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     int wpos;
  48     int pending;
  49     void *pcm_buf;
  50     snd_pcm_t *handle;
  51     struct pollhlp pollhlp;
  52     Audiodev *dev;
  53 } ALSAVoiceOut;
  54
  55 typedef struct ALSAVoiceIn {
  56     HWVoiceIn hw;
  57     snd_pcm_t *handle;
  58     void *pcm_buf;
  59     struct pollhlp pollhlp;
  60     Audiodev *dev;
  61 } ALSAVoiceIn;
  62
  63 struct alsa_params_req {
  64     int freq;
  65     snd_pcm_format_t fmt;
  66     int nchannels;
  67 };
  68
  69 struct alsa_params_obt {
  70     int freq;
  71     AudioFormat fmt;
  72     int endianness;
  73     int nchannels;
  74     snd_pcm_uframes_t samples;
  75 };
  76
  77 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
  78 {
  79     va_list ap;
  80
  81     va_start (ap, fmt);
  82     AUD_vlog (AUDIO_CAP, fmt, ap);
  83     va_end (ap);
  84
  85     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
  86 }
  87
  88 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
  89     int err,
  90     const char *typ,
  91     const char *fmt,
  92     ...
  93     )
  94 {
  95     va_list ap;
  96
  97     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
  98
  99     va_start (ap, fmt);
 100     AUD_vlog (AUDIO_CAP, fmt, ap);
 101     va_end (ap);
 102
 103     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 104 }
 105
 106 static void alsa_fini_poll (struct pollhlp *hlp)
 107 {
 108     int i;
 109     struct pollfd *pfds = hlp->pfds;
 110
 111     if (pfds) {
 112         for (i = 0; i < hlp->count; ++i) {
 113             qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
 114         }
 115         g_free (pfds);
 116     }
 117     hlp->pfds = NULL;
 118     hlp->count = 0;
 119     hlp->handle = NULL;
 120 }
 121
 122 static void alsa_anal_close1 (snd_pcm_t **handlep)
 123 {
 124     int err = snd_pcm_close (*handlep);
 125     if (err) {
 126         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 127     }
 128     *handlep = NULL;
 129 }
 130
 131 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
 132 {
 133     alsa_fini_poll (hlp);
 134     alsa_anal_close1 (handlep);
 135 }
 136
 137 static int alsa_recover (snd_pcm_t *handle)
 138 {
 139     int err = snd_pcm_prepare (handle);
 140     if (err < 0) {
 141         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 142         return -1;
 143     }
 144     return 0;
 145 }
 146
 147 static int alsa_resume (snd_pcm_t *handle)
 148 {
 149     int err = snd_pcm_resume (handle);
 150     if (err < 0) {
 151         alsa_logerr (err, "Failed to resume handle %p\n", handle);
 152         return -1;
 153     }
 154     return 0;
 155 }
 156
 157 static void alsa_poll_handler (void *opaque)
 158 {
 159     int err, count;
 160     snd_pcm_state_t state;
 161     struct pollhlp *hlp = opaque;
 162     unsigned short revents;
 163
 164     count = poll (hlp->pfds, hlp->count, 0);
 165     if (count < 0) {
 166         dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
 167         return;
 168     }
 169
 170     if (!count) {
 171         return;
 172     }
 173
 174     /* XXX: ALSA example uses initial count, not the one returned by
 175        poll, correct? */
 176     err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
 177                                             hlp->count, &revents);
 178     if (err < 0) {
 179         alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
 180         return;
 181     }
 182
 183     if (!(revents & hlp->mask)) {
 184         trace_alsa_revents(revents);
 185         return;
 186     }
 187
 188     state = snd_pcm_state (hlp->handle);
 189     switch (state) {
 190     case SND_PCM_STATE_SETUP:
 191         alsa_recover (hlp->handle);
 192         break;
 193
 194     case SND_PCM_STATE_XRUN:
 195         alsa_recover (hlp->handle);
 196         break;
 197
 198     case SND_PCM_STATE_SUSPENDED:
 199         alsa_resume (hlp->handle);
 200         break;
 201
 202     case SND_PCM_STATE_PREPARED:
 203         audio_run(hlp->s, "alsa run (prepared)");
 204         break;
 205
 206     case SND_PCM_STATE_RUNNING:
 207         audio_run(hlp->s, "alsa run (running)");
 208         break;
 209
 210     default:
 211         dolog ("Unexpected state %d\n", state);
 212     }
 213 }
 214
 215 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
 216 {
 217     int i, count, err;
 218     struct pollfd *pfds;
 219
 220     count = snd_pcm_poll_descriptors_count (handle);
 221     if (count <= 0) {
 222         dolog ("Could not initialize poll mode\n"
 223                "Invalid number of poll descriptors %d\n", count);
 224         return -1;
 225     }
 226
 227     pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
 228     if (!pfds) {
 229         dolog ("Could not initialize poll mode\n");
 230         return -1;
 231     }
 232
 233     err = snd_pcm_poll_descriptors (handle, pfds, count);
 234     if (err < 0) {
 235         alsa_logerr (err, "Could not initialize poll mode\n"
 236                      "Could not obtain poll descriptors\n");
 237         g_free (pfds);
 238         return -1;
 239     }
 240
 241     for (i = 0; i < count; ++i) {
 242         if (pfds[i].events & POLLIN) {
 243             qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
 244         }
 245         if (pfds[i].events & POLLOUT) {
 246             trace_alsa_pollout(i, pfds[i].fd);
 247             qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
 248         }
 249         trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
 250
 251     }
 252     hlp->pfds = pfds;
 253     hlp->count = count;
 254     hlp->handle = handle;
 255     hlp->mask = mask;
 256     return 0;
 257 }
 258
 259 static int alsa_poll_out (HWVoiceOut *hw)
 260 {
 261     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 262
 263     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
 264 }
 265
 266 static int alsa_poll_in (HWVoiceIn *hw)
 267 {
 268     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 269
 270     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
 271 }
 272
 273 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
 274 {
 275     return audio_pcm_sw_write (sw, buf, len);
 276 }
 277
 278 static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
 279 {
 280     switch (fmt) {
 281     case AUDIO_FORMAT_S8:
 282         return SND_PCM_FORMAT_S8;
 283
 284     case AUDIO_FORMAT_U8:
 285         return SND_PCM_FORMAT_U8;
 286
 287     case AUDIO_FORMAT_S16:
 288         if (endianness) {
 289             return SND_PCM_FORMAT_S16_BE;
 290         }
 291         else {
 292             return SND_PCM_FORMAT_S16_LE;
 293         }
 294
 295     case AUDIO_FORMAT_U16:
 296         if (endianness) {
 297             return SND_PCM_FORMAT_U16_BE;
 298         }
 299         else {
 300             return SND_PCM_FORMAT_U16_LE;
 301         }
 302
 303     case AUDIO_FORMAT_S32:
 304         if (endianness) {
 305             return SND_PCM_FORMAT_S32_BE;
 306         }
 307         else {
 308             return SND_PCM_FORMAT_S32_LE;
 309         }
 310
 311     case AUDIO_FORMAT_U32:
 312         if (endianness) {
 313             return SND_PCM_FORMAT_U32_BE;
 314         }
 315         else {
 316             return SND_PCM_FORMAT_U32_LE;
 317         }
 318
 319     default:
 320         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 321 #ifdef DEBUG_AUDIO
 322         abort ();
 323 #endif
 324         return SND_PCM_FORMAT_U8;
 325     }
 326 }
 327
 328 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
 329                            int *endianness)
 330 {
 331     switch (alsafmt) {
 332     case SND_PCM_FORMAT_S8:
 333         *endianness = 0;
 334         *fmt = AUDIO_FORMAT_S8;
 335         break;
 336
 337     case SND_PCM_FORMAT_U8:
 338         *endianness = 0;
 339         *fmt = AUDIO_FORMAT_U8;
 340         break;
 341
 342     case SND_PCM_FORMAT_S16_LE:
 343         *endianness = 0;
 344         *fmt = AUDIO_FORMAT_S16;
 345         break;
 346
 347     case SND_PCM_FORMAT_U16_LE:
 348         *endianness = 0;
 349         *fmt = AUDIO_FORMAT_U16;
 350         break;
 351
 352     case SND_PCM_FORMAT_S16_BE:
 353         *endianness = 1;
 354         *fmt = AUDIO_FORMAT_S16;
 355         break;
 356
 357     case SND_PCM_FORMAT_U16_BE:
 358         *endianness = 1;
 359         *fmt = AUDIO_FORMAT_U16;
 360         break;
 361
 362     case SND_PCM_FORMAT_S32_LE:
 363         *endianness = 0;
 364         *fmt = AUDIO_FORMAT_S32;
 365         break;
 366
 367     case SND_PCM_FORMAT_U32_LE:
 368         *endianness = 0;
 369         *fmt = AUDIO_FORMAT_U32;
 370         break;
 371
 372     case SND_PCM_FORMAT_S32_BE:
 373         *endianness = 1;
 374         *fmt = AUDIO_FORMAT_S32;
 375         break;
 376
 377     case SND_PCM_FORMAT_U32_BE:
 378         *endianness = 1;
 379         *fmt = AUDIO_FORMAT_U32;
 380         break;
 381
 382     default:
 383         dolog ("Unrecognized audio format %d\n", alsafmt);
 384         return -1;
 385     }
 386
 387     return 0;
 388 }
 389
 390 static void alsa_dump_info (struct alsa_params_req *req,
 391                             struct alsa_params_obt *obt,
 392                             snd_pcm_format_t obtfmt,
 393                             AudiodevAlsaPerDirectionOptions *apdo)
 394 {
 395     dolog("parameter | requested value | obtained value\n");
 396     dolog("format    |      %10d |     %10d\n", req->fmt, obtfmt);
 397     dolog("channels  |      %10d |     %10d\n",
 398           req->nchannels, obt->nchannels);
 399     dolog("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 400     dolog("============================================\n");
 401     dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
 402           apdo->buffer_length, apdo->period_length);
 403     dolog("obtained: samples %ld\n", obt->samples);
 404 }
 405
 406 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 407 {
 408     int err;
 409     snd_pcm_sw_params_t *sw_params;
 410
 411     snd_pcm_sw_params_alloca (&sw_params);
 412
 413     err = snd_pcm_sw_params_current (handle, sw_params);
 414     if (err < 0) {
 415         dolog ("Could not fully initialize DAC\n");
 416         alsa_logerr (err, "Failed to get current software parameters\n");
 417         return;
 418     }
 419
 420     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 421     if (err < 0) {
 422         dolog ("Could not fully initialize DAC\n");
 423         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 424                      threshold);
 425         return;
 426     }
 427
 428     err = snd_pcm_sw_params (handle, sw_params);
 429     if (err < 0) {
 430         dolog ("Could not fully initialize DAC\n");
 431         alsa_logerr (err, "Failed to set software parameters\n");
 432         return;
 433     }
 434 }
 435
 436 static int alsa_open(bool in, struct alsa_params_req *req,
 437                      struct alsa_params_obt *obt, snd_pcm_t **handlep,
 438                      Audiodev *dev)
 439 {
 440     AudiodevAlsaOptions *aopts = &dev->u.alsa;
 441     AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
 442     snd_pcm_t *handle;
 443     snd_pcm_hw_params_t *hw_params;
 444     int err;
 445     unsigned int freq, nchannels;
 446     const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
 447     snd_pcm_uframes_t obt_buffer_size;
 448     const char *typ = in ? "ADC" : "DAC";
 449     snd_pcm_format_t obtfmt;
 450
 451     freq = req->freq;
 452     nchannels = req->nchannels;
 453
 454     snd_pcm_hw_params_alloca (&hw_params);
 455
 456     err = snd_pcm_open (
 457         &handle,
 458         pcm_name,
 459         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 460         SND_PCM_NONBLOCK
 461         );
 462     if (err < 0) {
 463         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 464         return -1;
 465     }
 466
 467     err = snd_pcm_hw_params_any (handle, hw_params);
 468     if (err < 0) {
 469         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 470         goto err;
 471     }
 472
 473     err = snd_pcm_hw_params_set_access (
 474         handle,
 475         hw_params,
 476         SND_PCM_ACCESS_RW_INTERLEAVED
 477         );
 478     if (err < 0) {
 479         alsa_logerr2 (err, typ, "Failed to set access type\n");
 480         goto err;
 481     }
 482
 483     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 484     if (err < 0) {
 485         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 486     }
 487
 488     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 489     if (err < 0) {
 490         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 491         goto err;
 492     }
 493
 494     err = snd_pcm_hw_params_set_channels_near (
 495         handle,
 496         hw_params,
 497         &nchannels
 498         );
 499     if (err < 0) {
 500         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 501                       req->nchannels);
 502         goto err;
 503     }
 504
 505     if (nchannels != 1 && nchannels != 2) {
 506         alsa_logerr2 (err, typ,
 507                       "Can not handle obtained number of channels %d\n",
 508                       nchannels);
 509         goto err;
 510     }
 511
 512     if (apdo->buffer_length) {
 513         int dir = 0;
 514         unsigned int btime = apdo->buffer_length;
 515
 516         err = snd_pcm_hw_params_set_buffer_time_near(
 517             handle, hw_params, &btime, &dir);
 518
 519         if (err < 0) {
 520             alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
 521                          apdo->buffer_length);
 522             goto err;
 523         }
 524
 525         if (apdo->has_buffer_length && btime != apdo->buffer_length) {
 526             dolog("Requested buffer time %" PRId32
 527                   " was rejected, using %u\n", apdo->buffer_length, btime);
 528         }
 529     }
 530
 531     if (apdo->period_length) {
 532         int dir = 0;
 533         unsigned int ptime = apdo->period_length;
 534
 535         err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
 536                                                      &dir);
 537
 538         if (err < 0) {
 539             alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
 540                          apdo->period_length);
 541             goto err;
 542         }
 543
 544         if (apdo->has_period_length && ptime != apdo->period_length) {
 545             dolog("Requested period time %" PRId32 " was rejected, using %d\n",
 546                   apdo->period_length, ptime);
 547         }
 548     }
 549
 550     err = snd_pcm_hw_params (handle, hw_params);
 551     if (err < 0) {
 552         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 553         goto err;
 554     }
 555
 556     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 557     if (err < 0) {
 558         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 559         goto err;
 560     }
 561
 562     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 563     if (err < 0) {
 564         alsa_logerr2 (err, typ, "Failed to get format\n");
 565         goto err;
 566     }
 567
 568     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 569         dolog ("Invalid format was returned %d\n", obtfmt);
 570         goto err;
 571     }
 572
 573     err = snd_pcm_prepare (handle);
 574     if (err < 0) {
 575         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 576         goto err;
 577     }
 578
 579     if (!in && aopts->has_threshold && aopts->threshold) {
 580         struct audsettings as = { .freq = freq };
 581         alsa_set_threshold(
 582             handle,
 583             audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
 584                                 &as, aopts->threshold));
 585     }
 586
 587     obt->nchannels = nchannels;
 588     obt->freq = freq;
 589     obt->samples = obt_buffer_size;
 590
 591     *handlep = handle;
 592
 593     if (obtfmt != req->fmt ||
 594          obt->nchannels != req->nchannels ||
 595          obt->freq != req->freq) {
 596         dolog ("Audio parameters for %s\n", typ);
 597         alsa_dump_info(req, obt, obtfmt, apdo);
 598     }
 599
 600 #ifdef DEBUG
 601     alsa_dump_info(req, obt, obtfmt, pdo);
 602 #endif
 603     return 0;
 604
 605  err:
 606     alsa_anal_close1 (&handle);
 607     return -1;
 608 }
 609
 610 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 611 {
 612     snd_pcm_sframes_t avail;
 613
 614     avail = snd_pcm_avail_update (handle);
 615     if (avail < 0) {
 616         if (avail == -EPIPE) {
 617             if (!alsa_recover (handle)) {
 618                 avail = snd_pcm_avail_update (handle);
 619             }
 620         }
 621
 622         if (avail < 0) {
 623             alsa_logerr (avail,
 624                          "Could not obtain number of available frames\n");
 625             return -1;
 626         }
 627     }
 628
 629     return avail;
 630 }
 631
 632 static void alsa_write_pending (ALSAVoiceOut *alsa)
 633 {
 634     HWVoiceOut *hw = &alsa->hw;
 635
 636     while (alsa->pending) {
 637         int left_till_end_samples = hw->samples - alsa->wpos;
 638         int len = MIN (alsa->pending, left_till_end_samples);
 639         char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
 640
 641         while (len) {
 642             snd_pcm_sframes_t written;
 643
 644             written = snd_pcm_writei (alsa->handle, src, len);
 645
 646             if (written <= 0) {
 647                 switch (written) {
 648                 case 0:
 649                     trace_alsa_wrote_zero(len);
 650                     return;
 651
 652                 case -EPIPE:
 653                     if (alsa_recover (alsa->handle)) {
 654                         alsa_logerr (written, "Failed to write %d frames\n",
 655                                      len);
 656                         return;
 657                     }
 658                     trace_alsa_xrun_out();
 659                     continue;
 660
 661                 case -ESTRPIPE:
 662                     /* stream is suspended and waiting for an
 663                        application recovery */
 664                     if (alsa_resume (alsa->handle)) {
 665                         alsa_logerr (written, "Failed to write %d frames\n",
 666                                      len);
 667                         return;
 668                     }
 669                     trace_alsa_resume_out();
 670                     continue;
 671
 672                 case -EAGAIN:
 673                     return;
 674
 675                 default:
 676                     alsa_logerr (written, "Failed to write %d frames from %p\n",
 677                                  len, src);
 678                     return;
 679                 }
 680             }
 681
 682             alsa->wpos = (alsa->wpos + written) % hw->samples;
 683             alsa->pending -= written;
 684             len -= written;
 685         }
 686     }
 687 }
 688
 689 static int alsa_run_out (HWVoiceOut *hw, int live)
 690 {
 691     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 692     int decr;
 693     snd_pcm_sframes_t avail;
 694
 695     avail = alsa_get_avail (alsa->handle);
 696     if (avail < 0) {
 697         dolog ("Could not get number of available playback frames\n");
 698         return 0;
 699     }
 700
 701     decr = MIN (live, avail);
 702     decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
 703     alsa->pending += decr;
 704     alsa_write_pending (alsa);
 705     return decr;
 706 }
 707
 708 static void alsa_fini_out (HWVoiceOut *hw)
 709 {
 710     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 711
 712     ldebug ("alsa_fini\n");
 713     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 714
 715     g_free(alsa->pcm_buf);
 716     alsa->pcm_buf = NULL;
 717 }
 718
 719 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
 720                          void *drv_opaque)
 721 {
 722     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 723     struct alsa_params_req req;
 724     struct alsa_params_obt obt;
 725     snd_pcm_t *handle;
 726     struct audsettings obt_as;
 727     Audiodev *dev = drv_opaque;
 728
 729     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 730     req.freq = as->freq;
 731     req.nchannels = as->nchannels;
 732
 733     if (alsa_open(0, &req, &obt, &handle, dev)) {
 734         return -1;
 735     }
 736
 737     obt_as.freq = obt.freq;
 738     obt_as.nchannels = obt.nchannels;
 739     obt_as.fmt = obt.fmt;
 740     obt_as.endianness = obt.endianness;
 741
 742     audio_pcm_init_info (&hw->info, &obt_as);
 743     hw->samples = obt.samples;
 744
 745     alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
 746     if (!alsa->pcm_buf) {
 747         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 748                hw->samples, 1 << hw->info.shift);
 749         alsa_anal_close1 (&handle);
 750         return -1;
 751     }
 752
 753     alsa->pollhlp.s = hw->s;
 754     alsa->handle = handle;
 755     alsa->dev = dev;
 756     return 0;
 757 }
 758
 759 #define VOICE_CTL_PAUSE 0
 760 #define VOICE_CTL_PREPARE 1
 761 #define VOICE_CTL_START 2
 762
 763 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
 764 {
 765     int err;
 766
 767     if (ctl == VOICE_CTL_PAUSE) {
 768         err = snd_pcm_drop (handle);
 769         if (err < 0) {
 770             alsa_logerr (err, "Could not stop %s\n", typ);
 771             return -1;
 772         }
 773     }
 774     else {
 775         err = snd_pcm_prepare (handle);
 776         if (err < 0) {
 777             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 778             return -1;
 779         }
 780         if (ctl == VOICE_CTL_START) {
 781             err = snd_pcm_start(handle);
 782             if (err < 0) {
 783                 alsa_logerr (err, "Could not start handle for %s\n", typ);
 784                 return -1;
 785             }
 786         }
 787     }
 788
 789     return 0;
 790 }
 791
 792 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 793 {
 794     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 795     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
 796
 797     switch (cmd) {
 798     case VOICE_ENABLE:
 799         {
 800             bool poll_mode = apdo->try_poll;
 801
 802             ldebug ("enabling voice\n");
 803             if (poll_mode && alsa_poll_out (hw)) {
 804                 poll_mode = 0;
 805             }
 806             hw->poll_mode = poll_mode;
 807             return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
 808         }
 809
 810     case VOICE_DISABLE:
 811         ldebug ("disabling voice\n");
 812         if (hw->poll_mode) {
 813             hw->poll_mode = 0;
 814             alsa_fini_poll (&alsa->pollhlp);
 815         }
 816         return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
 817     }
 818
 819     return -1;
 820 }
 821
 822 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 823 {
 824     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 825     struct alsa_params_req req;
 826     struct alsa_params_obt obt;
 827     snd_pcm_t *handle;
 828     struct audsettings obt_as;
 829     Audiodev *dev = drv_opaque;
 830
 831     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 832     req.freq = as->freq;
 833     req.nchannels = as->nchannels;
 834
 835     if (alsa_open(1, &req, &obt, &handle, dev)) {
 836         return -1;
 837     }
 838
 839     obt_as.freq = obt.freq;
 840     obt_as.nchannels = obt.nchannels;
 841     obt_as.fmt = obt.fmt;
 842     obt_as.endianness = obt.endianness;
 843
 844     audio_pcm_init_info (&hw->info, &obt_as);
 845     hw->samples = obt.samples;
 846
 847     alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
 848     if (!alsa->pcm_buf) {
 849         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 850                hw->samples, 1 << hw->info.shift);
 851         alsa_anal_close1 (&handle);
 852         return -1;
 853     }
 854
 855     alsa->pollhlp.s = hw->s;
 856     alsa->handle = handle;
 857     alsa->dev = dev;
 858     return 0;
 859 }
 860
 861 static void alsa_fini_in (HWVoiceIn *hw)
 862 {
 863     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 864
 865     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 866
 867     g_free(alsa->pcm_buf);
 868     alsa->pcm_buf = NULL;
 869 }
 870
 871 static int alsa_run_in (HWVoiceIn *hw)
 872 {
 873     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 874     int hwshift = hw->info.shift;
 875     int i;
 876     int live = audio_pcm_hw_get_live_in (hw);
 877     int dead = hw->samples - live;
 878     int decr;
 879     struct {
 880         int add;
 881         int len;
 882     } bufs[2] = {
 883         { .add = hw->wpos, .len = 0 },
 884         { .add = 0,        .len = 0 }
 885     };
 886     snd_pcm_sframes_t avail;
 887     snd_pcm_uframes_t read_samples = 0;
 888
 889     if (!dead) {
 890         return 0;
 891     }
 892
 893     avail = alsa_get_avail (alsa->handle);
 894     if (avail < 0) {
 895         dolog ("Could not get number of captured frames\n");
 896         return 0;
 897     }
 898
 899     if (!avail) {
 900         snd_pcm_state_t state;
 901
 902         state = snd_pcm_state (alsa->handle);
 903         switch (state) {
 904         case SND_PCM_STATE_PREPARED:
 905             avail = hw->samples;
 906             break;
 907         case SND_PCM_STATE_SUSPENDED:
 908             /* stream is suspended and waiting for an application recovery */
 909             if (alsa_resume (alsa->handle)) {
 910                 dolog ("Failed to resume suspended input stream\n");
 911                 return 0;
 912             }
 913             trace_alsa_resume_in();
 914             break;
 915         default:
 916             trace_alsa_no_frames(state);
 917             return 0;
 918         }
 919     }
 920
 921     decr = MIN (dead, avail);
 922     if (!decr) {
 923         return 0;
 924     }
 925
 926     if (hw->wpos + decr > hw->samples) {
 927         bufs[0].len = (hw->samples - hw->wpos);
 928         bufs[1].len = (decr - (hw->samples - hw->wpos));
 929     }
 930     else {
 931         bufs[0].len = decr;
 932     }
 933
 934     for (i = 0; i < 2; ++i) {
 935         void *src;
 936         struct st_sample *dst;
 937         snd_pcm_sframes_t nread;
 938         snd_pcm_uframes_t len;
 939
 940         len = bufs[i].len;
 941
 942         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
 943         dst = hw->conv_buf + bufs[i].add;
 944
 945         while (len) {
 946             nread = snd_pcm_readi (alsa->handle, src, len);
 947
 948             if (nread <= 0) {
 949                 switch (nread) {
 950                 case 0:
 951                     trace_alsa_read_zero(len);
 952                     goto exit;
 953
 954                 case -EPIPE:
 955                     if (alsa_recover (alsa->handle)) {
 956                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
 957                         goto exit;
 958                     }
 959                     trace_alsa_xrun_in();
 960                     continue;
 961
 962                 case -EAGAIN:
 963                     goto exit;
 964
 965                 default:
 966                     alsa_logerr (
 967                         nread,
 968                         "Failed to read %ld frames from %p\n",
 969                         len,
 970                         src
 971                         );
 972                     goto exit;
 973                 }
 974             }
 975
 976             hw->conv (dst, src, nread);
 977
 978             src = advance (src, nread << hwshift);
 979             dst += nread;
 980
 981             read_samples += nread;
 982             len -= nread;
 983         }
 984     }
 985
 986  exit:
 987     hw->wpos = (hw->wpos + read_samples) % hw->samples;
 988     return read_samples;
 989 }
 990
 991 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
 992 {
 993     return audio_pcm_sw_read (sw, buf, size);
 994 }
 995
 996 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 997 {
 998     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 999     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
1000
1001     switch (cmd) {
1002     case VOICE_ENABLE:
1003         {
1004             bool poll_mode = apdo->try_poll;
1005
1006             ldebug ("enabling voice\n");
1007             if (poll_mode && alsa_poll_in (hw)) {
1008                 poll_mode = 0;
1009             }
1010             hw->poll_mode = poll_mode;
1011
1012             return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
1013         }
1014
1015     case VOICE_DISABLE:
1016         ldebug ("disabling voice\n");
1017         if (hw->poll_mode) {
1018             hw->poll_mode = 0;
1019             alsa_fini_poll (&alsa->pollhlp);
1020         }
1021         return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
1022     }
1023
1024     return -1;
1025 }
1026
1027 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
1028 {
1029     if (!apdo->has_try_poll) {
1030         apdo->try_poll = true;
1031         apdo->has_try_poll = true;
1032     }
1033 }
1034
1035 static void *alsa_audio_init(Audiodev *dev)
1036 {
1037     AudiodevAlsaOptions *aopts;
1038     assert(dev->driver == AUDIODEV_DRIVER_ALSA);
1039
1040     aopts = &dev->u.alsa;
1041     alsa_init_per_direction(aopts->in);
1042     alsa_init_per_direction(aopts->out);
1043
1044     /*
1045      * need to define them, as otherwise alsa produces no sound
1046      * doesn't set has_* so alsa_open can identify it wasn't set by the user
1047      */
1048     if (!dev->u.alsa.out->has_period_length) {
1049         /* 1024 frames assuming 44100Hz */
1050         dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
1051     }
1052     if (!dev->u.alsa.out->has_buffer_length) {
1053         /* 4096 frames assuming 44100Hz */
1054         dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
1055     }
1056
1057     /*
1058      * OptsVisitor sets unspecified optional fields to zero, but do not depend
1059      * on it...
1060      */
1061     if (!dev->u.alsa.in->has_period_length) {
1062         dev->u.alsa.in->period_length = 0;
1063     }
1064     if (!dev->u.alsa.in->has_buffer_length) {
1065         dev->u.alsa.in->buffer_length = 0;
1066     }
1067
1068     return dev;
1069 }
1070
1071 static void alsa_audio_fini (void *opaque)
1072 {
1073 }
1074
1075 static struct audio_pcm_ops alsa_pcm_ops = {
1076     .init_out = alsa_init_out,
1077     .fini_out = alsa_fini_out,
1078     .run_out  = alsa_run_out,
1079     .write    = alsa_write,
1080     .ctl_out  = alsa_ctl_out,
1081
1082     .init_in  = alsa_init_in,
1083     .fini_in  = alsa_fini_in,
1084     .run_in   = alsa_run_in,
1085     .read     = alsa_read,
1086     .ctl_in   = alsa_ctl_in,
1087 };
1088
1089 static struct audio_driver alsa_audio_driver = {
1090     .name           = "alsa",
1091     .descr          = "ALSA http://www.alsa-project.org",
1092     .init           = alsa_audio_init,
1093     .fini           = alsa_audio_fini,
1094     .pcm_ops        = &alsa_pcm_ops,
1095     .can_be_default = 1,
1096     .max_voices_out = INT_MAX,
1097     .max_voices_in  = INT_MAX,
1098     .voice_size_out = sizeof (ALSAVoiceOut),
1099     .voice_size_in  = sizeof (ALSAVoiceIn)
1100 };
1101
1102 static void register_audio_alsa(void)
1103 {
1104     audio_driver_register(&alsa_audio_driver);
1105 }
1106 type_init(register_audio_alsa);