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