audio/alsaaudio.c

   1 /*
   2  * QEMU ALSA audio driver
   3  *
   4  * Copyright (c) 2005 Vassili Karpov (malc)
   5  *
   6  * Permission is hereby granted, free of charge, to any person obtaining a copy
   7  * of this software and associated documentation files (the "Software"), to deal
   8  * in the Software without restriction, including without limitation the rights
   9  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  10  * copies of the Software, and to permit persons to whom the Software is
  11  * furnished to do so, subject to the following conditions:
  12  *
  13  * The above copyright notice and this permission notice shall be included in
  14  * all copies or substantial portions of the Software.
  15  *
  16  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  17  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
  18  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
  19  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
  20  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  21  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  22  * THE SOFTWARE.
  23  */
  24
  25 #include "qemu/osdep.h"
  26 #include <alsa/asoundlib.h>
  27 #include "qemu/main-loop.h"
  28 #include "qemu/module.h"
  29 #include "audio.h"
  30 #include "trace.h"
  31
  32 #pragma GCC diagnostic ignored "-Waddress"
  33
  34 #define AUDIO_CAP "alsa"
  35 #include "audio_int.h"
  36
  37 struct pollhlp {
  38     snd_pcm_t *handle;
  39     struct pollfd *pfds;
  40     int count;
  41     int mask;
  42     AudioState *s;
  43 };
  44
  45 typedef struct ALSAVoiceOut {
  46     HWVoiceOut hw;
  47     snd_pcm_t *handle;
  48     struct pollhlp pollhlp;
  49     Audiodev *dev;
  50 } ALSAVoiceOut;
  51
  52 typedef struct ALSAVoiceIn {
  53     HWVoiceIn hw;
  54     snd_pcm_t *handle;
  55     struct pollhlp pollhlp;
  56     Audiodev *dev;
  57 } ALSAVoiceIn;
  58
  59 struct alsa_params_req {
  60     int freq;
  61     snd_pcm_format_t fmt;
  62     int nchannels;
  63 };
  64
  65 struct alsa_params_obt {
  66     int freq;
  67     AudioFormat fmt;
  68     int endianness;
  69     int nchannels;
  70     snd_pcm_uframes_t samples;
  71 };
  72
  73 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
  74 {
  75     va_list ap;
  76
  77     va_start (ap, fmt);
  78     AUD_vlog (AUDIO_CAP, fmt, ap);
  79     va_end (ap);
  80
  81     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
  82 }
  83
  84 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
  85     int err,
  86     const char *typ,
  87     const char *fmt,
  88     ...
  89     )
  90 {
  91     va_list ap;
  92
  93     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
  94
  95     va_start (ap, fmt);
  96     AUD_vlog (AUDIO_CAP, fmt, ap);
  97     va_end (ap);
  98
  99     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 100 }
 101
 102 static void alsa_fini_poll (struct pollhlp *hlp)
 103 {
 104     int i;
 105     struct pollfd *pfds = hlp->pfds;
 106
 107     if (pfds) {
 108         for (i = 0; i < hlp->count; ++i) {
 109             qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
 110         }
 111         g_free (pfds);
 112     }
 113     hlp->pfds = NULL;
 114     hlp->count = 0;
 115     hlp->handle = NULL;
 116 }
 117
 118 static void alsa_anal_close1 (snd_pcm_t **handlep)
 119 {
 120     int err = snd_pcm_close (*handlep);
 121     if (err) {
 122         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 123     }
 124     *handlep = NULL;
 125 }
 126
 127 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
 128 {
 129     alsa_fini_poll (hlp);
 130     alsa_anal_close1 (handlep);
 131 }
 132
 133 static int alsa_recover (snd_pcm_t *handle)
 134 {
 135     int err = snd_pcm_prepare (handle);
 136     if (err < 0) {
 137         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 138         return -1;
 139     }
 140     return 0;
 141 }
 142
 143 static int alsa_resume (snd_pcm_t *handle)
 144 {
 145     int err = snd_pcm_resume (handle);
 146     if (err < 0) {
 147         alsa_logerr (err, "Failed to resume handle %p\n", handle);
 148         return -1;
 149     }
 150     return 0;
 151 }
 152
 153 static void alsa_poll_handler (void *opaque)
 154 {
 155     int err, count;
 156     snd_pcm_state_t state;
 157     struct pollhlp *hlp = opaque;
 158     unsigned short revents;
 159
 160     count = poll (hlp->pfds, hlp->count, 0);
 161     if (count < 0) {
 162         dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
 163         return;
 164     }
 165
 166     if (!count) {
 167         return;
 168     }
 169
 170     /* XXX: ALSA example uses initial count, not the one returned by
 171        poll, correct? */
 172     err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
 173                                             hlp->count, &revents);
 174     if (err < 0) {
 175         alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
 176         return;
 177     }
 178
 179     if (!(revents & hlp->mask)) {
 180         trace_alsa_revents(revents);
 181         return;
 182     }
 183
 184     state = snd_pcm_state (hlp->handle);
 185     switch (state) {
 186     case SND_PCM_STATE_SETUP:
 187         alsa_recover (hlp->handle);
 188         break;
 189
 190     case SND_PCM_STATE_XRUN:
 191         alsa_recover (hlp->handle);
 192         break;
 193
 194     case SND_PCM_STATE_SUSPENDED:
 195         alsa_resume (hlp->handle);
 196         break;
 197
 198     case SND_PCM_STATE_PREPARED:
 199         audio_run(hlp->s, "alsa run (prepared)");
 200         break;
 201
 202     case SND_PCM_STATE_RUNNING:
 203         audio_run(hlp->s, "alsa run (running)");
 204         break;
 205
 206     default:
 207         dolog ("Unexpected state %d\n", state);
 208     }
 209 }
 210
 211 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
 212 {
 213     int i, count, err;
 214     struct pollfd *pfds;
 215
 216     count = snd_pcm_poll_descriptors_count (handle);
 217     if (count <= 0) {
 218         dolog ("Could not initialize poll mode\n"
 219                "Invalid number of poll descriptors %d\n", count);
 220         return -1;
 221     }
 222
 223     pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
 224     if (!pfds) {
 225         dolog ("Could not initialize poll mode\n");
 226         return -1;
 227     }
 228
 229     err = snd_pcm_poll_descriptors (handle, pfds, count);
 230     if (err < 0) {
 231         alsa_logerr (err, "Could not initialize poll mode\n"
 232                      "Could not obtain poll descriptors\n");
 233         g_free (pfds);
 234         return -1;
 235     }
 236
 237     for (i = 0; i < count; ++i) {
 238         if (pfds[i].events & POLLIN) {
 239             qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
 240         }
 241         if (pfds[i].events & POLLOUT) {
 242             trace_alsa_pollout(i, pfds[i].fd);
 243             qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
 244         }
 245         trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
 246
 247     }
 248     hlp->pfds = pfds;
 249     hlp->count = count;
 250     hlp->handle = handle;
 251     hlp->mask = mask;
 252     return 0;
 253 }
 254
 255 static int alsa_poll_out (HWVoiceOut *hw)
 256 {
 257     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 258
 259     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
 260 }
 261
 262 static int alsa_poll_in (HWVoiceIn *hw)
 263 {
 264     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 265
 266     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
 267 }
 268
 269 static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
 270 {
 271     switch (fmt) {
 272     case AUDIO_FORMAT_S8:
 273         return SND_PCM_FORMAT_S8;
 274
 275     case AUDIO_FORMAT_U8:
 276         return SND_PCM_FORMAT_U8;
 277
 278     case AUDIO_FORMAT_S16:
 279         if (endianness) {
 280             return SND_PCM_FORMAT_S16_BE;
 281         }
 282         else {
 283             return SND_PCM_FORMAT_S16_LE;
 284         }
 285
 286     case AUDIO_FORMAT_U16:
 287         if (endianness) {
 288             return SND_PCM_FORMAT_U16_BE;
 289         }
 290         else {
 291             return SND_PCM_FORMAT_U16_LE;
 292         }
 293
 294     case AUDIO_FORMAT_S32:
 295         if (endianness) {
 296             return SND_PCM_FORMAT_S32_BE;
 297         }
 298         else {
 299             return SND_PCM_FORMAT_S32_LE;
 300         }
 301
 302     case AUDIO_FORMAT_U32:
 303         if (endianness) {
 304             return SND_PCM_FORMAT_U32_BE;
 305         }
 306         else {
 307             return SND_PCM_FORMAT_U32_LE;
 308         }
 309
 310     case AUDIO_FORMAT_F32:
 311         if (endianness) {
 312             return SND_PCM_FORMAT_FLOAT_BE;
 313         } else {
 314             return SND_PCM_FORMAT_FLOAT_LE;
 315         }
 316
 317     default:
 318         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 319 #ifdef DEBUG_AUDIO
 320         abort ();
 321 #endif
 322         return SND_PCM_FORMAT_U8;
 323     }
 324 }
 325
 326 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
 327                            int *endianness)
 328 {
 329     switch (alsafmt) {
 330     case SND_PCM_FORMAT_S8:
 331         *endianness = 0;
 332         *fmt = AUDIO_FORMAT_S8;
 333         break;
 334
 335     case SND_PCM_FORMAT_U8:
 336         *endianness = 0;
 337         *fmt = AUDIO_FORMAT_U8;
 338         break;
 339
 340     case SND_PCM_FORMAT_S16_LE:
 341         *endianness = 0;
 342         *fmt = AUDIO_FORMAT_S16;
 343         break;
 344
 345     case SND_PCM_FORMAT_U16_LE:
 346         *endianness = 0;
 347         *fmt = AUDIO_FORMAT_U16;
 348         break;
 349
 350     case SND_PCM_FORMAT_S16_BE:
 351         *endianness = 1;
 352         *fmt = AUDIO_FORMAT_S16;
 353         break;
 354
 355     case SND_PCM_FORMAT_U16_BE:
 356         *endianness = 1;
 357         *fmt = AUDIO_FORMAT_U16;
 358         break;
 359
 360     case SND_PCM_FORMAT_S32_LE:
 361         *endianness = 0;
 362         *fmt = AUDIO_FORMAT_S32;
 363         break;
 364
 365     case SND_PCM_FORMAT_U32_LE:
 366         *endianness = 0;
 367         *fmt = AUDIO_FORMAT_U32;
 368         break;
 369
 370     case SND_PCM_FORMAT_S32_BE:
 371         *endianness = 1;
 372         *fmt = AUDIO_FORMAT_S32;
 373         break;
 374
 375     case SND_PCM_FORMAT_U32_BE:
 376         *endianness = 1;
 377         *fmt = AUDIO_FORMAT_U32;
 378         break;
 379
 380     case SND_PCM_FORMAT_FLOAT_LE:
 381         *endianness = 0;
 382         *fmt = AUDIO_FORMAT_F32;
 383         break;
 384
 385     case SND_PCM_FORMAT_FLOAT_BE:
 386         *endianness = 1;
 387         *fmt = AUDIO_FORMAT_F32;
 388         break;
 389
 390     default:
 391         dolog ("Unrecognized audio format %d\n", alsafmt);
 392         return -1;
 393     }
 394
 395     return 0;
 396 }
 397
 398 static void alsa_dump_info (struct alsa_params_req *req,
 399                             struct alsa_params_obt *obt,
 400                             snd_pcm_format_t obtfmt,
 401                             AudiodevAlsaPerDirectionOptions *apdo)
 402 {
 403     dolog("parameter | requested value | obtained value\n");
 404     dolog("format    |      %10d |     %10d\n", req->fmt, obtfmt);
 405     dolog("channels  |      %10d |     %10d\n",
 406           req->nchannels, obt->nchannels);
 407     dolog("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 408     dolog("============================================\n");
 409     dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
 410           apdo->buffer_length, apdo->period_length);
 411     dolog("obtained: samples %ld\n", obt->samples);
 412 }
 413
 414 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 415 {
 416     int err;
 417     snd_pcm_sw_params_t *sw_params;
 418
 419     snd_pcm_sw_params_alloca (&sw_params);
 420
 421     err = snd_pcm_sw_params_current (handle, sw_params);
 422     if (err < 0) {
 423         dolog ("Could not fully initialize DAC\n");
 424         alsa_logerr (err, "Failed to get current software parameters\n");
 425         return;
 426     }
 427
 428     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 429     if (err < 0) {
 430         dolog ("Could not fully initialize DAC\n");
 431         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 432                      threshold);
 433         return;
 434     }
 435
 436     err = snd_pcm_sw_params (handle, sw_params);
 437     if (err < 0) {
 438         dolog ("Could not fully initialize DAC\n");
 439         alsa_logerr (err, "Failed to set software parameters\n");
 440         return;
 441     }
 442 }
 443
 444 static int alsa_open(bool in, struct alsa_params_req *req,
 445                      struct alsa_params_obt *obt, snd_pcm_t **handlep,
 446                      Audiodev *dev)
 447 {
 448     AudiodevAlsaOptions *aopts = &dev->u.alsa;
 449     AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
 450     snd_pcm_t *handle;
 451     snd_pcm_hw_params_t *hw_params;
 452     int err;
 453     unsigned int freq, nchannels;
 454     const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
 455     snd_pcm_uframes_t obt_buffer_size;
 456     const char *typ = in ? "ADC" : "DAC";
 457     snd_pcm_format_t obtfmt;
 458
 459     freq = req->freq;
 460     nchannels = req->nchannels;
 461
 462     snd_pcm_hw_params_alloca (&hw_params);
 463
 464     err = snd_pcm_open (
 465         &handle,
 466         pcm_name,
 467         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 468         SND_PCM_NONBLOCK
 469         );
 470     if (err < 0) {
 471         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 472         return -1;
 473     }
 474
 475     err = snd_pcm_hw_params_any (handle, hw_params);
 476     if (err < 0) {
 477         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 478         goto err;
 479     }
 480
 481     err = snd_pcm_hw_params_set_access (
 482         handle,
 483         hw_params,
 484         SND_PCM_ACCESS_RW_INTERLEAVED
 485         );
 486     if (err < 0) {
 487         alsa_logerr2 (err, typ, "Failed to set access type\n");
 488         goto err;
 489     }
 490
 491     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 492     if (err < 0) {
 493         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 494     }
 495
 496     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 497     if (err < 0) {
 498         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 499         goto err;
 500     }
 501
 502     err = snd_pcm_hw_params_set_channels_near (
 503         handle,
 504         hw_params,
 505         &nchannels
 506         );
 507     if (err < 0) {
 508         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 509                       req->nchannels);
 510         goto err;
 511     }
 512
 513     if (apdo->buffer_length) {
 514         int dir = 0;
 515         unsigned int btime = apdo->buffer_length;
 516
 517         err = snd_pcm_hw_params_set_buffer_time_near(
 518             handle, hw_params, &btime, &dir);
 519
 520         if (err < 0) {
 521             alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
 522                          apdo->buffer_length);
 523             goto err;
 524         }
 525
 526         if (apdo->has_buffer_length && btime != apdo->buffer_length) {
 527             dolog("Requested buffer time %" PRId32
 528                   " was rejected, using %u\n", apdo->buffer_length, btime);
 529         }
 530     }
 531
 532     if (apdo->period_length) {
 533         int dir = 0;
 534         unsigned int ptime = apdo->period_length;
 535
 536         err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
 537                                                      &dir);
 538
 539         if (err < 0) {
 540             alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
 541                          apdo->period_length);
 542             goto err;
 543         }
 544
 545         if (apdo->has_period_length && ptime != apdo->period_length) {
 546             dolog("Requested period time %" PRId32 " was rejected, using %d\n",
 547                   apdo->period_length, ptime);
 548         }
 549     }
 550
 551     err = snd_pcm_hw_params (handle, hw_params);
 552     if (err < 0) {
 553         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 554         goto err;
 555     }
 556
 557     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 558     if (err < 0) {
 559         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 560         goto err;
 561     }
 562
 563     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 564     if (err < 0) {
 565         alsa_logerr2 (err, typ, "Failed to get format\n");
 566         goto err;
 567     }
 568
 569     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 570         dolog ("Invalid format was returned %d\n", obtfmt);
 571         goto err;
 572     }
 573
 574     err = snd_pcm_prepare (handle);
 575     if (err < 0) {
 576         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 577         goto err;
 578     }
 579
 580     if (!in && aopts->has_threshold && aopts->threshold) {
 581         struct audsettings as = { .freq = freq };
 582         alsa_set_threshold(
 583             handle,
 584             audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
 585                                 &as, aopts->threshold));
 586     }
 587
 588     obt->nchannels = nchannels;
 589     obt->freq = freq;
 590     obt->samples = obt_buffer_size;
 591
 592     *handlep = handle;
 593
 594     if (obtfmt != req->fmt ||
 595          obt->nchannels != req->nchannels ||
 596          obt->freq != req->freq) {
 597         dolog ("Audio parameters for %s\n", typ);
 598         alsa_dump_info(req, obt, obtfmt, apdo);
 599     }
 600
 601 #ifdef DEBUG
 602     alsa_dump_info(req, obt, obtfmt, pdo);
 603 #endif
 604     return 0;
 605
 606  err:
 607     alsa_anal_close1 (&handle);
 608     return -1;
 609 }
 610
 611 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
 612 {
 613     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 614     size_t pos = 0;
 615     size_t len_frames = len / hw->info.bytes_per_frame;
 616
 617     while (len_frames) {
 618         char *src = advance(buf, pos);
 619         snd_pcm_sframes_t written;
 620
 621         written = snd_pcm_writei(alsa->handle, src, len_frames);
 622
 623         if (written <= 0) {
 624             switch (written) {
 625             case 0:
 626                 trace_alsa_wrote_zero(len_frames);
 627                 return pos;
 628
 629             case -EPIPE:
 630                 if (alsa_recover(alsa->handle)) {
 631                     alsa_logerr(written, "Failed to write %zu frames\n",
 632                                 len_frames);
 633                     return pos;
 634                 }
 635                 trace_alsa_xrun_out();
 636                 continue;
 637
 638             case -ESTRPIPE:
 639                 /*
 640                  * stream is suspended and waiting for an application
 641                  * recovery
 642                  */
 643                 if (alsa_resume(alsa->handle)) {
 644                     alsa_logerr(written, "Failed to write %zu frames\n",
 645                                 len_frames);
 646                     return pos;
 647                 }
 648                 trace_alsa_resume_out();
 649                 continue;
 650
 651             case -EAGAIN:
 652                 return pos;
 653
 654             default:
 655                 alsa_logerr(written, "Failed to write %zu frames from %p\n",
 656                             len, src);
 657                 return pos;
 658             }
 659         }
 660
 661         pos += written * hw->info.bytes_per_frame;
 662         if (written < len_frames) {
 663             break;
 664         }
 665         len_frames -= written;
 666     }
 667
 668     return pos;
 669 }
 670
 671 static void alsa_fini_out (HWVoiceOut *hw)
 672 {
 673     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 674
 675     ldebug ("alsa_fini\n");
 676     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 677 }
 678
 679 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
 680                          void *drv_opaque)
 681 {
 682     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 683     struct alsa_params_req req;
 684     struct alsa_params_obt obt;
 685     snd_pcm_t *handle;
 686     struct audsettings obt_as;
 687     Audiodev *dev = drv_opaque;
 688
 689     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 690     req.freq = as->freq;
 691     req.nchannels = as->nchannels;
 692
 693     if (alsa_open(0, &req, &obt, &handle, dev)) {
 694         return -1;
 695     }
 696
 697     obt_as.freq = obt.freq;
 698     obt_as.nchannels = obt.nchannels;
 699     obt_as.fmt = obt.fmt;
 700     obt_as.endianness = obt.endianness;
 701
 702     audio_pcm_init_info (&hw->info, &obt_as);
 703     hw->samples = obt.samples;
 704
 705     alsa->pollhlp.s = hw->s;
 706     alsa->handle = handle;
 707     alsa->dev = dev;
 708     return 0;
 709 }
 710
 711 #define VOICE_CTL_PAUSE 0
 712 #define VOICE_CTL_PREPARE 1
 713 #define VOICE_CTL_START 2
 714
 715 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
 716 {
 717     int err;
 718
 719     if (ctl == VOICE_CTL_PAUSE) {
 720         err = snd_pcm_drop (handle);
 721         if (err < 0) {
 722             alsa_logerr (err, "Could not stop %s\n", typ);
 723             return -1;
 724         }
 725     }
 726     else {
 727         err = snd_pcm_prepare (handle);
 728         if (err < 0) {
 729             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 730             return -1;
 731         }
 732         if (ctl == VOICE_CTL_START) {
 733             err = snd_pcm_start(handle);
 734             if (err < 0) {
 735                 alsa_logerr (err, "Could not start handle for %s\n", typ);
 736                 return -1;
 737             }
 738         }
 739     }
 740
 741     return 0;
 742 }
 743
 744 static void alsa_enable_out(HWVoiceOut *hw, bool enable)
 745 {
 746     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 747     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
 748
 749     if (enable) {
 750         bool poll_mode = apdo->try_poll;
 751
 752         ldebug("enabling voice\n");
 753         if (poll_mode && alsa_poll_out(hw)) {
 754             poll_mode = 0;
 755         }
 756         hw->poll_mode = poll_mode;
 757         alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
 758     } else {
 759         ldebug("disabling voice\n");
 760         if (hw->poll_mode) {
 761             hw->poll_mode = 0;
 762             alsa_fini_poll(&alsa->pollhlp);
 763         }
 764         alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
 765     }
 766 }
 767
 768 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 769 {
 770     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 771     struct alsa_params_req req;
 772     struct alsa_params_obt obt;
 773     snd_pcm_t *handle;
 774     struct audsettings obt_as;
 775     Audiodev *dev = drv_opaque;
 776
 777     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 778     req.freq = as->freq;
 779     req.nchannels = as->nchannels;
 780
 781     if (alsa_open(1, &req, &obt, &handle, dev)) {
 782         return -1;
 783     }
 784
 785     obt_as.freq = obt.freq;
 786     obt_as.nchannels = obt.nchannels;
 787     obt_as.fmt = obt.fmt;
 788     obt_as.endianness = obt.endianness;
 789
 790     audio_pcm_init_info (&hw->info, &obt_as);
 791     hw->samples = obt.samples;
 792
 793     alsa->pollhlp.s = hw->s;
 794     alsa->handle = handle;
 795     alsa->dev = dev;
 796     return 0;
 797 }
 798
 799 static void alsa_fini_in (HWVoiceIn *hw)
 800 {
 801     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 802
 803     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 804 }
 805
 806 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
 807 {
 808     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 809     size_t pos = 0;
 810
 811     while (len) {
 812         void *dst = advance(buf, pos);
 813         snd_pcm_sframes_t nread;
 814
 815         nread = snd_pcm_readi(
 816             alsa->handle, dst, len / hw->info.bytes_per_frame);
 817
 818         if (nread <= 0) {
 819             switch (nread) {
 820             case 0:
 821                 trace_alsa_read_zero(len);
 822                 return pos;
 823
 824             case -EPIPE:
 825                 if (alsa_recover(alsa->handle)) {
 826                     alsa_logerr(nread, "Failed to read %zu frames\n", len);
 827                     return pos;
 828                 }
 829                 trace_alsa_xrun_in();
 830                 continue;
 831
 832             case -EAGAIN:
 833                 return pos;
 834
 835             default:
 836                 alsa_logerr(nread, "Failed to read %zu frames to %p\n",
 837                             len, dst);
 838                 return pos;
 839             }
 840         }
 841
 842         pos += nread * hw->info.bytes_per_frame;
 843         len -= nread * hw->info.bytes_per_frame;
 844     }
 845
 846     return pos;
 847 }
 848
 849 static void alsa_enable_in(HWVoiceIn *hw, bool enable)
 850 {
 851     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 852     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
 853
 854     if (enable) {
 855         bool poll_mode = apdo->try_poll;
 856
 857         ldebug("enabling voice\n");
 858         if (poll_mode && alsa_poll_in(hw)) {
 859             poll_mode = 0;
 860         }
 861         hw->poll_mode = poll_mode;
 862
 863         alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
 864     } else {
 865         ldebug ("disabling voice\n");
 866         if (hw->poll_mode) {
 867             hw->poll_mode = 0;
 868             alsa_fini_poll(&alsa->pollhlp);
 869         }
 870         alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
 871     }
 872 }
 873
 874 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
 875 {
 876     if (!apdo->has_try_poll) {
 877         apdo->try_poll = true;
 878         apdo->has_try_poll = true;
 879     }
 880 }
 881
 882 static void *alsa_audio_init(Audiodev *dev)
 883 {
 884     AudiodevAlsaOptions *aopts;
 885     assert(dev->driver == AUDIODEV_DRIVER_ALSA);
 886
 887     aopts = &dev->u.alsa;
 888     alsa_init_per_direction(aopts->in);
 889     alsa_init_per_direction(aopts->out);
 890
 891     /*
 892      * need to define them, as otherwise alsa produces no sound
 893      * doesn't set has_* so alsa_open can identify it wasn't set by the user
 894      */
 895     if (!dev->u.alsa.out->has_period_length) {
 896         /* 1024 frames assuming 44100Hz */
 897         dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
 898     }
 899     if (!dev->u.alsa.out->has_buffer_length) {
 900         /* 4096 frames assuming 44100Hz */
 901         dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
 902     }
 903
 904     /*
 905      * OptsVisitor sets unspecified optional fields to zero, but do not depend
 906      * on it...
 907      */
 908     if (!dev->u.alsa.in->has_period_length) {
 909         dev->u.alsa.in->period_length = 0;
 910     }
 911     if (!dev->u.alsa.in->has_buffer_length) {
 912         dev->u.alsa.in->buffer_length = 0;
 913     }
 914
 915     return dev;
 916 }
 917
 918 static void alsa_audio_fini (void *opaque)
 919 {
 920 }
 921
 922 static struct audio_pcm_ops alsa_pcm_ops = {
 923     .init_out = alsa_init_out,
 924     .fini_out = alsa_fini_out,
 925     .write    = alsa_write,
 926     .run_buffer_out = audio_generic_run_buffer_out,
 927     .enable_out = alsa_enable_out,
 928
 929     .init_in  = alsa_init_in,
 930     .fini_in  = alsa_fini_in,
 931     .read     = alsa_read,
 932     .enable_in = alsa_enable_in,
 933 };
 934
 935 static struct audio_driver alsa_audio_driver = {
 936     .name           = "alsa",
 937     .descr          = "ALSA http://www.alsa-project.org",
 938     .init           = alsa_audio_init,
 939     .fini           = alsa_audio_fini,
 940     .pcm_ops        = &alsa_pcm_ops,
 941     .can_be_default = 1,
 942     .max_voices_out = INT_MAX,
 943     .max_voices_in  = INT_MAX,
 944     .voice_size_out = sizeof (ALSAVoiceOut),
 945     .voice_size_in  = sizeof (ALSAVoiceIn)
 946 };
 947
 948 static void register_audio_alsa(void)
 949 {
 950     audio_driver_register(&alsa_audio_driver);
 951 }
 952 type_init(register_audio_alsa);