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