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         } else {
 282             return SND_PCM_FORMAT_S16_LE;
 283         }
 284
 285     case AUDIO_FORMAT_U16:
 286         if (endianness) {
 287             return SND_PCM_FORMAT_U16_BE;
 288         } else {
 289             return SND_PCM_FORMAT_U16_LE;
 290         }
 291
 292     case AUDIO_FORMAT_S32:
 293         if (endianness) {
 294             return SND_PCM_FORMAT_S32_BE;
 295         } else {
 296             return SND_PCM_FORMAT_S32_LE;
 297         }
 298
 299     case AUDIO_FORMAT_U32:
 300         if (endianness) {
 301             return SND_PCM_FORMAT_U32_BE;
 302         } else {
 303             return SND_PCM_FORMAT_U32_LE;
 304         }
 305
 306     case AUDIO_FORMAT_F32:
 307         if (endianness) {
 308             return SND_PCM_FORMAT_FLOAT_BE;
 309         } else {
 310             return SND_PCM_FORMAT_FLOAT_LE;
 311         }
 312
 313     default:
 314         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 315 #ifdef DEBUG_AUDIO
 316         abort ();
 317 #endif
 318         return SND_PCM_FORMAT_U8;
 319     }
 320 }
 321
 322 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
 323                            int *endianness)
 324 {
 325     switch (alsafmt) {
 326     case SND_PCM_FORMAT_S8:
 327         *endianness = 0;
 328         *fmt = AUDIO_FORMAT_S8;
 329         break;
 330
 331     case SND_PCM_FORMAT_U8:
 332         *endianness = 0;
 333         *fmt = AUDIO_FORMAT_U8;
 334         break;
 335
 336     case SND_PCM_FORMAT_S16_LE:
 337         *endianness = 0;
 338         *fmt = AUDIO_FORMAT_S16;
 339         break;
 340
 341     case SND_PCM_FORMAT_U16_LE:
 342         *endianness = 0;
 343         *fmt = AUDIO_FORMAT_U16;
 344         break;
 345
 346     case SND_PCM_FORMAT_S16_BE:
 347         *endianness = 1;
 348         *fmt = AUDIO_FORMAT_S16;
 349         break;
 350
 351     case SND_PCM_FORMAT_U16_BE:
 352         *endianness = 1;
 353         *fmt = AUDIO_FORMAT_U16;
 354         break;
 355
 356     case SND_PCM_FORMAT_S32_LE:
 357         *endianness = 0;
 358         *fmt = AUDIO_FORMAT_S32;
 359         break;
 360
 361     case SND_PCM_FORMAT_U32_LE:
 362         *endianness = 0;
 363         *fmt = AUDIO_FORMAT_U32;
 364         break;
 365
 366     case SND_PCM_FORMAT_S32_BE:
 367         *endianness = 1;
 368         *fmt = AUDIO_FORMAT_S32;
 369         break;
 370
 371     case SND_PCM_FORMAT_U32_BE:
 372         *endianness = 1;
 373         *fmt = AUDIO_FORMAT_U32;
 374         break;
 375
 376     case SND_PCM_FORMAT_FLOAT_LE:
 377         *endianness = 0;
 378         *fmt = AUDIO_FORMAT_F32;
 379         break;
 380
 381     case SND_PCM_FORMAT_FLOAT_BE:
 382         *endianness = 1;
 383         *fmt = AUDIO_FORMAT_F32;
 384         break;
 385
 386     default:
 387         dolog ("Unrecognized audio format %d\n", alsafmt);
 388         return -1;
 389     }
 390
 391     return 0;
 392 }
 393
 394 static void alsa_dump_info (struct alsa_params_req *req,
 395                             struct alsa_params_obt *obt,
 396                             snd_pcm_format_t obtfmt,
 397                             AudiodevAlsaPerDirectionOptions *apdo)
 398 {
 399     dolog("parameter | requested value | obtained value\n");
 400     dolog("format    |      %10d |     %10d\n", req->fmt, obtfmt);
 401     dolog("channels  |      %10d |     %10d\n",
 402           req->nchannels, obt->nchannels);
 403     dolog("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 404     dolog("============================================\n");
 405     dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
 406           apdo->buffer_length, apdo->period_length);
 407     dolog("obtained: samples %ld\n", obt->samples);
 408 }
 409
 410 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 411 {
 412     int err;
 413     snd_pcm_sw_params_t *sw_params;
 414
 415     snd_pcm_sw_params_alloca (&sw_params);
 416
 417     err = snd_pcm_sw_params_current (handle, sw_params);
 418     if (err < 0) {
 419         dolog ("Could not fully initialize DAC\n");
 420         alsa_logerr (err, "Failed to get current software parameters\n");
 421         return;
 422     }
 423
 424     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 425     if (err < 0) {
 426         dolog ("Could not fully initialize DAC\n");
 427         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 428                      threshold);
 429         return;
 430     }
 431
 432     err = snd_pcm_sw_params (handle, sw_params);
 433     if (err < 0) {
 434         dolog ("Could not fully initialize DAC\n");
 435         alsa_logerr (err, "Failed to set software parameters\n");
 436         return;
 437     }
 438 }
 439
 440 static int alsa_open(bool in, struct alsa_params_req *req,
 441                      struct alsa_params_obt *obt, snd_pcm_t **handlep,
 442                      Audiodev *dev)
 443 {
 444     AudiodevAlsaOptions *aopts = &dev->u.alsa;
 445     AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
 446     snd_pcm_t *handle;
 447     snd_pcm_hw_params_t *hw_params;
 448     int err;
 449     unsigned int freq, nchannels;
 450     const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
 451     snd_pcm_uframes_t obt_buffer_size;
 452     const char *typ = in ? "ADC" : "DAC";
 453     snd_pcm_format_t obtfmt;
 454
 455     freq = req->freq;
 456     nchannels = req->nchannels;
 457
 458     snd_pcm_hw_params_alloca (&hw_params);
 459
 460     err = snd_pcm_open (
 461         &handle,
 462         pcm_name,
 463         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 464         SND_PCM_NONBLOCK
 465         );
 466     if (err < 0) {
 467         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 468         return -1;
 469     }
 470
 471     err = snd_pcm_hw_params_any (handle, hw_params);
 472     if (err < 0) {
 473         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 474         goto err;
 475     }
 476
 477     err = snd_pcm_hw_params_set_access (
 478         handle,
 479         hw_params,
 480         SND_PCM_ACCESS_RW_INTERLEAVED
 481         );
 482     if (err < 0) {
 483         alsa_logerr2 (err, typ, "Failed to set access type\n");
 484         goto err;
 485     }
 486
 487     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 488     if (err < 0) {
 489         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 490     }
 491
 492     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 493     if (err < 0) {
 494         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 495         goto err;
 496     }
 497
 498     err = snd_pcm_hw_params_set_channels_near (
 499         handle,
 500         hw_params,
 501         &nchannels
 502         );
 503     if (err < 0) {
 504         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 505                       req->nchannels);
 506         goto err;
 507     }
 508
 509     if (apdo->buffer_length) {
 510         int dir = 0;
 511         unsigned int btime = apdo->buffer_length;
 512
 513         err = snd_pcm_hw_params_set_buffer_time_near(
 514             handle, hw_params, &btime, &dir);
 515
 516         if (err < 0) {
 517             alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
 518                          apdo->buffer_length);
 519             goto err;
 520         }
 521
 522         if (apdo->has_buffer_length && btime != apdo->buffer_length) {
 523             dolog("Requested buffer time %" PRId32
 524                   " was rejected, using %u\n", apdo->buffer_length, btime);
 525         }
 526     }
 527
 528     if (apdo->period_length) {
 529         int dir = 0;
 530         unsigned int ptime = apdo->period_length;
 531
 532         err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
 533                                                      &dir);
 534
 535         if (err < 0) {
 536             alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
 537                          apdo->period_length);
 538             goto err;
 539         }
 540
 541         if (apdo->has_period_length && ptime != apdo->period_length) {
 542             dolog("Requested period time %" PRId32 " was rejected, using %d\n",
 543                   apdo->period_length, ptime);
 544         }
 545     }
 546
 547     err = snd_pcm_hw_params (handle, hw_params);
 548     if (err < 0) {
 549         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 550         goto err;
 551     }
 552
 553     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 554     if (err < 0) {
 555         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 556         goto err;
 557     }
 558
 559     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 560     if (err < 0) {
 561         alsa_logerr2 (err, typ, "Failed to get format\n");
 562         goto err;
 563     }
 564
 565     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 566         dolog ("Invalid format was returned %d\n", obtfmt);
 567         goto err;
 568     }
 569
 570     err = snd_pcm_prepare (handle);
 571     if (err < 0) {
 572         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 573         goto err;
 574     }
 575
 576     if (!in && aopts->has_threshold && aopts->threshold) {
 577         struct audsettings as = { .freq = freq };
 578         alsa_set_threshold(
 579             handle,
 580             audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
 581                                 &as, aopts->threshold));
 582     }
 583
 584     obt->nchannels = nchannels;
 585     obt->freq = freq;
 586     obt->samples = obt_buffer_size;
 587
 588     *handlep = handle;
 589
 590     if (obtfmt != req->fmt ||
 591          obt->nchannels != req->nchannels ||
 592          obt->freq != req->freq) {
 593         dolog ("Audio parameters for %s\n", typ);
 594         alsa_dump_info(req, obt, obtfmt, apdo);
 595     }
 596
 597 #ifdef DEBUG
 598     alsa_dump_info(req, obt, obtfmt, apdo);
 599 #endif
 600     return 0;
 601
 602  err:
 603     alsa_anal_close1 (&handle);
 604     return -1;
 605 }
 606
 607 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
 608 {
 609     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 610     size_t pos = 0;
 611     size_t len_frames = len / hw->info.bytes_per_frame;
 612
 613     while (len_frames) {
 614         char *src = advance(buf, pos);
 615         snd_pcm_sframes_t written;
 616
 617         written = snd_pcm_writei(alsa->handle, src, len_frames);
 618
 619         if (written <= 0) {
 620             switch (written) {
 621             case 0:
 622                 trace_alsa_wrote_zero(len_frames);
 623                 return pos;
 624
 625             case -EPIPE:
 626                 if (alsa_recover(alsa->handle)) {
 627                     alsa_logerr(written, "Failed to write %zu frames\n",
 628                                 len_frames);
 629                     return pos;
 630                 }
 631                 trace_alsa_xrun_out();
 632                 continue;
 633
 634             case -ESTRPIPE:
 635                 /*
 636                  * stream is suspended and waiting for an application
 637                  * recovery
 638                  */
 639                 if (alsa_resume(alsa->handle)) {
 640                     alsa_logerr(written, "Failed to write %zu frames\n",
 641                                 len_frames);
 642                     return pos;
 643                 }
 644                 trace_alsa_resume_out();
 645                 continue;
 646
 647             case -EAGAIN:
 648                 return pos;
 649
 650             default:
 651                 alsa_logerr(written, "Failed to write %zu frames from %p\n",
 652                             len, src);
 653                 return pos;
 654             }
 655         }
 656
 657         pos += written * hw->info.bytes_per_frame;
 658         if (written < len_frames) {
 659             break;
 660         }
 661         len_frames -= written;
 662     }
 663
 664     return pos;
 665 }
 666
 667 static void alsa_fini_out (HWVoiceOut *hw)
 668 {
 669     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 670
 671     ldebug ("alsa_fini\n");
 672     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 673 }
 674
 675 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
 676                          void *drv_opaque)
 677 {
 678     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 679     struct alsa_params_req req;
 680     struct alsa_params_obt obt;
 681     snd_pcm_t *handle;
 682     struct audsettings obt_as;
 683     Audiodev *dev = drv_opaque;
 684
 685     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 686     req.freq = as->freq;
 687     req.nchannels = as->nchannels;
 688
 689     if (alsa_open(0, &req, &obt, &handle, dev)) {
 690         return -1;
 691     }
 692
 693     obt_as.freq = obt.freq;
 694     obt_as.nchannels = obt.nchannels;
 695     obt_as.fmt = obt.fmt;
 696     obt_as.endianness = obt.endianness;
 697
 698     audio_pcm_init_info (&hw->info, &obt_as);
 699     hw->samples = obt.samples;
 700
 701     alsa->pollhlp.s = hw->s;
 702     alsa->handle = handle;
 703     alsa->dev = dev;
 704     return 0;
 705 }
 706
 707 #define VOICE_CTL_PAUSE 0
 708 #define VOICE_CTL_PREPARE 1
 709 #define VOICE_CTL_START 2
 710
 711 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
 712 {
 713     int err;
 714
 715     if (ctl == VOICE_CTL_PAUSE) {
 716         err = snd_pcm_drop (handle);
 717         if (err < 0) {
 718             alsa_logerr (err, "Could not stop %s\n", typ);
 719             return -1;
 720         }
 721     } else {
 722         err = snd_pcm_prepare (handle);
 723         if (err < 0) {
 724             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 725             return -1;
 726         }
 727         if (ctl == VOICE_CTL_START) {
 728             err = snd_pcm_start(handle);
 729             if (err < 0) {
 730                 alsa_logerr (err, "Could not start handle for %s\n", typ);
 731                 return -1;
 732             }
 733         }
 734     }
 735
 736     return 0;
 737 }
 738
 739 static void alsa_enable_out(HWVoiceOut *hw, bool enable)
 740 {
 741     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 742     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
 743
 744     if (enable) {
 745         bool poll_mode = apdo->try_poll;
 746
 747         ldebug("enabling voice\n");
 748         if (poll_mode && alsa_poll_out(hw)) {
 749             poll_mode = 0;
 750         }
 751         hw->poll_mode = poll_mode;
 752         alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
 753     } else {
 754         ldebug("disabling voice\n");
 755         if (hw->poll_mode) {
 756             hw->poll_mode = 0;
 757             alsa_fini_poll(&alsa->pollhlp);
 758         }
 759         alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
 760     }
 761 }
 762
 763 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 764 {
 765     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 766     struct alsa_params_req req;
 767     struct alsa_params_obt obt;
 768     snd_pcm_t *handle;
 769     struct audsettings obt_as;
 770     Audiodev *dev = drv_opaque;
 771
 772     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 773     req.freq = as->freq;
 774     req.nchannels = as->nchannels;
 775
 776     if (alsa_open(1, &req, &obt, &handle, dev)) {
 777         return -1;
 778     }
 779
 780     obt_as.freq = obt.freq;
 781     obt_as.nchannels = obt.nchannels;
 782     obt_as.fmt = obt.fmt;
 783     obt_as.endianness = obt.endianness;
 784
 785     audio_pcm_init_info (&hw->info, &obt_as);
 786     hw->samples = obt.samples;
 787
 788     alsa->pollhlp.s = hw->s;
 789     alsa->handle = handle;
 790     alsa->dev = dev;
 791     return 0;
 792 }
 793
 794 static void alsa_fini_in (HWVoiceIn *hw)
 795 {
 796     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 797
 798     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 799 }
 800
 801 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
 802 {
 803     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 804     size_t pos = 0;
 805
 806     while (len) {
 807         void *dst = advance(buf, pos);
 808         snd_pcm_sframes_t nread;
 809
 810         nread = snd_pcm_readi(
 811             alsa->handle, dst, len / hw->info.bytes_per_frame);
 812
 813         if (nread <= 0) {
 814             switch (nread) {
 815             case 0:
 816                 trace_alsa_read_zero(len);
 817                 return pos;
 818
 819             case -EPIPE:
 820                 if (alsa_recover(alsa->handle)) {
 821                     alsa_logerr(nread, "Failed to read %zu frames\n", len);
 822                     return pos;
 823                 }
 824                 trace_alsa_xrun_in();
 825                 continue;
 826
 827             case -EAGAIN:
 828                 return pos;
 829
 830             default:
 831                 alsa_logerr(nread, "Failed to read %zu frames to %p\n",
 832                             len, dst);
 833                 return pos;
 834             }
 835         }
 836
 837         pos += nread * hw->info.bytes_per_frame;
 838         len -= nread * hw->info.bytes_per_frame;
 839     }
 840
 841     return pos;
 842 }
 843
 844 static void alsa_enable_in(HWVoiceIn *hw, bool enable)
 845 {
 846     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 847     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
 848
 849     if (enable) {
 850         bool poll_mode = apdo->try_poll;
 851
 852         ldebug("enabling voice\n");
 853         if (poll_mode && alsa_poll_in(hw)) {
 854             poll_mode = 0;
 855         }
 856         hw->poll_mode = poll_mode;
 857
 858         alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
 859     } else {
 860         ldebug ("disabling voice\n");
 861         if (hw->poll_mode) {
 862             hw->poll_mode = 0;
 863             alsa_fini_poll(&alsa->pollhlp);
 864         }
 865         alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
 866     }
 867 }
 868
 869 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
 870 {
 871     if (!apdo->has_try_poll) {
 872         apdo->try_poll = true;
 873         apdo->has_try_poll = true;
 874     }
 875 }
 876
 877 static void *alsa_audio_init(Audiodev *dev)
 878 {
 879     AudiodevAlsaOptions *aopts;
 880     assert(dev->driver == AUDIODEV_DRIVER_ALSA);
 881
 882     aopts = &dev->u.alsa;
 883     alsa_init_per_direction(aopts->in);
 884     alsa_init_per_direction(aopts->out);
 885
 886     /*
 887      * need to define them, as otherwise alsa produces no sound
 888      * doesn't set has_* so alsa_open can identify it wasn't set by the user
 889      */
 890     if (!dev->u.alsa.out->has_period_length) {
 891         /* 1024 frames assuming 44100Hz */
 892         dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
 893     }
 894     if (!dev->u.alsa.out->has_buffer_length) {
 895         /* 4096 frames assuming 44100Hz */
 896         dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
 897     }
 898
 899     /*
 900      * OptsVisitor sets unspecified optional fields to zero, but do not depend
 901      * on it...
 902      */
 903     if (!dev->u.alsa.in->has_period_length) {
 904         dev->u.alsa.in->period_length = 0;
 905     }
 906     if (!dev->u.alsa.in->has_buffer_length) {
 907         dev->u.alsa.in->buffer_length = 0;
 908     }
 909
 910     return dev;
 911 }
 912
 913 static void alsa_audio_fini (void *opaque)
 914 {
 915 }
 916
 917 static struct audio_pcm_ops alsa_pcm_ops = {
 918     .init_out = alsa_init_out,
 919     .fini_out = alsa_fini_out,
 920     .write    = alsa_write,
 921     .run_buffer_out = audio_generic_run_buffer_out,
 922     .enable_out = alsa_enable_out,
 923
 924     .init_in  = alsa_init_in,
 925     .fini_in  = alsa_fini_in,
 926     .read     = alsa_read,
 927     .run_buffer_in = audio_generic_run_buffer_in,
 928     .enable_in = alsa_enable_in,
 929 };
 930
 931 static struct audio_driver alsa_audio_driver = {
 932     .name           = "alsa",
 933     .descr          = "ALSA http://www.alsa-project.org",
 934     .init           = alsa_audio_init,
 935     .fini           = alsa_audio_fini,
 936     .pcm_ops        = &alsa_pcm_ops,
 937     .can_be_default = 1,
 938     .max_voices_out = INT_MAX,
 939     .max_voices_in  = INT_MAX,
 940     .voice_size_out = sizeof (ALSAVoiceOut),
 941     .voice_size_in  = sizeof (ALSAVoiceIn)
 942 };
 943
 944 static void register_audio_alsa(void)
 945 {
 946     audio_driver_register(&alsa_audio_driver);
 947 }
 948 type_init(register_audio_alsa);