcm_open(&dsp_out, pcm_name, stream, open_mode);
 427
 428         if(err < 0)
 429         {
 430                 dsp_out = 0;
 431                 printf("AudioALSA::open_output %s: %s\n", pcm_name, snd_strerror(err));
 432                 return 1;
 433         }
 434
 435         err = set_params(dsp_out, MODEPLAY,
 436                 device->get_ochannels(),
 437                 device->out_config->alsa_out_bits,
 438                 device->out_samplerate,
 439                 device->out_samples);
 440         if(err) {
 441                 fprintf(stderr, "AudioALSA::open_output: set_params failed.  Aborting playback.\n");
 442                 close_output();
 443                 return 1;
 444         }
 445
 446         timer->update();
 447         return 0;
 448 }
 449
 450 int AudioALSA::stop_output()
 451 {
 452 //printf("AudioALSA::stop_output\n");
 453         if(!device->out_config->interrupt_workaround)
 454         {
 455                 if( get_output() )
 456                         snd_pcm_drop(get_output());
 457         }
 458         else
 459                 flush_device();
 460         return 0;
 461 }
 462
 463 int AudioALSA::close_output()
 464 {
 465 //printf("AudioALSA::close_output\n");
 466         if(device->w && dsp_out) {
 467                 stop_output();
 468                 snd_pcm_close(dsp_out);
 469                 dsp_out = 0;
 470         }
 471         return 0;
 472 }
 473
 474 int AudioALSA::close_input()
 475 {
 476 //printf("AudioALSA::close_input\n");
 477         if(device->r && dsp_in) {
 478 //              snd_pcm_reset(dsp_in);
 479                 snd_pcm_drop(dsp_in);
 480                 snd_pcm_drain(dsp_in);
 481                 snd_pcm_close(dsp_in);
 482                 dsp_in = 0;
 483         }
 484         return 0;
 485 }
 486
 487 int AudioALSA::close_all()
 488 {
 489 //printf("AudioALSA::close_all\n");
 490         close_input();
 491         close_output();
 492         buffer_position = 0;
 493         samples_written = 0;
 494         delay = 0;
 495         interrupted = 0;
 496         return 0;
 497 }
 498
 499 // Undocumented
 500 int64_t AudioALSA::device_position()
 501 {
 502         timer_lock->lock("AudioALSA::device_position");
 503         int64_t delta = timer->get_scaled_difference(device->out_samplerate);
 504         int64_t result = buffer_position - delay + delta;
 505 //printf("AudioALSA::device_position 1 w=%jd dt=%jd dly=%d pos=%jd\n",
 506 // buffer_position, delta, delay, result);
 507         timer_lock->unlock();
 508         return result;
 509 }
 510
 511 int AudioALSA::read_buffer(char *buffer, int size)
 512 {
 513 //printf("AudioALSA::read_buffer 1\n");
 514         int attempts = 0;
 515         int done = 0;
 516         int frame_size = (device->in_bits / 8) * device->get_ichannels();
 517         int result = 0;
 518
 519         if(!get_input())
 520         {
 521                 sleep(1);
 522                 return 0;
 523         }
 524
 525         while(attempts < 1 && !done)
 526         {
 527                 snd_pcm_uframes_t frames = size / frame_size;
 528                 result = snd_pcm_readi(get_input(), buffer, frames);
 529                 if( result < 0)
 530                 {
 531                         printf("AudioALSA::read_buffer overrun at sample %jd\n",
 532                                 device->total_samples_read);
 533 //                      snd_pcm_resume(get_input());
 534                         close_input();  open_input();
 535                         attempts++;
 536                 }
 537                 else
 538                         done = 1;
 539 //printf("AudioALSA::read_buffer %d result=%d done=%d\n", __LINE__, result, done);
 540         }
 541         return 0;
 542 }
 543
 544 int AudioALSA::write_buffer(char *buffer, int size)
 545 {
 546 //printf("AudioALSA::write_buffer %d\n",size);
 547 // Don't give up and drop the buffer on the first error.
 548         int attempts = 0;
 549         int done = 0;
 550         int sample_size = (device->out_bits / 8) * device->get_ochannels();
 551         int samples = size / sample_size;
 552 //printf("AudioALSA::write_buffer %d\n",samples);
 553         int count = samples;
 554         snd_pcm_sframes_t delay = 0;
 555
 556 // static FILE *debug_fd = 0;
 557 // if(!debug_fd)
 558 // {
 559 //      debug_fd = fopen("/tmp/debug.pcm", "w");
 560 // }
 561 // fwrite(buffer, size, 1, debug_fd);
 562 // fflush(debug_fd);
 563
 564
 565         if(!get_output()) return 0;
 566         if( buffer_position == 0 )
 567                 timer->update();
 568
 569         AudioThread *audio_out = device->audio_out;
 570         while( count > 0 && attempts < 2 && !done ) {
 571                 if( device->playback_interrupted ) break;
 572 // Buffers written must be equal to period_time
 573                 audio_out->Thread::enable_cancel();
 574                 int ret = snd_pcm_avail_update(get_output());
 575                 if( ret >= period_size ) {
 576                         if( ret > count ) ret = count;
 577 //FILE *alsa_fp = 0;
 578 //if( !alsa_fp ) alsa_fp = fopen("/tmp/alsa.raw","w");
 579 //if( alsa_fp ) fwrite(buffer, sample_size, ret, alsa_fp);
 580 //printf("AudioALSA::snd_pcm_writei start %d\n",count);
 581                         ret = snd_pcm_writei(get_output(),buffer,ret);
 582 //printf("AudioALSA::snd_pcm_writei done %d\n", ret);
 583                 }
 584                 else if( ret >= 0 || ret == -EAGAIN ) {
 585                         ret = snd_pcm_wait(get_output(),15);
 586                         if( ret > 0 ) ret = 0;
 587                 }
 588                 audio_out->Thread::disable_cancel();
 589                 if( device->playback_interrupted ) break;
 590                 if( ret == 0 ) continue;
 591
 592                 if( ret > 0 ) {
 593                         samples_written += ret;
 594                         if( (count-=ret) > 0 ) {
 595                                 buffer += ret * sample_size;
 596                                 attempts = 0;
 597                         }
 598                         else
 599                                 done = 1;
 600                 }
 601                 else {
 602                         printf("AudioALSA::write_buffer err %d(%s) at sample %jd\n",
 603                                 ret, snd_strerror(ret), device->current_position());
 604                         Timer::delay(50);
 605 //                      snd_pcm_resume(get_output());
 606                         snd_pcm_recover(get_output(), ret, 1);
 607 //                      close_output();  open_output();
 608                         attempts++;
 609                 }
 610         }
 611
 612         if( !interrupted && device->playback_interrupted )
 613         {
 614                 interrupted = 1;
 615 //printf("AudioALSA::write_buffer interrupted\n");
 616                 stop_output();
 617         }
 618
 619         if(done)
 620         {
 621                 timer_lock->lock("AudioALSA::write_buffer");
 622                 snd_pcm_delay(get_output(), &delay);
 623                 this->delay = delay;
 624                 timer->update();
 625                 buffer_position += samples;
 626 //printf("AudioALSA::write_buffer ** wrote %d, delay %d\n",samples,(int)delay);
 627                 timer_lock->unlock();
 628         }
 629         return 0;
 630 }
 631
 632 //this delay seems to prevent a problem where the sound system outputs
 633 //a lot of silence while waiting for the device drain to happen.
 634 int AudioALSA::output_wait()
 635 {
 636         snd_pcm_sframes_t delay = 0;
 637         snd_pcm_delay(get_output(), &delay);
 638         if( delay <= 0 ) return 0;
 639         int64_t udelay = 1e6 * delay / device->out_samplerate;
 640         // don't allow more than 10 seconds
 641         if( udelay > 10000000 ) udelay = 10000000;
 642         while( udelay > 0 && !device->playback_interrupted ) {
 643                 int64_t usecs = udelay;
 644                 if( usecs > 100000 ) usecs = 100000;
 645                 usleep(usecs);
 646                 udelay -= usecs;
 647         }
 648         if( device->playback_interrupted &&
 649             !device->out_config->interrupt_workaround )
 650                 snd_pcm_drop(get_output());
 651         return 0;
 652 }
 653
 654 int AudioALSA::flush_device()
 655 {
 656 //printf("AudioALSA::flush_device\n");
 657         if(get_output())
 658         {
 659                 output_wait();
 660                 //this causes the output to stutter.
 661                 //snd_pcm_nonblock(get_output(), 0);
 662                 snd_pcm_drain(get_output());
 663                 //snd_pcm_nonblock(get_output(), 1);
 664         }
 665         return 0;
 666 }
 667
 668 int AudioALSA::interrupt_playback()
 669 {
 670 //printf("AudioALSA::interrupt_playback *********\n");
 671 //      if(get_output())
 672 //      {
 673 // Interrupts the playback but may not have caused snd_pcm_writei to exit.
 674 // With some soundcards it causes snd_pcm_writei to freeze for a few seconds.
 675 //              if(!device->out_config->interrupt_workaround)
 676 //                      snd_pcm_drop(get_output());
 677
 678 // Makes sure the current buffer finishes before stopping.
 679 //              snd_pcm_drain(get_output());
 680
 681 // The only way to ensure snd_pcm_writei exits, but
 682 // got a lot of crashes when doing this.
 683 //              device->Thread::cancel();
 684 //      }
 685         return 0;
 686 }
 687
 688
 689 snd_pcm_t* AudioALSA::get_output()
 690 {
 691         return dsp_out;
 692 }
 693
 694 snd_pcm_t* AudioALSA::get_input()
 695 {
 696         return dsp_in;
 697 }
 698
 699 #endif