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 #define DEBUG_ALSA 0
  38
  39 struct pollhlp {
  40     snd_pcm_t *handle;
  41     struct pollfd *pfds;
  42     int count;
  43     int mask;
  44     AudioState *s;
  45 };
  46
  47 typedef struct ALSAVoiceOut {
  48     HWVoiceOut hw;
  49     snd_pcm_t *handle;
  50     struct pollhlp pollhlp;
  51     Audiodev *dev;
  52 } ALSAVoiceOut;
  53
  54 typedef struct ALSAVoiceIn {
  55     HWVoiceIn hw;
  56     snd_pcm_t *handle;
  57     struct pollhlp pollhlp;
  58     Audiodev *dev;
  59 } ALSAVoiceIn;
  60
  61 struct alsa_params_req {
  62     int freq;
  63     snd_pcm_format_t fmt;
  64     int nchannels;
  65 };
  66
  67 struct alsa_params_obt {
  68     int freq;
  69     AudioFormat fmt;
  70     int endianness;
  71     int nchannels;
  72     snd_pcm_uframes_t samples;
  73 };
  74
  75 static void G_GNUC_PRINTF (2, 3) alsa_logerr (int err, const char *fmt, ...)
  76 {
  77     va_list ap;
  78
  79     va_start (ap, fmt);
  80     AUD_vlog (AUDIO_CAP, fmt, ap);
  81     va_end (ap);
  82
  83     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
  84 }
  85
  86 static void G_GNUC_PRINTF (3, 4) alsa_logerr2 (
  87     int err,
  88     const char *typ,
  89     const char *fmt,
  90     ...
  91     )
  92 {
  93     va_list ap;
  94
  95     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
  96
  97     va_start (ap, fmt);
  98     AUD_vlog (AUDIO_CAP, fmt, ap);
  99     va_end (ap);
 100
 101     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 102 }
 103
 104 static void alsa_fini_poll (struct pollhlp *hlp)
 105 {
 106     int i;
 107     struct pollfd *pfds = hlp->pfds;
 108
 109     if (pfds) {
 110         for (i = 0; i < hlp->count; ++i) {
 111             qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
 112         }
 113         g_free (pfds);
 114     }
 115     hlp->pfds = NULL;
 116     hlp->count = 0;
 117     hlp->handle = NULL;
 118 }
 119
 120 static void alsa_anal_close1 (snd_pcm_t **handlep)
 121 {
 122     int err = snd_pcm_close (*handlep);
 123     if (err) {
 124         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 125     }
 126     *handlep = NULL;
 127 }
 128
 129 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
 130 {
 131     alsa_fini_poll (hlp);
 132     alsa_anal_close1 (handlep);
 133 }
 134
 135 static int alsa_recover (snd_pcm_t *handle)
 136 {
 137     int err = snd_pcm_prepare (handle);
 138     if (err < 0) {
 139         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 140         return -1;
 141     }
 142     return 0;
 143 }
 144
 145 static int alsa_resume (snd_pcm_t *handle)
 146 {
 147     int err = snd_pcm_resume (handle);
 148     if (err < 0) {
 149         alsa_logerr (err, "Failed to resume handle %p\n", handle);
 150         return -1;
 151     }
 152     return 0;
 153 }
 154
 155 static void alsa_poll_handler (void *opaque)
 156 {
 157     int err, count;
 158     snd_pcm_state_t state;
 159     struct pollhlp *hlp = opaque;
 160     unsigned short revents;
 161
 162     count = poll (hlp->pfds, hlp->count, 0);
 163     if (count < 0) {
 164         dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
 165         return;
 166     }
 167
 168     if (!count) {
 169         return;
 170     }
 171
 172     /* XXX: ALSA example uses initial count, not the one returned by
 173        poll, correct? */
 174     err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
 175                                             hlp->count, &revents);
 176     if (err < 0) {
 177         alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
 178         return;
 179     }
 180
 181     if (!(revents & hlp->mask)) {
 182         trace_alsa_revents(revents);
 183         return;
 184     }
 185
 186     state = snd_pcm_state (hlp->handle);
 187     switch (state) {
 188     case SND_PCM_STATE_SETUP:
 189         alsa_recover (hlp->handle);
 190         break;
 191
 192     case SND_PCM_STATE_XRUN:
 193         alsa_recover (hlp->handle);
 194         break;
 195
 196     case SND_PCM_STATE_SUSPENDED:
 197         alsa_resume (hlp->handle);
 198         break;
 199
 200     case SND_PCM_STATE_PREPARED:
 201         audio_run(hlp->s, "alsa run (prepared)");
 202         break;
 203
 204     case SND_PCM_STATE_RUNNING:
 205         audio_run(hlp->s, "alsa run (running)");
 206         break;
 207
 208     default:
 209         dolog ("Unexpected state %d\n", state);
 210     }
 211 }
 212
 213 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
 214 {
 215     int i, count, err;
 216     struct pollfd *pfds;
 217
 218     count = snd_pcm_poll_descriptors_count (handle);
 219     if (count <= 0) {
 220         dolog ("Could not initialize poll mode\n"
 221                "Invalid number of poll descriptors %d\n", count);
 222         return -1;
 223     }
 224
 225     pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
 226     if (!pfds) {
 227         dolog ("Could not initialize poll mode\n");
 228         return -1;
 229     }
 230
 231     err = snd_pcm_poll_descriptors (handle, pfds, count);
 232     if (err < 0) {
 233         alsa_logerr (err, "Could not initialize poll mode\n"
 234                      "Could not obtain poll descriptors\n");
 235         g_free (pfds);
 236         return -1;
 237     }
 238
 239     for (i = 0; i < count; ++i) {
 240         if (pfds[i].events & POLLIN) {
 241             qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
 242         }
 243         if (pfds[i].events & POLLOUT) {
 244             trace_alsa_pollout(i, pfds[i].fd);
 245             qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
 246         }
 247         trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
 248
 249     }
 250     hlp->pfds = pfds;
 251     hlp->count = count;
 252     hlp->handle = handle;
 253     hlp->mask = mask;
 254     return 0;
 255 }
 256
 257 static int alsa_poll_out (HWVoiceOut *hw)
 258 {
 259     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 260
 261     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
 262 }
 263
 264 static int alsa_poll_in (HWVoiceIn *hw)
 265 {
 266     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 267
 268     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
 269 }
 270
 271 static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
 272 {
 273     switch (fmt) {
 274     case AUDIO_FORMAT_S8:
 275         return SND_PCM_FORMAT_S8;
 276
 277     case AUDIO_FORMAT_U8:
 278         return SND_PCM_FORMAT_U8;
 279
 280     case AUDIO_FORMAT_S16:
 281         if (endianness) {
 282             return SND_PCM_FORMAT_S16_BE;
 283         } else {
 284             return SND_PCM_FORMAT_S16_LE;
 285         }
 286
 287     case AUDIO_FORMAT_U16:
 288         if (endianness) {
 289             return SND_PCM_FORMAT_U16_BE;
 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         } else {
 298             return SND_PCM_FORMAT_S32_LE;
 299         }
 300
 301     case AUDIO_FORMAT_U32:
 302         if (endianness) {
 303             return SND_PCM_FORMAT_U32_BE;
 304         } else {
 305             return SND_PCM_FORMAT_U32_LE;
 306         }
 307
 308     case AUDIO_FORMAT_F32:
 309         if (endianness) {
 310             return SND_PCM_FORMAT_FLOAT_BE;
 311         } else {
 312             return SND_PCM_FORMAT_FLOAT_LE;
 313         }
 314
 315     default:
 316         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 317 #ifdef DEBUG_AUDIO
 318         abort ();
 319 #endif
 320         return SND_PCM_FORMAT_U8;
 321     }
 322 }
 323
 324 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
 325                            int *endianness)
 326 {
 327     switch (alsafmt) {
 328     case SND_PCM_FORMAT_S8:
 329         *endianness = 0;
 330         *fmt = AUDIO_FORMAT_S8;
 331         break;
 332
 333     case SND_PCM_FORMAT_U8:
 334         *endianness = 0;
 335         *fmt = AUDIO_FORMAT_U8;
 336         break;
 337
 338     case SND_PCM_FORMAT_S16_LE:
 339         *endianness = 0;
 340         *fmt = AUDIO_FORMAT_S16;
 341         break;
 342
 343     case SND_PCM_FORMAT_U16_LE:
 344         *endianness = 0;
 345         *fmt = AUDIO_FORMAT_U16;
 346         break;
 347
 348     case SND_PCM_FORMAT_S16_BE:
 349         *endianness = 1;
 350         *fmt = AUDIO_FORMAT_S16;
 351         break;
 352
 353     case SND_PCM_FORMAT_U16_BE:
 354         *endianness = 1;
 355         *fmt = AUDIO_FORMAT_U16;
 356         break;
 357
 358     case SND_PCM_FORMAT_S32_LE:
 359         *endianness = 0;
 360         *fmt = AUDIO_FORMAT_S32;
 361         break;
 362
 363     case SND_PCM_FORMAT_U32_LE:
 364         *endianness = 0;
 365         *fmt = AUDIO_FORMAT_U32;
 366         break;
 367
 368     case SND_PCM_FORMAT_S32_BE:
 369         *endianness = 1;
 370         *fmt = AUDIO_FORMAT_S32;
 371         break;
 372
 373     case SND_PCM_FORMAT_U32_BE:
 374         *endianness = 1;
 375         *fmt = AUDIO_FORMAT_U32;
 376         break;
 377
 378     case SND_PCM_FORMAT_FLOAT_LE:
 379         *endianness = 0;
 380         *fmt = AUDIO_FORMAT_F32;
 381         break;
 382
 383     case SND_PCM_FORMAT_FLOAT_BE:
 384         *endianness = 1;
 385         *fmt = AUDIO_FORMAT_F32;
 386         break;
 387
 388     default:
 389         dolog ("Unrecognized audio format %d\n", alsafmt);
 390         return -1;
 391     }
 392
 393     return 0;
 394 }
 395
 396 static void alsa_dump_info (struct alsa_params_req *req,
 397                             struct alsa_params_obt *obt,
 398                             snd_pcm_format_t obtfmt,
 399                             AudiodevAlsaPerDirectionOptions *apdo)
 400 {
 401     dolog("parameter | requested value | obtained value\n");
 402     dolog("format    |      %10d |     %10d\n", req->fmt, obtfmt);
 403     dolog("channels  |      %10d |     %10d\n",
 404           req->nchannels, obt->nchannels);
 405     dolog("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 406     dolog("============================================\n");
 407     dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
 408           apdo->buffer_length, apdo->period_length);
 409     dolog("obtained: samples %ld\n", obt->samples);
 410 }
 411
 412 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 413 {
 414     int err;
 415     snd_pcm_sw_params_t *sw_params;
 416
 417     snd_pcm_sw_params_alloca (&sw_params);
 418
 419     err = snd_pcm_sw_params_current (handle, sw_params);
 420     if (err < 0) {
 421         dolog ("Could not fully initialize DAC\n");
 422         alsa_logerr (err, "Failed to get current software parameters\n");
 423         return;
 424     }
 425
 426     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 427     if (err < 0) {
 428         dolog ("Could not fully initialize DAC\n");
 429         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 430                      threshold);
 431         return;
 432     }
 433
 434     err = snd_pcm_sw_params (handle, sw_params);
 435     if (err < 0) {
 436         dolog ("Could not fully initialize DAC\n");
 437         alsa_logerr (err, "Failed to set software parameters\n");
 438         return;
 439     }
 440 }
 441
 442 static int alsa_open(bool in, struct alsa_params_req *req,
 443                      struct alsa_params_obt *obt, snd_pcm_t **handlep,
 444                      Audiodev *dev)
 445 {
 446     AudiodevAlsaOptions *aopts = &dev->u.alsa;
 447     AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
 448     snd_pcm_t *handle;
 449     snd_pcm_hw_params_t *hw_params;
 450     int err;
 451     unsigned int freq, nchannels;
 452     const char *pcm_name = apdo->dev ?: "default";
 453     snd_pcm_uframes_t obt_buffer_size;
 454     const char *typ = in ? "ADC" : "DAC";
 455     snd_pcm_format_t obtfmt;
 456
 457     freq = req->freq;
 458     nchannels = req->nchannels;
 459
 460     snd_pcm_hw_params_alloca (&hw_params);
 461
 462     err = snd_pcm_open (
 463         &handle,
 464         pcm_name,
 465         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 466         SND_PCM_NONBLOCK
 467         );
 468     if (err < 0) {
 469         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 470         return -1;
 471     }
 472
 473     err = snd_pcm_hw_params_any (handle, hw_params);
 474     if (err < 0) {
 475         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 476         goto err;
 477     }
 478
 479     err = snd_pcm_hw_params_set_access (
 480         handle,
 481         hw_params,
 482         SND_PCM_ACCESS_RW_INTERLEAVED
 483         );
 484     if (err < 0) {
 485         alsa_logerr2 (err, typ, "Failed to set access type\n");
 486         goto err;
 487     }
 488
 489     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 490     if (err < 0) {
 491         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 492     }
 493
 494     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 495     if (err < 0) {
 496         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 497         goto err;
 498     }
 499
 500     err = snd_pcm_hw_params_set_channels_near (
 501         handle,
 502         hw_params,
 503         &nchannels
 504         );
 505     if (err < 0) {
 506         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 507                       req->nchannels);
 508         goto err;
 509     }
 510
 511     if (apdo->buffer_length) {
 512         int dir = 0;
 513         unsigned int btime = apdo->buffer_length;
 514
 515         err = snd_pcm_hw_params_set_buffer_time_near(
 516             handle, hw_params, &btime, &dir);
 517
 518         if (err < 0) {
 519             alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
 520                          apdo->buffer_length);
 521             goto err;
 522         }
 523
 524         if (apdo->has_buffer_length && btime != apdo->buffer_length) {
 525             dolog("Requested buffer time %" PRId32
 526                   " was rejected, using %u\n", apdo->buffer_length, btime);
 527         }
 528     }
 529
 530     if (apdo->period_length) {
 531         int dir = 0;
 532         unsigned int ptime = apdo->period_length;
 533
 534         err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
 535                                                      &dir);
 536
 537         if (err < 0) {
 538             alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
 539                          apdo->period_length);
 540             goto err;
 541         }
 542
 543         if (apdo->has_period_length && ptime != apdo->period_length) {
 544             dolog("Requested period time %" PRId32 " was rejected, using %d\n",
 545                   apdo->period_length, ptime);
 546         }
 547     }
 548
 549     err = snd_pcm_hw_params (handle, hw_params);
 550     if (err < 0) {
 551         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 552         goto err;
 553     }
 554
 555     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 556     if (err < 0) {
 557         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 558         goto err;
 559     }
 560
 561     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 562     if (err < 0) {
 563         alsa_logerr2 (err, typ, "Failed to get format\n");
 564         goto err;
 565     }
 566
 567     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 568         dolog ("Invalid format was returned %d\n", obtfmt);
 569         goto err;
 570     }
 571
 572     err = snd_pcm_prepare (handle);
 573     if (err < 0) {
 574         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 575         goto err;
 576     }
 577
 578     if (!in && aopts->has_threshold && aopts->threshold) {
 579         struct audsettings as = { .freq = freq };
 580         alsa_set_threshold(
 581             handle,
 582             audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
 583                                 &as, aopts->threshold));
 584     }
 585
 586     obt->nchannels = nchannels;
 587     obt->freq = freq;
 588     obt->samples = obt_buffer_size;
 589
 590     *handlep = handle;
 591
 592     if (DEBUG_ALSA || obtfmt != req->fmt ||
 593         obt->nchannels != req->nchannels || obt->freq != req->freq) {
 594         dolog ("Audio parameters for %s\n", typ);
 595         alsa_dump_info(req, obt, obtfmt, apdo);
 596     }
 597
 598     return 0;
 599
 600  err:
 601     alsa_anal_close1 (&handle);
 602     return -1;
 603 }
 604
 605 static size_t alsa_buffer_get_free(HWVoiceOut *hw)
 606 {
 607     ALSAVoiceOut *alsa = (ALSAVoiceOut *)hw;
 608     snd_pcm_sframes_t avail;
 609     size_t alsa_free, generic_free, generic_in_use;
 610
 611     avail = snd_pcm_avail_update(alsa->handle);
 612     if (avail < 0) {
 613         if (avail == -EPIPE) {
 614             if (!alsa_recover(alsa->handle)) {
 615                 avail = snd_pcm_avail_update(alsa->handle);
 616             }
 617         }
 618         if (avail < 0) {
 619             alsa_logerr(avail,
 620                         "Could not obtain number of available frames\n");
 621             avail = 0;
 622         }
 623     }
 624
 625     alsa_free = avail * hw->info.bytes_per_frame;
 626     generic_free = audio_generic_buffer_get_free(hw);
 627     generic_in_use = hw->samples * hw->info.bytes_per_frame - generic_free;
 628     if (generic_in_use) {
 629         /*
 630          * This code can only be reached in the unlikely case that
 631          * snd_pcm_avail_update() returned a larger number of frames
 632          * than snd_pcm_writei() could write. Make sure that all
 633          * remaining bytes in the generic buffer can be written.
 634          */
 635         alsa_free = alsa_free > generic_in_use ? alsa_free - generic_in_use : 0;
 636     }
 637
 638     return alsa_free;
 639 }
 640
 641 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
 642 {
 643     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 644     size_t pos = 0;
 645     size_t len_frames = len / hw->info.bytes_per_frame;
 646
 647     while (len_frames) {
 648         char *src = advance(buf, pos);
 649         snd_pcm_sframes_t written;
 650
 651         written = snd_pcm_writei(alsa->handle, src, len_frames);
 652
 653         if (written <= 0) {
 654             switch (written) {
 655             case 0:
 656                 trace_alsa_wrote_zero(len_frames);
 657                 return pos;
 658
 659             case -EPIPE:
 660                 if (alsa_recover(alsa->handle)) {
 661                     alsa_logerr(written, "Failed to write %zu frames\n",
 662                                 len_frames);
 663                     return pos;
 664                 }
 665                 trace_alsa_xrun_out();
 666                 continue;
 667
 668             case -ESTRPIPE:
 669                 /*
 670                  * stream is suspended and waiting for an application
 671                  * recovery
 672                  */
 673                 if (alsa_resume(alsa->handle)) {
 674                     alsa_logerr(written, "Failed to write %zu frames\n",
 675                                 len_frames);
 676                     return pos;
 677                 }
 678                 trace_alsa_resume_out();
 679                 continue;
 680
 681             case -EAGAIN:
 682                 return pos;
 683
 684             default:
 685                 alsa_logerr(written, "Failed to write %zu frames from %p\n",
 686                             len, src);
 687                 return pos;
 688             }
 689         }
 690
 691         pos += written * hw->info.bytes_per_frame;
 692         if (written < len_frames) {
 693             break;
 694         }
 695         len_frames -= written;
 696     }
 697
 698     return pos;
 699 }
 700
 701 static void alsa_fini_out (HWVoiceOut *hw)
 702 {
 703     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 704
 705     ldebug ("alsa_fini\n");
 706     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 707 }
 708
 709 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
 710                          void *drv_opaque)
 711 {
 712     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 713     struct alsa_params_req req;
 714     struct alsa_params_obt obt;
 715     snd_pcm_t *handle;
 716     struct audsettings obt_as;
 717     Audiodev *dev = drv_opaque;
 718
 719     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 720     req.freq = as->freq;
 721     req.nchannels = as->nchannels;
 722
 723     if (alsa_open(0, &req, &obt, &handle, dev)) {
 724         return -1;
 725     }
 726
 727     obt_as.freq = obt.freq;
 728     obt_as.nchannels = obt.nchannels;
 729     obt_as.fmt = obt.fmt;
 730     obt_as.endianness = obt.endianness;
 731
 732     audio_pcm_init_info (&hw->info, &obt_as);
 733     hw->samples = obt.samples;
 734
 735     alsa->pollhlp.s = hw->s;
 736     alsa->handle = handle;
 737     alsa->dev = dev;
 738     return 0;
 739 }
 740
 741 #define VOICE_CTL_PAUSE 0
 742 #define VOICE_CTL_PREPARE 1
 743 #define VOICE_CTL_START 2
 744
 745 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
 746 {
 747     int err;
 748
 749     if (ctl == VOICE_CTL_PAUSE) {
 750         err = snd_pcm_drop (handle);
 751         if (err < 0) {
 752             alsa_logerr (err, "Could not stop %s\n", typ);
 753             return -1;
 754         }
 755     } else {
 756         err = snd_pcm_prepare (handle);
 757         if (err < 0) {
 758             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 759             return -1;
 760         }
 761         if (ctl == VOICE_CTL_START) {
 762             err = snd_pcm_start(handle);
 763             if (err < 0) {
 764                 alsa_logerr (err, "Could not start handle for %s\n", typ);
 765                 return -1;
 766             }
 767         }
 768     }
 769
 770     return 0;
 771 }
 772
 773 static void alsa_enable_out(HWVoiceOut *hw, bool enable)
 774 {
 775     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 776     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
 777
 778     if (enable) {
 779         bool poll_mode = apdo->try_poll;
 780
 781         ldebug("enabling voice\n");
 782         if (poll_mode && alsa_poll_out(hw)) {
 783             poll_mode = 0;
 784         }
 785         hw->poll_mode = poll_mode;
 786         alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
 787     } else {
 788         ldebug("disabling voice\n");
 789         if (hw->poll_mode) {
 790             hw->poll_mode = 0;
 791             alsa_fini_poll(&alsa->pollhlp);
 792         }
 793         alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
 794     }
 795 }
 796
 797 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 798 {
 799     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 800     struct alsa_params_req req;
 801     struct alsa_params_obt obt;
 802     snd_pcm_t *handle;
 803     struct audsettings obt_as;
 804     Audiodev *dev = drv_opaque;
 805
 806     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 807     req.freq = as->freq;
 808     req.nchannels = as->nchannels;
 809
 810     if (alsa_open(1, &req, &obt, &handle, dev)) {
 811         return -1;
 812     }
 813
 814     obt_as.freq = obt.freq;
 815     obt_as.nchannels = obt.nchannels;
 816     obt_as.fmt = obt.fmt;
 817     obt_as.endianness = obt.endianness;
 818
 819     audio_pcm_init_info (&hw->info, &obt_as);
 820     hw->samples = obt.samples;
 821
 822     alsa->pollhlp.s = hw->s;
 823     alsa->handle = handle;
 824     alsa->dev = dev;
 825     return 0;
 826 }
 827
 828 static void alsa_fini_in (HWVoiceIn *hw)
 829 {
 830     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 831
 832     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 833 }
 834
 835 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
 836 {
 837     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 838     size_t pos = 0;
 839
 840     while (len) {
 841         void *dst = advance(buf, pos);
 842         snd_pcm_sframes_t nread;
 843
 844         nread = snd_pcm_readi(
 845             alsa->handle, dst, len / hw->info.bytes_per_frame);
 846
 847         if (nread <= 0) {
 848             switch (nread) {
 849             case 0:
 850                 trace_alsa_read_zero(len);
 851                 return pos;
 852
 853             case -EPIPE:
 854                 if (alsa_recover(alsa->handle)) {
 855                     alsa_logerr(nread, "Failed to read %zu frames\n", len);
 856                     return pos;
 857                 }
 858                 trace_alsa_xrun_in();
 859                 continue;
 860
 861             case -EAGAIN:
 862                 return pos;
 863
 864             default:
 865                 alsa_logerr(nread, "Failed to read %zu frames to %p\n",
 866                             len, dst);
 867                 return pos;
 868             }
 869         }
 870
 871         pos += nread * hw->info.bytes_per_frame;
 872         len -= nread * hw->info.bytes_per_frame;
 873     }
 874
 875     return pos;
 876 }
 877
 878 static void alsa_enable_in(HWVoiceIn *hw, bool enable)
 879 {
 880     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 881     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
 882
 883     if (enable) {
 884         bool poll_mode = apdo->try_poll;
 885
 886         ldebug("enabling voice\n");
 887         if (poll_mode && alsa_poll_in(hw)) {
 888             poll_mode = 0;
 889         }
 890         hw->poll_mode = poll_mode;
 891
 892         alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
 893     } else {
 894         ldebug ("disabling voice\n");
 895         if (hw->poll_mode) {
 896             hw->poll_mode = 0;
 897             alsa_fini_poll(&alsa->pollhlp);
 898         }
 899         alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
 900     }
 901 }
 902
 903 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
 904 {
 905     if (!apdo->has_try_poll) {
 906         apdo->try_poll = true;
 907         apdo->has_try_poll = true;
 908     }
 909 }
 910
 911 static void *alsa_audio_init(Audiodev *dev)
 912 {
 913     AudiodevAlsaOptions *aopts;
 914     assert(dev->driver == AUDIODEV_DRIVER_ALSA);
 915
 916     aopts = &dev->u.alsa;
 917     alsa_init_per_direction(aopts->in);
 918     alsa_init_per_direction(aopts->out);
 919
 920     /*
 921      * need to define them, as otherwise alsa produces no sound
 922      * doesn't set has_* so alsa_open can identify it wasn't set by the user
 923      */
 924     if (!dev->u.alsa.out->has_period_length) {
 925         /* 1024 frames assuming 44100Hz */
 926         dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
 927     }
 928     if (!dev->u.alsa.out->has_buffer_length) {
 929         /* 4096 frames assuming 44100Hz */
 930         dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
 931     }
 932
 933     /*
 934      * OptsVisitor sets unspecified optional fields to zero, but do not depend
 935      * on it...
 936      */
 937     if (!dev->u.alsa.in->has_period_length) {
 938         dev->u.alsa.in->period_length = 0;
 939     }
 940     if (!dev->u.alsa.in->has_buffer_length) {
 941         dev->u.alsa.in->buffer_length = 0;
 942     }
 943
 944     return dev;
 945 }
 946
 947 static void alsa_audio_fini (void *opaque)
 948 {
 949 }
 950
 951 static struct audio_pcm_ops alsa_pcm_ops = {
 952     .init_out = alsa_init_out,
 953     .fini_out = alsa_fini_out,
 954     .write    = alsa_write,
 955     .buffer_get_free = alsa_buffer_get_free,
 956     .run_buffer_out = audio_generic_run_buffer_out,
 957     .enable_out = alsa_enable_out,
 958
 959     .init_in  = alsa_init_in,
 960     .fini_in  = alsa_fini_in,
 961     .read     = alsa_read,
 962     .run_buffer_in = audio_generic_run_buffer_in,
 963     .enable_in = alsa_enable_in,
 964 };
 965
 966 static struct audio_driver alsa_audio_driver = {
 967     .name           = "alsa",
 968     .descr          = "ALSA http://www.alsa-project.org",
 969     .init           = alsa_audio_init,
 970     .fini           = alsa_audio_fini,
 971     .pcm_ops        = &alsa_pcm_ops,
 972     .can_be_default = 1,
 973     .max_voices_out = INT_MAX,
 974     .max_voices_in  = INT_MAX,
 975     .voice_size_out = sizeof (ALSAVoiceOut),
 976     .voice_size_in  = sizeof (ALSAVoiceIn)
 977 };
 978
 979 static void register_audio_alsa(void)
 980 {
 981     audio_driver_register(&alsa_audio_driver);
 982 }
 983 type_init(register_audio_alsa);