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:
 238                 return SND_PCM_FORMAT_S8;
 239                 break;
 240         case 16:
 241                 return SND_PCM_FORMAT_S16_LE;
 242                 break;
 243         case 24:
 244                 return SND_PCM_FORMAT_S24_LE;
 245                 break;
 246         case 32:
 247                 return SND_PCM_FORMAT_S32_LE;
 248                 break;
 249         }
 250         return SND_PCM_FORMAT_UNKNOWN;
 251 }
 252
 253 int AudioALSA::set_params(snd_pcm_t *dsp, int mode,
 254         int channels, int bits, int samplerate, int samples)
 255 {
 256         snd_pcm_hw_params_t *params;
 257         snd_pcm_sw_params_t *swparams;
 258         int err;
 259
 260         snd_pcm_hw_params_alloca(&params);
 261         snd_pcm_sw_params_alloca(&swparams);
 262         err = snd_pcm_hw_params_any(dsp, params);
 263
 264         if (err < 0) {
 265                 fprintf(stderr, "AudioALSA::set_params: ");
 266                 fprintf(stderr, _("no PCM configurations available\n"));
 267                 return 1;
 268         }
 269
 270         err=snd_pcm_hw_params_set_access(dsp,
 271                 params,
 272                 SND_PCM_ACCESS_RW_INTERLEAVED);
 273         if(err) {
 274                 fprintf(stderr, "AudioALSA::set_params: ");
 275                 fprintf(stderr, _("failed to set up interleaved device access.\n"));
 276                 return 1;
 277         }
 278
 279         err=snd_pcm_hw_params_set_format(dsp,
 280                 params,
 281                 translate_format(bits));
 282         if(err) {
 283                 fprintf(stderr, "AudioALSA::set_params: ");
 284                 fprintf(stderr, _("failed to set output format.\n"));
 285                 return 1;
 286         }
 287
 288         err=snd_pcm_hw_params_set_channels(dsp,
 289                 params,
 290                 channels);
 291         if(err) {
 292                 fprintf(stderr, "AudioALSA::set_params: ");
 293                 fprintf(stderr, _("Configured ALSA device does not support %d channel operation.\n"),
 294                         channels);
 295                 return 1;
 296         }
 297
 298         err=snd_pcm_hw_params_set_rate_near(dsp,
 299                 params,
 300                 (unsigned int*)&samplerate,
 301                 (int*)0);
 302         if(err) {
 303                 fprintf(stderr, "AudioALSA::set_params: ");
 304                 fprintf(stderr, _(" Configured ALSA device does not support %u Hz playback.\n"),
 305                         (unsigned int)samplerate);
 306                 return 1;
 307         }
 308
 309 // Buffers written must be equal to period_time
 310         int buffer_time = 0;
 311         int period_time = (int)(1000000 * (double)samples / samplerate);
 312         switch( mode ) {
 313         case MODERECORD:
 314                 buffer_time = 10000000;
 315                 break;
 316         case MODEPLAY:
 317                 buffer_time = 2 * period_time;
 318                 break;
 319         }
 320
 321 //printf("AudioALSA::set_params 1 %d %d %d\n", samples, buffer_time, period_time);
 322         snd_pcm_hw_params_set_buffer_time_near(dsp,
 323                 params,
 324                 (unsigned int*)&buffer_time,
 325                 (int*)0);
 326         snd_pcm_hw_params_set_period_time_near(dsp,
 327                 params,
 328                 (unsigned int*)&period_time,
 329                 (int*)0);
 330 //printf("AudioALSA::set_params 5 %d %d\n", buffer_time, period_time);
 331         err = snd_pcm_hw_params(dsp, params);
 332         if(err < 0) {
 333                 fprintf(stderr, "AudioALSA::set_params: hw_params failed\n");
 334                 return 1;
 335         }
 336
 337         snd_pcm_uframes_t chunk_size = 1024;
 338         snd_pcm_uframes_t buffer_size = 262144;
 339         snd_pcm_hw_params_get_period_size(params, &chunk_size, 0);
 340         snd_pcm_hw_params_get_buffer_size(params, &buffer_size);
 341 //printf("AudioALSA::set_params 10 %d %d\n", chunk_size, buffer_size);
 342
 343         snd_pcm_sw_params_current(dsp, swparams);
 344         //snd_pcm_uframes_t xfer_align = 1;
 345         //snd_pcm_sw_params_get_xfer_align(swparams, &xfer_align);
 346         //unsigned int sleep_min = 0;
 347         //err = snd_pcm_sw_params_set_sleep_min(dsp, swparams, sleep_min);
 348         period_size = chunk_size;
 349         err = snd_pcm_sw_params_set_avail_min(dsp, swparams, period_size);
 350         //err = snd_pcm_sw_params_set_xfer_align(dsp, swparams, xfer_align);
 351         if(snd_pcm_sw_params(dsp, swparams) < 0) {
 352                 /* we can continue staggering along even if this fails */
 353                 fprintf(stderr, "AudioALSA::set_params: snd_pcm_sw_params failed\n");
 354         }
 355
 356         device->device_buffer = samples * bits / 8 * channels;
 357         period_size /= 2;
 358 //printf("AudioALSA::set_params 100 %d %d\n", samples,  device->device_buffer);
 359
 360 //      snd_pcm_hw_params_free(params);
 361 //      snd_pcm_sw_params_free(swparams);
 362         return 0;
 363 }
 364
 365 int AudioALSA::open_input()
 366 {
 367         char pcm_name[BCTEXTLEN];
 368         snd_pcm_stream_t stream = SND_PCM_STREAM_CAPTURE;
 369         int open_mode = 0;
 370         int err;
 371
 372         device->in_channels = device->get_ichannels();
 373         device->in_bits = device->in_config->alsa_in_bits;
 374
 375         translate_name(pcm_name, device->in_config->alsa_in_device,MODERECORD);
 376 //printf("AudioALSA::open_input %s\n", pcm_name);
 377
 378         err = snd_pcm_open(&dsp_in, pcm_name, stream, open_mode);
 379
 380         if(err < 0) {
 381                 dsp_in = 0;
 382                 printf("AudioALSA::open_input: %s\n", snd_strerror(err));
 383                 return 1;
 384         }
 385
 386         err = set_params(dsp_in, MODERECORD,
 387                 device->get_ichannels(),
 388                 device->in_config->alsa_in_bits,
 389                 device->in_samplerate,
 390                 device->in_samples);
 391         if(err) {
 392                 fprintf(stderr, "AudioALSA::open_input: set_params failed.  Aborting sampling.\n");
 393                 close_input();
 394                 return 1;
 395         }
 396
 397         return 0;
 398 }
 399
 400 int AudioALSA::open_output()
 401 {
 402         char pcm_name[BCTEXTLEN];
 403         snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
 404         int open_mode = SND_PCM_NONBLOCK;
 405         int err;
 406
 407         device->out_channels = device->get_ochannels();
 408         device->out_bits = device->out_config->alsa_out_bits;
 409
 410 //printf("AudioALSA::open_output out_device %s\n", device->out_config->alsa_out_device);
 411         translate_name(pcm_name, device->out_config->alsa_out_device,MODEPLAY);
 412 //printf("AudioALSA::open_output pcm_name %s\n", pcm_name);
 413
 414         err = snd_pcm_open(&dsp_out, pcm_name, stream, open_mode);
 415
 416         if(err < 0)
 417         {
 418                 dsp_out = 0;
 419                 printf("AudioALSA::open_output %s: %s\n", pcm_name, snd_strerror(err));
 420                 return 1;
 421         }
 422
 423         err = set_params(dsp_out, MODEPLAY,
 424                 device->get_ochannels(),
 425                 device->out_config->alsa_out_bits,
 426                 device->out_samplerate,
 427                 device->out_samples);
 428         if(err) {
 429                 fprintf(stderr, "AudioALSA::open_output: set_params failed.  Aborting playback.\n");
 430                 close_output();
 431                 return 1;
 432         }
 433
 434         timer->update();
 435         return 0;
 436 }
 437
 438 int AudioALSA::stop_output()
 439 {
 440 //printf("AudioALSA::stop_output\n");
 441         if(!device->out_config->interrupt_workaround)
 442         {
 443                 if( get_output() )
 444                         snd_pcm_drop(get_output());
 445         }
 446         else
 447                 flush_device();
 448         return 0;
 449 }
 450
 451 int AudioALSA::close_output()
 452 {
 453 //printf("AudioALSA::close_output\n");
 454         if(device->w && dsp_out) {
 455                 stop_output();
 456                 snd_pcm_close(dsp_out);
 457                 dsp_out = 0;
 458         }
 459         return 0;
 460 }
 461
 462 int AudioALSA::close_input()
 463 {
 464 //printf("AudioALSA::close_input\n");
 465         if(device->r && dsp_in) {
 466 //              snd_pcm_reset(dsp_in);
 467                 snd_pcm_drop(dsp_in);
 468                 snd_pcm_drain(dsp_in);
 469                 snd_pcm_close(dsp_in);
 470                 dsp_in = 0;
 471         }
 472         return 0;
 473 }
 474
 475 int AudioALSA::close_all()
 476 {
 477 //printf("AudioALSA::close_all\n");
 478         close_input();
 479         close_output();
 480         buffer_position = 0;
 481         samples_written = 0;
 482         delay = 0;
 483         interrupted = 0;
 484         return 0;
 485 }
 486
 487 // Undocumented
 488 int64_t AudioALSA::device_position()
 489 {
 490         timer_lock->lock("AudioALSA::device_position");
 491         int64_t delta = timer->get_scaled_difference(device->out_samplerate);
 492         int64_t result = buffer_position - delay + delta;
 493 //printf("AudioALSA::device_position 1 w=%jd dt=%jd dly=%d pos=%jd\n",
 494 // buffer_position, delta, delay, result);
 495         timer_lock->unlock();
 496         return result;
 497 }
 498
 499 int AudioALSA::read_buffer(char *buffer, int size)
 500 {
 501 //printf("AudioALSA::read_buffer 1\n");
 502         int attempts = 0;
 503         int done = 0;
 504         int frame_size = (device->in_bits / 8) * device->get_ichannels();
 505         int result = 0;
 506
 507         if(!get_input())
 508         {
 509                 sleep(1);
 510                 return 0;
 511         }
 512
 513         while(attempts < 1 && !done)
 514         {
 515                 snd_pcm_uframes_t frames = size / frame_size;
 516                 result = snd_pcm_readi(get_input(), buffer, frames);
 517                 if( result < 0)
 518                 {
 519                         printf("AudioALSA::read_buffer overrun at sample %jd\n",
 520                                 device->total_samples_read);
 521 //                      snd_pcm_resume(get_input());
 522                         close_input();  open_input();
 523                         attempts++;
 524                 }
 525                 else
 526                         done = 1;
 527 //printf("AudioALSA::read_buffer %d result=%d done=%d\n", __LINE__, result, done);
 528         }
 529         return 0;
 530 }
 531
 532 int AudioALSA::write_buffer(char *buffer, int size)
 533 {
 534 //printf("AudioALSA::write_buffer %d\n",size);
 535 // Don't give up and drop the buffer on the first error.
 536         int attempts = 0;
 537         int done = 0;
 538         int sample_size = (device->out_bits / 8) * device->get_ochannels();
 539         int samples = size / sample_size;
 540 //printf("AudioALSA::write_buffer %d\n",samples);
 541         int count = samples;
 542         snd_pcm_sframes_t delay = 0;
 543
 544 // static FILE *debug_fd = 0;
 545 // if(!debug_fd)
 546 // {
 547 //      debug_fd = fopen("/tmp/debug.pcm", "w");
 548 // }
 549 // fwrite(buffer, size, 1, debug_fd);
 550 // fflush(debug_fd);
 551
 552
 553         if(!get_output()) return 0;
 554         if( buffer_position == 0 )
 555                 timer->update();
 556
 557         AudioThread *audio_out = device->audio_out;
 558         while( count > 0 && attempts < 2 && !done ) {
 559                 if( device->playback_interrupted ) break;
 560 // Buffers written must be equal to period_time
 561                 audio_out->Thread::enable_cancel();
 562                 int ret = snd_pcm_avail_update(get_output());
 563                 if( ret >= period_size ) {
 564                         if( ret > count ) ret = count;
 565 //FILE *alsa_fp = 0;
 566 //if( !alsa_fp ) alsa_fp = fopen("/tmp/alsa.raw","w");
 567 //if( alsa_fp ) fwrite(buffer, sample_size, ret, alsa_fp);
 568 //printf("AudioALSA::snd_pcm_writei start %d\n",count);
 569                         ret = snd_pcm_writei(get_output(),buffer,ret);
 570 //printf("AudioALSA::snd_pcm_writei done %d\n", ret);
 571                 }
 572                 else if( ret >= 0 || ret == -EAGAIN ) {
 573                         ret = snd_pcm_wait(get_output(),15);
 574                         if( ret > 0 ) ret = 0;
 575                 }
 576                 audio_out->Thread::disable_cancel();
 577                 if( device->playback_interrupted ) break;
 578                 if( ret == 0 ) continue;
 579
 580                 if( ret > 0 ) {
 581                         samples_written += ret;
 582                         if( (count-=ret) > 0 ) {
 583                                 buffer += ret * sample_size;
 584                                 attempts = 0;
 585                         }
 586                         else
 587                                 done = 1;
 588                 }
 589                 else {
 590                         printf("AudioALSA::write_buffer err %d(%s) at sample %jd\n",
 591                                 ret, snd_strerror(ret), device->current_position());
 592                         Timer::delay(50);
 593 //                      snd_pcm_resume(get_output());
 594                         snd_pcm_recover(get_output(), ret, 1);
 595 //                      close_output();  open_output();
 596                         attempts++;
 597                 }
 598         }
 599
 600         if( !interrupted && device->playback_interrupted )
 601         {
 602                 interrupted = 1;
 603 //printf("AudioALSA::write_buffer interrupted\n");
 604                 stop_output();
 605         }
 606
 607         if(done)
 608         {
 609                 timer_lock->lock("AudioALSA::write_buffer");
 610                 snd_pcm_delay(get_output(), &delay);
 611                 this->delay = delay;
 612                 timer->update();
 613                 buffer_position += samples;
 614 //printf("AudioALSA::write_buffer ** wrote %d, delay %d\n",samples,(int)delay);
 615                 timer_lock->unlock();
 616         }
 617         return 0;
 618 }
 619
 620 //this delay seems to prevent a problem where the sound system outputs
 621 //a lot of silence while waiting for the device drain to happen.
 622 int AudioALSA::output_wait()
 623 {
 624         snd_pcm_sframes_t delay = 0;
 625         snd_pcm_delay(get_output(), &delay);
 626         if( delay <= 0 ) return 0;
 627         int64_t udelay = 1e6 * delay / device->out_samplerate;
 628         // don't allow more than 10 seconds
 629         if( udelay > 10000000 ) udelay = 10000000;
 630         while( udelay > 0 && !device->playback_interrupted ) {
 631                 int64_t usecs = udelay;
 632                 if( usecs > 100000 ) usecs = 100000;
 633                 usleep(usecs);
 634                 udelay -= usecs;
 635         }
 636         if( device->playback_interrupted &&
 637             !device->out_config->interrupt_workaround )
 638                 snd_pcm_drop(get_output());
 639         return 0;
 640 }
 641
 642 int AudioALSA::flush_device()
 643 {
 644 //printf("AudioALSA::flush_device\n");
 645         if(get_output())
 646         {
 647                 output_wait();
 648                 //this causes the output to stutter.
 649                 //snd_pcm_nonblock(get_output(), 0);
 650                 snd_pcm_drain(get_output());
 651                 //snd_pcm_nonblock(get_output(), 1);
 652         }
 653         return 0;
 654 }
 655
 656 int AudioALSA::interrupt_playback()
 657 {
 658 //printf("AudioALSA::interrupt_playback *********\n");
 659 //      if(get_output())
 660 //      {
 661 // Interrupts the playback but may not have caused snd_pcm_writei to exit.
 662 // With some soundcards it causes snd_pcm_writei to freeze for a few seconds.
 663 //              if(!device->out_config->interrupt_workaround)
 664 //                      snd_pcm_drop(get_output());
 665
 666 // Makes sure the current buffer finishes before stopping.
 667 //              snd_pcm_drain(get_output());
 668
 669 // The only way to ensure snd_pcm_writei exits, but
 670 // got a lot of crashes when doing this.
 671 //              device->Thread::cancel();
 672 //      }
 673         return 0;
 674 }
 675
 676
 677 snd_pcm_t* AudioALSA::get_output()
 678 {
 679         return dsp_out;
 680 }
 681
 682 snd_pcm_t* AudioALSA::get_input()
 683 {
 684         return dsp_in;
 685 }
 686
 687 #endif