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     default:
 311         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 312 #ifdef DEBUG_AUDIO
 313         abort ();
 314 #endif
 315         return SND_PCM_FORMAT_U8;
 316     }
 317 }
 318
 319 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
 320                            int *endianness)
 321 {
 322     switch (alsafmt) {
 323     case SND_PCM_FORMAT_S8:
 324         *endianness = 0;
 325         *fmt = AUDIO_FORMAT_S8;
 326         break;
 327
 328     case SND_PCM_FORMAT_U8:
 329         *endianness = 0;
 330         *fmt = AUDIO_FORMAT_U8;
 331         break;
 332
 333     case SND_PCM_FORMAT_S16_LE:
 334         *endianness = 0;
 335         *fmt = AUDIO_FORMAT_S16;
 336         break;
 337
 338     case SND_PCM_FORMAT_U16_LE:
 339         *endianness = 0;
 340         *fmt = AUDIO_FORMAT_U16;
 341         break;
 342
 343     case SND_PCM_FORMAT_S16_BE:
 344         *endianness = 1;
 345         *fmt = AUDIO_FORMAT_S16;
 346         break;
 347
 348     case SND_PCM_FORMAT_U16_BE:
 349         *endianness = 1;
 350         *fmt = AUDIO_FORMAT_U16;
 351         break;
 352
 353     case SND_PCM_FORMAT_S32_LE:
 354         *endianness = 0;
 355         *fmt = AUDIO_FORMAT_S32;
 356         break;
 357
 358     case SND_PCM_FORMAT_U32_LE:
 359         *endianness = 0;
 360         *fmt = AUDIO_FORMAT_U32;
 361         break;
 362
 363     case SND_PCM_FORMAT_S32_BE:
 364         *endianness = 1;
 365         *fmt = AUDIO_FORMAT_S32;
 366         break;
 367
 368     case SND_PCM_FORMAT_U32_BE:
 369         *endianness = 1;
 370         *fmt = AUDIO_FORMAT_U32;
 371         break;
 372
 373     default:
 374         dolog ("Unrecognized audio format %d\n", alsafmt);
 375         return -1;
 376     }
 377
 378     return 0;
 379 }
 380
 381 static void alsa_dump_info (struct alsa_params_req *req,
 382                             struct alsa_params_obt *obt,
 383                             snd_pcm_format_t obtfmt,
 384                             AudiodevAlsaPerDirectionOptions *apdo)
 385 {
 386     dolog("parameter | requested value | obtained value\n");
 387     dolog("format    |      %10d |     %10d\n", req->fmt, obtfmt);
 388     dolog("channels  |      %10d |     %10d\n",
 389           req->nchannels, obt->nchannels);
 390     dolog("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 391     dolog("============================================\n");
 392     dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
 393           apdo->buffer_length, apdo->period_length);
 394     dolog("obtained: samples %ld\n", obt->samples);
 395 }
 396
 397 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 398 {
 399     int err;
 400     snd_pcm_sw_params_t *sw_params;
 401
 402     snd_pcm_sw_params_alloca (&sw_params);
 403
 404     err = snd_pcm_sw_params_current (handle, sw_params);
 405     if (err < 0) {
 406         dolog ("Could not fully initialize DAC\n");
 407         alsa_logerr (err, "Failed to get current software parameters\n");
 408         return;
 409     }
 410
 411     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 412     if (err < 0) {
 413         dolog ("Could not fully initialize DAC\n");
 414         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 415                      threshold);
 416         return;
 417     }
 418
 419     err = snd_pcm_sw_params (handle, sw_params);
 420     if (err < 0) {
 421         dolog ("Could not fully initialize DAC\n");
 422         alsa_logerr (err, "Failed to set software parameters\n");
 423         return;
 424     }
 425 }
 426
 427 static int alsa_open(bool in, struct alsa_params_req *req,
 428                      struct alsa_params_obt *obt, snd_pcm_t **handlep,
 429                      Audiodev *dev)
 430 {
 431     AudiodevAlsaOptions *aopts = &dev->u.alsa;
 432     AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
 433     snd_pcm_t *handle;
 434     snd_pcm_hw_params_t *hw_params;
 435     int err;
 436     unsigned int freq, nchannels;
 437     const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
 438     snd_pcm_uframes_t obt_buffer_size;
 439     const char *typ = in ? "ADC" : "DAC";
 440     snd_pcm_format_t obtfmt;
 441
 442     freq = req->freq;
 443     nchannels = req->nchannels;
 444
 445     snd_pcm_hw_params_alloca (&hw_params);
 446
 447     err = snd_pcm_open (
 448         &handle,
 449         pcm_name,
 450         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 451         SND_PCM_NONBLOCK
 452         );
 453     if (err < 0) {
 454         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 455         return -1;
 456     }
 457
 458     err = snd_pcm_hw_params_any (handle, hw_params);
 459     if (err < 0) {
 460         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 461         goto err;
 462     }
 463
 464     err = snd_pcm_hw_params_set_access (
 465         handle,
 466         hw_params,
 467         SND_PCM_ACCESS_RW_INTERLEAVED
 468         );
 469     if (err < 0) {
 470         alsa_logerr2 (err, typ, "Failed to set access type\n");
 471         goto err;
 472     }
 473
 474     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 475     if (err < 0) {
 476         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 477     }
 478
 479     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 480     if (err < 0) {
 481         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 482         goto err;
 483     }
 484
 485     err = snd_pcm_hw_params_set_channels_near (
 486         handle,
 487         hw_params,
 488         &nchannels
 489         );
 490     if (err < 0) {
 491         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 492                       req->nchannels);
 493         goto err;
 494     }
 495
 496     if (nchannels != 1 && nchannels != 2) {
 497         alsa_logerr2 (err, typ,
 498                       "Can not handle obtained number of channels %d\n",
 499                       nchannels);
 500         goto err;
 501     }
 502
 503     if (apdo->buffer_length) {
 504         int dir = 0;
 505         unsigned int btime = apdo->buffer_length;
 506
 507         err = snd_pcm_hw_params_set_buffer_time_near(
 508             handle, hw_params, &btime, &dir);
 509
 510         if (err < 0) {
 511             alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
 512                          apdo->buffer_length);
 513             goto err;
 514         }
 515
 516         if (apdo->has_buffer_length && btime != apdo->buffer_length) {
 517             dolog("Requested buffer time %" PRId32
 518                   " was rejected, using %u\n", apdo->buffer_length, btime);
 519         }
 520     }
 521
 522     if (apdo->period_length) {
 523         int dir = 0;
 524         unsigned int ptime = apdo->period_length;
 525
 526         err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
 527                                                      &dir);
 528
 529         if (err < 0) {
 530             alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
 531                          apdo->period_length);
 532             goto err;
 533         }
 534
 535         if (apdo->has_period_length && ptime != apdo->period_length) {
 536             dolog("Requested period time %" PRId32 " was rejected, using %d\n",
 537                   apdo->period_length, ptime);
 538         }
 539     }
 540
 541     err = snd_pcm_hw_params (handle, hw_params);
 542     if (err < 0) {
 543         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 544         goto err;
 545     }
 546
 547     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 548     if (err < 0) {
 549         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 550         goto err;
 551     }
 552
 553     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 554     if (err < 0) {
 555         alsa_logerr2 (err, typ, "Failed to get format\n");
 556         goto err;
 557     }
 558
 559     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 560         dolog ("Invalid format was returned %d\n", obtfmt);
 561         goto err;
 562     }
 563
 564     err = snd_pcm_prepare (handle);
 565     if (err < 0) {
 566         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 567         goto err;
 568     }
 569
 570     if (!in && aopts->has_threshold && aopts->threshold) {
 571         struct audsettings as = { .freq = freq };
 572         alsa_set_threshold(
 573             handle,
 574             audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
 575                                 &as, aopts->threshold));
 576     }
 577
 578     obt->nchannels = nchannels;
 579     obt->freq = freq;
 580     obt->samples = obt_buffer_size;
 581
 582     *handlep = handle;
 583
 584     if (obtfmt != req->fmt ||
 585          obt->nchannels != req->nchannels ||
 586          obt->freq != req->freq) {
 587         dolog ("Audio parameters for %s\n", typ);
 588         alsa_dump_info(req, obt, obtfmt, apdo);
 589     }
 590
 591 #ifdef DEBUG
 592     alsa_dump_info(req, obt, obtfmt, pdo);
 593 #endif
 594     return 0;
 595
 596  err:
 597     alsa_anal_close1 (&handle);
 598     return -1;
 599 }
 600
 601 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
 602 {
 603     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 604     size_t pos = 0;
 605     size_t len_frames = len >> hw->info.shift;
 606
 607     while (len_frames) {
 608         char *src = advance(buf, pos);
 609         snd_pcm_sframes_t written;
 610
 611         written = snd_pcm_writei(alsa->handle, src, len_frames);
 612
 613         if (written <= 0) {
 614             switch (written) {
 615             case 0:
 616                 trace_alsa_wrote_zero(len_frames);
 617                 return pos;
 618
 619             case -EPIPE:
 620                 if (alsa_recover(alsa->handle)) {
 621                     alsa_logerr(written, "Failed to write %zu frames\n",
 622                                 len_frames);
 623                     return pos;
 624                 }
 625                 trace_alsa_xrun_out();
 626                 continue;
 627
 628             case -ESTRPIPE:
 629                 /*
 630                  * stream is suspended and waiting for an application
 631                  * recovery
 632                  */
 633                 if (alsa_resume(alsa->handle)) {
 634                     alsa_logerr(written, "Failed to write %zu frames\n",
 635                                 len_frames);
 636                     return pos;
 637                 }
 638                 trace_alsa_resume_out();
 639                 continue;
 640
 641             case -EAGAIN:
 642                 return pos;
 643
 644             default:
 645                 alsa_logerr(written, "Failed to write %zu frames from %p\n",
 646                             len, src);
 647                 return pos;
 648             }
 649         }
 650
 651         pos += written << hw->info.shift;
 652         if (written < len_frames) {
 653             break;
 654         }
 655         len_frames -= written;
 656     }
 657
 658     return pos;
 659 }
 660
 661 static void alsa_fini_out (HWVoiceOut *hw)
 662 {
 663     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 664
 665     ldebug ("alsa_fini\n");
 666     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 667 }
 668
 669 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
 670                          void *drv_opaque)
 671 {
 672     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 673     struct alsa_params_req req;
 674     struct alsa_params_obt obt;
 675     snd_pcm_t *handle;
 676     struct audsettings obt_as;
 677     Audiodev *dev = drv_opaque;
 678
 679     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 680     req.freq = as->freq;
 681     req.nchannels = as->nchannels;
 682
 683     if (alsa_open(0, &req, &obt, &handle, dev)) {
 684         return -1;
 685     }
 686
 687     obt_as.freq = obt.freq;
 688     obt_as.nchannels = obt.nchannels;
 689     obt_as.fmt = obt.fmt;
 690     obt_as.endianness = obt.endianness;
 691
 692     audio_pcm_init_info (&hw->info, &obt_as);
 693     hw->samples = obt.samples;
 694
 695     alsa->pollhlp.s = hw->s;
 696     alsa->handle = handle;
 697     alsa->dev = dev;
 698     return 0;
 699 }
 700
 701 #define VOICE_CTL_PAUSE 0
 702 #define VOICE_CTL_PREPARE 1
 703 #define VOICE_CTL_START 2
 704
 705 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
 706 {
 707     int err;
 708
 709     if (ctl == VOICE_CTL_PAUSE) {
 710         err = snd_pcm_drop (handle);
 711         if (err < 0) {
 712             alsa_logerr (err, "Could not stop %s\n", typ);
 713             return -1;
 714         }
 715     }
 716     else {
 717         err = snd_pcm_prepare (handle);
 718         if (err < 0) {
 719             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 720             return -1;
 721         }
 722         if (ctl == VOICE_CTL_START) {
 723             err = snd_pcm_start(handle);
 724             if (err < 0) {
 725                 alsa_logerr (err, "Could not start handle for %s\n", typ);
 726                 return -1;
 727             }
 728         }
 729     }
 730
 731     return 0;
 732 }
 733
 734 static void alsa_enable_out(HWVoiceOut *hw, bool enable)
 735 {
 736     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 737     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
 738
 739     if (enable) {
 740         bool poll_mode = apdo->try_poll;
 741
 742         ldebug("enabling voice\n");
 743         if (poll_mode && alsa_poll_out(hw)) {
 744             poll_mode = 0;
 745         }
 746         hw->poll_mode = poll_mode;
 747         alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
 748     } else {
 749         ldebug("disabling voice\n");
 750         if (hw->poll_mode) {
 751             hw->poll_mode = 0;
 752             alsa_fini_poll(&alsa->pollhlp);
 753         }
 754         alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
 755     }
 756 }
 757
 758 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 759 {
 760     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 761     struct alsa_params_req req;
 762     struct alsa_params_obt obt;
 763     snd_pcm_t *handle;
 764     struct audsettings obt_as;
 765     Audiodev *dev = drv_opaque;
 766
 767     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 768     req.freq = as->freq;
 769     req.nchannels = as->nchannels;
 770
 771     if (alsa_open(1, &req, &obt, &handle, dev)) {
 772         return -1;
 773     }
 774
 775     obt_as.freq = obt.freq;
 776     obt_as.nchannels = obt.nchannels;
 777     obt_as.fmt = obt.fmt;
 778     obt_as.endianness = obt.endianness;
 779
 780     audio_pcm_init_info (&hw->info, &obt_as);
 781     hw->samples = obt.samples;
 782
 783     alsa->pollhlp.s = hw->s;
 784     alsa->handle = handle;
 785     alsa->dev = dev;
 786     return 0;
 787 }
 788
 789 static void alsa_fini_in (HWVoiceIn *hw)
 790 {
 791     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 792
 793     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 794 }
 795
 796 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
 797 {
 798     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 799     size_t pos = 0;
 800
 801     while (len) {
 802         void *dst = advance(buf, pos);
 803         snd_pcm_sframes_t nread;
 804
 805         nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);
 806
 807         if (nread <= 0) {
 808             switch (nread) {
 809             case 0:
 810                 trace_alsa_read_zero(len);
 811                 return pos;;
 812
 813             case -EPIPE:
 814                 if (alsa_recover(alsa->handle)) {
 815                     alsa_logerr(nread, "Failed to read %zu frames\n", len);
 816                     return pos;
 817                 }
 818                 trace_alsa_xrun_in();
 819                 continue;
 820
 821             case -EAGAIN:
 822                 return pos;
 823
 824             default:
 825                 alsa_logerr(nread, "Failed to read %zu frames to %p\n",
 826                             len, dst);
 827                 return pos;;
 828             }
 829         }
 830
 831         pos += nread << hw->info.shift;
 832         len -= nread << hw->info.shift;
 833     }
 834
 835     return pos;
 836 }
 837
 838 static void alsa_enable_in(HWVoiceIn *hw, bool enable)
 839 {
 840     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 841     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
 842
 843     if (enable) {
 844         bool poll_mode = apdo->try_poll;
 845
 846         ldebug("enabling voice\n");
 847         if (poll_mode && alsa_poll_in(hw)) {
 848             poll_mode = 0;
 849         }
 850         hw->poll_mode = poll_mode;
 851
 852         alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
 853     } else {
 854         ldebug ("disabling voice\n");
 855         if (hw->poll_mode) {
 856             hw->poll_mode = 0;
 857             alsa_fini_poll(&alsa->pollhlp);
 858         }
 859         alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
 860     }
 861 }
 862
 863 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
 864 {
 865     if (!apdo->has_try_poll) {
 866         apdo->try_poll = true;
 867         apdo->has_try_poll = true;
 868     }
 869 }
 870
 871 static void *alsa_audio_init(Audiodev *dev)
 872 {
 873     AudiodevAlsaOptions *aopts;
 874     assert(dev->driver == AUDIODEV_DRIVER_ALSA);
 875
 876     aopts = &dev->u.alsa;
 877     alsa_init_per_direction(aopts->in);
 878     alsa_init_per_direction(aopts->out);
 879
 880     /*
 881      * need to define them, as otherwise alsa produces no sound
 882      * doesn't set has_* so alsa_open can identify it wasn't set by the user
 883      */
 884     if (!dev->u.alsa.out->has_period_length) {
 885         /* 1024 frames assuming 44100Hz */
 886         dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
 887     }
 888     if (!dev->u.alsa.out->has_buffer_length) {
 889         /* 4096 frames assuming 44100Hz */
 890         dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
 891     }
 892
 893     /*
 894      * OptsVisitor sets unspecified optional fields to zero, but do not depend
 895      * on it...
 896      */
 897     if (!dev->u.alsa.in->has_period_length) {
 898         dev->u.alsa.in->period_length = 0;
 899     }
 900     if (!dev->u.alsa.in->has_buffer_length) {
 901         dev->u.alsa.in->buffer_length = 0;
 902     }
 903
 904     return dev;
 905 }
 906
 907 static void alsa_audio_fini (void *opaque)
 908 {
 909 }
 910
 911 static struct audio_pcm_ops alsa_pcm_ops = {
 912     .init_out = alsa_init_out,
 913     .fini_out = alsa_fini_out,
 914     .write    = alsa_write,
 915     .enable_out = alsa_enable_out,
 916
 917     .init_in  = alsa_init_in,
 918     .fini_in  = alsa_fini_in,
 919     .read     = alsa_read,
 920     .enable_in = alsa_enable_in,
 921 };
 922
 923 static struct audio_driver alsa_audio_driver = {
 924     .name           = "alsa",
 925     .descr          = "ALSA http://www.alsa-project.org",
 926     .init           = alsa_audio_init,
 927     .fini           = alsa_audio_fini,
 928     .pcm_ops        = &alsa_pcm_ops,
 929     .can_be_default = 1,
 930     .max_voices_out = INT_MAX,
 931     .max_voices_in  = INT_MAX,
 932     .voice_size_out = sizeof (ALSAVoiceOut),
 933     .voice_size_in  = sizeof (ALSAVoiceIn)
 934 };
 935
 936 static void register_audio_alsa(void)
 937 {
 938     audio_driver_register(&alsa_audio_driver);
 939 }
 940 type_init(register_audio_alsa);