libao2/ao_sun.c

   1 #include <stdio.h>
   2 #include <stdlib.h>
   3 #include <string.h>
   4
   5 #include <unistd.h>
   6 #include <fcntl.h>
   7 #include <errno.h>
   8 #include <sys/ioctl.h>
   9 #include <sys/time.h>
  10 #include <sys/types.h>
  11 #include <sys/stat.h>
  12 #include <sys/audioio.h>
  13 #ifdef  AUDIO_SWFEATURE_MIXER   /* solaris8 or newer? */
  14 # define HAVE_SYS_MIXER_H 1
  15 #endif
  16 #if     HAVE_SYS_MIXER_H
  17 # include <sys/mixer.h>
  18 #endif
  19 #ifdef  __svr4__
  20 #include <stropts.h>
  21 #endif
  22
  23 #include "config.h"
  24 #include "mixer.h"
  25
  26 #include "audio_out.h"
  27 #include "audio_out_internal.h"
  28 #include "libaf/af_format.h"
  29 #include "mp_msg.h"
  30 #include "help_mp.h"
  31
  32 static ao_info_t info =
  33 {
  34     "Sun audio output",
  35     "sun",
  36     "Juergen Keil",
  37     ""
  38 };
  39
  40 LIBAO_EXTERN(sun)
  41
  42
  43 /* These defines are missing on NetBSD */
  44 #ifndef AUDIO_PRECISION_8
  45 #define AUDIO_PRECISION_8       8
  46 #define AUDIO_PRECISION_16      16
  47 #endif
  48 #ifndef AUDIO_CHANNELS_MONO
  49 #define AUDIO_CHANNELS_MONO     1
  50 #define AUDIO_CHANNELS_STEREO   2
  51 #endif
  52
  53
  54 static char *sun_mixer_device = NULL;
  55 static char *audio_dev = NULL;
  56 static int queued_bursts = 0;
  57 static int queued_samples = 0;
  58 static int bytes_per_sample = 0;
  59 static int byte_per_sec = 0;
  60 static int audio_fd = -1;
  61 static enum {
  62     RTSC_UNKNOWN = 0,
  63     RTSC_ENABLED,
  64     RTSC_DISABLED
  65 } enable_sample_timing;
  66
  67
  68 static void flush_audio(int fd) {
  69 #ifdef AUDIO_FLUSH
  70   ioctl(fd, AUDIO_FLUSH, 0);
  71 #elif defined(__svr4__)
  72   ioctl(fd, I_FLUSH, FLUSHW);
  73 #endif
  74 }
  75
  76 // convert an OSS audio format specification into a sun audio encoding
  77 static int af2sunfmt(int format)
  78 {
  79     switch (format){
  80     case AF_FORMAT_MU_LAW:
  81         return AUDIO_ENCODING_ULAW;
  82     case AF_FORMAT_A_LAW:
  83         return AUDIO_ENCODING_ALAW;
  84     case AF_FORMAT_S16_NE:
  85         return AUDIO_ENCODING_LINEAR;
  86 #ifdef  AUDIO_ENCODING_LINEAR8  // Missing on SunOS 5.5.1...
  87     case AF_FORMAT_U8:
  88         return AUDIO_ENCODING_LINEAR8;
  89 #endif
  90     case AF_FORMAT_S8:
  91         return AUDIO_ENCODING_LINEAR;
  92 #ifdef  AUDIO_ENCODING_DVI      // Missing on NetBSD...
  93     case AF_FORMAT_IMA_ADPCM:
  94         return AUDIO_ENCODING_DVI;
  95 #endif
  96     default:
  97         return AUDIO_ENCODING_NONE;
  98   }
  99 }
 100
 101 // try to figure out, if the soundcard driver provides usable (precise)
 102 // sample counter information
 103 static int realtime_samplecounter_available(char *dev)
 104 {
 105     int fd = -1;
 106     audio_info_t info;
 107     int rtsc_ok = RTSC_DISABLED;
 108     int len;
 109     void *silence = NULL;
 110     struct timeval start, end;
 111     struct timespec delay;
 112     int usec_delay;
 113     unsigned last_samplecnt;
 114     unsigned increment;
 115     unsigned min_increment;
 116
 117     len = 44100 * 4 / 4;    /* amount of data for 0.25sec of 44.1khz, stereo,
 118                              * 16bit.  44kbyte can be sent to all supported
 119                              * sun audio devices without blocking in the
 120                              * "write" below.
 121                              */
 122     silence = calloc(1, len);
 123     if (silence == NULL)
 124         goto error;
 125
 126     if ((fd = open(dev, O_WRONLY)) < 0)
 127         goto error;
 128
 129     AUDIO_INITINFO(&info);
 130     info.play.sample_rate = 44100;
 131     info.play.channels = AUDIO_CHANNELS_STEREO;
 132     info.play.precision = AUDIO_PRECISION_16;
 133     info.play.encoding = AUDIO_ENCODING_LINEAR;
 134     info.play.samples = 0;
 135     if (ioctl(fd, AUDIO_SETINFO, &info)) {
 136         if ( mp_msg_test(MSGT_AO,MSGL_V) )
 137             mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscSetinfoFailed);
 138         goto error;
 139     }
 140
 141     if (write(fd, silence, len) != len) {
 142         if ( mp_msg_test(MSGT_AO,MSGL_V) )
 143             mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_RtscWriteFailed);
 144         goto error;
 145     }
 146
 147     if (ioctl(fd, AUDIO_GETINFO, &info)) {
 148         if ( mp_msg_test(MSGT_AO,MSGL_V) )
 149             perror("rtsc: GETINFO1");
 150         goto error;
 151     }
 152
 153     last_samplecnt = info.play.samples;
 154     min_increment = ~0;
 155
 156     gettimeofday(&start, NULL);
 157     for (;;) {
 158         delay.tv_sec = 0;
 159         delay.tv_nsec = 10000000;
 160         nanosleep(&delay, NULL);
 161         gettimeofday(&end, NULL);
 162         usec_delay = (end.tv_sec - start.tv_sec) * 1000000
 163             + end.tv_usec - start.tv_usec;
 164
 165         // stop monitoring sample counter after 0.2 seconds
 166         if (usec_delay > 200000)
 167             break;
 168
 169         if (ioctl(fd, AUDIO_GETINFO, &info)) {
 170             if ( mp_msg_test(MSGT_AO,MSGL_V) )
 171                 perror("rtsc: GETINFO2 failed");
 172             goto error;
 173         }
 174         if (info.play.samples < last_samplecnt) {
 175             if ( mp_msg_test(MSGT_AO,MSGL_V) )
 176                 mp_msg(MSGT_AO,MSGL_V,"rtsc: %d > %d?\n", last_samplecnt, info.play.samples);
 177             goto error;
 178         }
 179
 180         if ((increment = info.play.samples - last_samplecnt) > 0) {
 181             if ( mp_msg_test(MSGT_AO,MSGL_V) )
 182                 mp_msg(MSGT_AO,MSGL_V,"ao_sun: sample counter increment: %d\n", increment);
 183             if (increment < min_increment) {
 184                 min_increment = increment;
 185                 if (min_increment < 2000)
 186                     break;      // looks good
 187             }
 188         }
 189         last_samplecnt = info.play.samples;
 190     }
 191
 192     /*
 193      * For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes
 194      * chunks (== 4096 samples) to the audio device.  If we see a minimum
 195      * sample counter increment from the soundcard driver of less than
 196      * 2000 samples,  we assume that the driver provides a useable realtime
 197      * sample counter in the AUDIO_INFO play.samples field.  Timing based
 198      * on sample counts should be much more accurate than counting whole
 199      * 16kbyte chunks.
 200      */
 201     if (min_increment < 2000)
 202         rtsc_ok = RTSC_ENABLED;
 203
 204     if ( mp_msg_test(MSGT_AO,MSGL_V) )
 205         mp_msg(MSGT_AO,MSGL_V,"ao_sun: minimum sample counter increment per 10msec interval: %d\n"
 206                "\t%susing sample counter based timing code\n",
 207                min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not ");
 208
 209
 210 error:
 211     if (silence != NULL) free(silence);
 212     if (fd >= 0) {
 213         // remove the 0 bytes from the above measurement from the
 214         // audio driver's STREAMS queue
 215         flush_audio(fd);
 216         close(fd);
 217     }
 218
 219     return rtsc_ok;
 220 }
 221
 222
 223 // match the requested sample rate |sample_rate| against the
 224 // sample rates supported by the audio device |dev|.  Return
 225 // a supported sample rate,  if that sample rate is close to
 226 // (< 1% difference) the requested rate; return 0 otherwise.
 227
 228 #define MAX_RATE_ERR    1
 229
 230 static unsigned
 231 find_close_samplerate_match(int dev, unsigned sample_rate)
 232 {
 233 #if     HAVE_SYS_MIXER_H
 234     am_sample_rates_t *sr;
 235     unsigned i, num, err, best_err, best_rate;
 236
 237     for (num = 16; num < 1024; num *= 2) {
 238         sr = malloc(AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(num));
 239         if (!sr)
 240             return 0;
 241         sr->type = AUDIO_PLAY;
 242         sr->flags = 0;
 243         sr->num_samp_rates = num;
 244         if (ioctl(dev, AUDIO_MIXER_GET_SAMPLE_RATES, sr)) {
 245             free(sr);
 246             return 0;
 247         }
 248         if (sr->num_samp_rates <= num)
 249             break;
 250         free(sr);
 251     }
 252
 253     if (sr->flags & MIXER_SR_LIMITS) {
 254         /*
 255          * HW can playback any rate between
 256          * sr->samp_rates[0] .. sr->samp_rates[1]
 257          */
 258         free(sr);
 259         return 0;
 260     } else {
 261         /* HW supports fixed sample rates only */
 262
 263         best_err = 65535;
 264         best_rate = 0;
 265
 266         for (i = 0; i < sr->num_samp_rates; i++) {
 267             err = abs(sr->samp_rates[i] - sample_rate);
 268             if (err == 0) {
 269                 /*
 270                  * exact supported sample rate match, no need to
 271                  * retry something else
 272                  */
 273                 best_rate = 0;
 274                 break;
 275             }
 276             if (err < best_err) {
 277                 best_err = err;
 278                 best_rate = sr->samp_rates[i];
 279             }
 280         }
 281
 282         free(sr);
 283
 284         if (best_rate > 0 && (100/MAX_RATE_ERR)*best_err < sample_rate) {
 285             /* found a supported sample rate with <1% error? */
 286             return best_rate;
 287         }
 288         return 0;
 289     }
 290 #else   /* old audioio driver, cannot return list of supported rates */
 291     /* XXX: hardcoded sample rates */
 292     unsigned i, err;
 293     unsigned audiocs_rates[] = {
 294         5510, 6620, 8000, 9600, 11025, 16000, 18900, 22050,
 295         27420, 32000, 33075, 37800, 44100, 48000, 0
 296     };
 297
 298     for (i = 0; audiocs_rates[i]; i++) {
 299         err = abs(audiocs_rates[i] - sample_rate);
 300         if (err == 0) {
 301             /*
 302              * exact supported sample rate match, no need to
 303              * retry something elise
 304              */
 305             return 0;
 306         }
 307         if ((100/MAX_RATE_ERR)*err < audiocs_rates[i]) {
 308             /* <1% error? */
 309             return audiocs_rates[i];
 310         }
 311     }
 312
 313     return 0;
 314 #endif
 315 }
 316
 317
 318 // return the highest sample rate supported by audio device |dev|.
 319 static unsigned
 320 find_highest_samplerate(int dev)
 321 {
 322 #if     HAVE_SYS_MIXER_H
 323     am_sample_rates_t *sr;
 324     unsigned i, num, max_rate;
 325
 326     for (num = 16; num < 1024; num *= 2) {
 327         sr = malloc(AUDIO_MIXER_SAMP_RATES_STRUCT_SIZE(num));
 328         if (!sr)
 329             return 0;
 330         sr->type = AUDIO_PLAY;
 331         sr->flags = 0;
 332         sr->num_samp_rates = num;
 333         if (ioctl(dev, AUDIO_MIXER_GET_SAMPLE_RATES, sr)) {
 334             free(sr);
 335             return 0;
 336         }
 337         if (sr->num_samp_rates <= num)
 338             break;
 339         free(sr);
 340     }
 341
 342     if (sr->flags & MIXER_SR_LIMITS) {
 343         /*
 344          * HW can playback any rate between
 345          * sr->samp_rates[0] .. sr->samp_rates[1]
 346          */
 347         max_rate = sr->samp_rates[1];
 348     } else {
 349         /* HW supports fixed sample rates only */
 350         max_rate = 0;
 351         for (i = 0; i < sr->num_samp_rates; i++) {
 352             if (sr->samp_rates[i] > max_rate)
 353                 max_rate = sr->samp_rates[i];
 354         }
 355     }
 356     free(sr);
 357     return max_rate;
 358
 359 #else   /* old audioio driver, cannot return list of supported rates */
 360     return 44100;       /* should be supported even on old ISA SB cards */
 361 #endif
 362 }
 363
 364
 365 static void setup_device_paths(void)
 366 {
 367     if (audio_dev == NULL) {
 368         if ((audio_dev = getenv("AUDIODEV")) == NULL)
 369             audio_dev = "/dev/audio";
 370     }
 371
 372     if (sun_mixer_device == NULL) {
 373         if ((sun_mixer_device = mixer_device) == NULL || !sun_mixer_device[0]) {
 374             sun_mixer_device = malloc(strlen(audio_dev) + 4);
 375             strcpy(sun_mixer_device, audio_dev);
 376             strcat(sun_mixer_device, "ctl");
 377         }
 378     }
 379
 380     if (ao_subdevice) audio_dev = ao_subdevice;
 381 }
 382
 383 // to set/get/query special features/parameters
 384 static int control(int cmd,void *arg){
 385     switch(cmd){
 386     case AOCONTROL_SET_DEVICE:
 387         audio_dev=(char*)arg;
 388         return CONTROL_OK;
 389     case AOCONTROL_QUERY_FORMAT:
 390         return CONTROL_TRUE;
 391     case AOCONTROL_GET_VOLUME:
 392     {
 393         int fd;
 394
 395         if ( !sun_mixer_device )    /* control function is used before init? */
 396             setup_device_paths();
 397
 398         fd=open( sun_mixer_device,O_RDONLY );
 399         if ( fd != -1 )
 400         {
 401             ao_control_vol_t *vol = (ao_control_vol_t *)arg;
 402             float volume;
 403             struct audio_info info;
 404             ioctl( fd,AUDIO_GETINFO,&info);
 405             volume = info.play.gain * 100. / AUDIO_MAX_GAIN;
 406             if ( info.play.balance == AUDIO_MID_BALANCE ) {
 407                 vol->right = vol->left = volume;
 408             } else if ( info.play.balance < AUDIO_MID_BALANCE ) {
 409                 vol->left  = volume;
 410                 vol->right = volume * info.play.balance / AUDIO_MID_BALANCE;
 411             } else {
 412                 vol->left  = volume * (AUDIO_RIGHT_BALANCE-info.play.balance)
 413                                                         / AUDIO_MID_BALANCE;
 414                 vol->right = volume;
 415             }
 416             close( fd );
 417             return CONTROL_OK;
 418         }
 419         return CONTROL_ERROR;
 420     }
 421     case AOCONTROL_SET_VOLUME:
 422     {
 423         ao_control_vol_t *vol = (ao_control_vol_t *)arg;
 424         int fd;
 425
 426         if ( !sun_mixer_device )    /* control function is used before init? */
 427             setup_device_paths();
 428
 429         fd=open( sun_mixer_device,O_RDONLY );
 430         if ( fd != -1 )
 431         {
 432             struct audio_info info;
 433             float volume;
 434             AUDIO_INITINFO(&info);
 435             volume = vol->right > vol->left ? vol->right : vol->left;
 436             if ( volume != 0 ) {
 437                 info.play.gain = volume * AUDIO_MAX_GAIN / 100;
 438                 if ( vol->right == vol->left )
 439                     info.play.balance = AUDIO_MID_BALANCE;
 440                 else
 441                     info.play.balance = (vol->right - vol->left + volume) * AUDIO_RIGHT_BALANCE / (2*volume);
 442             }
 443 #if !defined (__OpenBSD__) && !defined (__NetBSD__)
 444             info.output_muted = (volume == 0);
 445 #endif
 446             ioctl( fd,AUDIO_SETINFO,&info );
 447             close( fd );
 448             return CONTROL_OK;
 449         }
 450         return CONTROL_ERROR;
 451     }
 452     }
 453     return CONTROL_UNKNOWN;
 454 }
 455
 456 // open & setup audio device
 457 // return: 1=success 0=fail
 458 static int init(int rate,int channels,int format,int flags){
 459
 460     audio_info_t info;
 461     int pass;
 462     int ok;
 463     int convert_u8_s8;
 464
 465     setup_device_paths();
 466
 467     if (enable_sample_timing == RTSC_UNKNOWN
 468         && !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) {
 469         enable_sample_timing = realtime_samplecounter_available(audio_dev);
 470     }
 471
 472     mp_msg(MSGT_AO,MSGL_STATUS,"ao2: %d Hz  %d chans  %s [0x%X]\n",
 473            rate,channels,af_fmt2str_short(format),format);
 474
 475     audio_fd=open(audio_dev, O_WRONLY);
 476     if(audio_fd<0){
 477         mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_CantOpenAudioDev, audio_dev, strerror(errno));
 478         return 0;
 479     }
 480
 481     ioctl(audio_fd, AUDIO_DRAIN, 0);
 482
 483     if (af2sunfmt(format) == AUDIO_ENCODING_NONE)
 484       format = AF_FORMAT_S16_NE;
 485
 486     for (ok = pass = 0; pass <= 5; pass++) { /* pass 6&7 not useful */
 487
 488         AUDIO_INITINFO(&info);
 489         info.play.encoding = af2sunfmt(ao_data.format = format);
 490         info.play.precision =
 491             (format==AF_FORMAT_S16_NE
 492              ? AUDIO_PRECISION_16
 493              : AUDIO_PRECISION_8);
 494         info.play.channels = ao_data.channels = channels;
 495         info.play.sample_rate = ao_data.samplerate = rate;
 496
 497         convert_u8_s8 = 0;
 498
 499         if (pass & 1) {
 500             /*
 501              * on some sun audio drivers, 8-bit unsigned LINEAR8 encoding is
 502              * not supported, but 8-bit signed encoding is.
 503              *
 504              * Try S8, and if it works, use our own U8->S8 conversion before
 505              * sending the samples to the sound driver.
 506              */
 507 #ifdef AUDIO_ENCODING_LINEAR8
 508             if (info.play.encoding != AUDIO_ENCODING_LINEAR8)
 509 #endif
 510                 continue;
 511             info.play.encoding = AUDIO_ENCODING_LINEAR;
 512             convert_u8_s8 = 1;
 513         }
 514
 515         if (pass & 2) {
 516             /*
 517              * on some sun audio drivers, only certain fixed sample rates are
 518              * supported.
 519              *
 520              * In case the requested sample rate is very close to one of the
 521              * supported rates,  use the fixed supported rate instead.
 522              */
 523             if (!(info.play.sample_rate =
 524                   find_close_samplerate_match(audio_fd, rate)))
 525               continue;
 526
 527             /*
 528              * I'm not returning the correct sample rate in
 529              * |ao_data.samplerate|, to avoid software resampling.
 530              *
 531              * ao_data.samplerate = info.play.sample_rate;
 532              */
 533         }
 534
 535         if (pass & 4) {
 536             /* like "pass & 2", but use the highest supported sample rate */
 537             if (!(info.play.sample_rate
 538                   = ao_data.samplerate
 539                   = find_highest_samplerate(audio_fd)))
 540                 continue;
 541         }
 542
 543         ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
 544         if (ok) {
 545             /* audio format accepted by audio driver */
 546             break;
 547         }
 548
 549         /*
 550          * format not supported?
 551          * retry with different encoding and/or sample rate
 552          */
 553     }
 554
 555     if (!ok) {
 556         char buf[128];
 557         mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SUN_UnsupSampleRate,
 558                channels, af_fmt2str(format, buf, 128), rate);
 559         return 0;
 560     }
 561
 562     if (convert_u8_s8)
 563       ao_data.format = AF_FORMAT_S8;
 564
 565     bytes_per_sample = channels * info.play.precision / 8;
 566     ao_data.bps = byte_per_sec = bytes_per_sample * ao_data.samplerate;
 567     ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192;
 568
 569     AUDIO_INITINFO(&info);
 570     info.play.samples = 0;
 571     info.play.eof = 0;
 572     info.play.error = 0;
 573     ioctl (audio_fd, AUDIO_SETINFO, &info);
 574
 575     queued_bursts = 0;
 576     queued_samples = 0;
 577
 578     return 1;
 579 }
 580
 581 // close audio device
 582 static void uninit(int immed){
 583     // throw away buffered data in the audio driver's STREAMS queue
 584     if (immed)
 585         flush_audio(audio_fd);
 586     close(audio_fd);
 587 }
 588
 589 // stop playing and empty buffers (for seeking/pause)
 590 static void reset(void){
 591     audio_info_t info;
 592
 593     uninit(1);
 594     audio_fd=open(audio_dev, O_WRONLY);
 595     if(audio_fd<0){
 596         mp_msg(MSGT_AO, MSGL_FATAL, MSGTR_AO_SUN_CantReopenReset, strerror(errno));
 597         return;
 598     }
 599
 600     ioctl(audio_fd, AUDIO_DRAIN, 0);
 601
 602     AUDIO_INITINFO(&info);
 603     info.play.encoding = af2sunfmt(ao_data.format);
 604     info.play.precision =
 605         (ao_data.format==AF_FORMAT_S16_NE
 606          ? AUDIO_PRECISION_16
 607          : AUDIO_PRECISION_8);
 608     info.play.channels = ao_data.channels;
 609     info.play.sample_rate = ao_data.samplerate;
 610     info.play.samples = 0;
 611     info.play.eof = 0;
 612     info.play.error = 0;
 613     ioctl (audio_fd, AUDIO_SETINFO, &info);
 614     queued_bursts = 0;
 615     queued_samples = 0;
 616 }
 617
 618 // stop playing, keep buffers (for pause)
 619 static void audio_pause(void)
 620 {
 621     struct audio_info info;
 622     AUDIO_INITINFO(&info);
 623     info.play.pause = 1;
 624     ioctl(audio_fd, AUDIO_SETINFO, &info);
 625 }
 626
 627 // resume playing, after audio_pause()
 628 static void audio_resume(void)
 629 {
 630     struct audio_info info;
 631     AUDIO_INITINFO(&info);
 632     info.play.pause = 0;
 633     ioctl(audio_fd, AUDIO_SETINFO, &info);
 634 }
 635
 636
 637 // return: how many bytes can be played without blocking
 638 static int get_space(void){
 639     audio_info_t info;
 640
 641     // check buffer
 642 #ifdef HAVE_AUDIO_SELECT
 643     {
 644         fd_set rfds;
 645         struct timeval tv;
 646         FD_ZERO(&rfds);
 647         FD_SET(audio_fd, &rfds);
 648         tv.tv_sec = 0;
 649         tv.tv_usec = 0;
 650         if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block!
 651     }
 652 #endif
 653
 654     ioctl(audio_fd, AUDIO_GETINFO, &info);
 655 #if !defined (__OpenBSD__) && !defined(__NetBSD__)
 656     if (queued_bursts - info.play.eof > 2)
 657         return 0;
 658     return ao_data.outburst;
 659 #else
 660     return info.hiwat * info.blocksize - info.play.seek;
 661 #endif
 662
 663 }
 664
 665 // plays 'len' bytes of 'data'
 666 // it should round it down to outburst*n
 667 // return: number of bytes played
 668 static int play(void* data,int len,int flags){
 669     if (len < ao_data.outburst) return 0;
 670     len /= ao_data.outburst;
 671     len *= ao_data.outburst;
 672
 673     len = write(audio_fd, data, len);
 674     if(len > 0) {
 675         queued_samples += len / bytes_per_sample;
 676         if (write(audio_fd,data,0) < 0)
 677             perror("ao_sun: send EOF audio record");
 678         else
 679             queued_bursts ++;
 680     }
 681     return len;
 682 }
 683
 684
 685 // return: delay in seconds between first and last sample in buffer
 686 static float get_delay(void){
 687     audio_info_t info;
 688     ioctl(audio_fd, AUDIO_GETINFO, &info);
 689 #if defined (__OpenBSD__) || defined(__NetBSD__)
 690     return (float) info.play.seek/ (float)byte_per_sec ;
 691 #else
 692     if (info.play.samples && enable_sample_timing == RTSC_ENABLED)
 693         return (float)(queued_samples - info.play.samples) / (float)ao_data.samplerate;
 694     else
 695         return (float)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec;
 696 #endif
 697 }
 698