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 = g_new0(struct pollfd, count);
 226
 227     err = snd_pcm_poll_descriptors (handle, pfds, count);
 228     if (err < 0) {
 229         alsa_logerr (err, "Could not initialize poll mode\n"
 230                      "Could not obtain poll descriptors\n");
 231         g_free (pfds);
 232         return -1;
 233     }
 234
 235     for (i = 0; i < count; ++i) {
 236         if (pfds[i].events & POLLIN) {
 237             qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler, NULL, hlp);
 238         }
 239         if (pfds[i].events & POLLOUT) {
 240             trace_alsa_pollout(i, pfds[i].fd);
 241             qemu_set_fd_handler (pfds[i].fd, NULL, alsa_poll_handler, hlp);
 242         }
 243         trace_alsa_set_handler(pfds[i].events, i, pfds[i].fd, err);
 244
 245     }
 246     hlp->pfds = pfds;
 247     hlp->count = count;
 248     hlp->handle = handle;
 249     hlp->mask = mask;
 250     return 0;
 251 }
 252
 253 static int alsa_poll_out (HWVoiceOut *hw)
 254 {
 255     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 256
 257     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
 258 }
 259
 260 static int alsa_poll_in (HWVoiceIn *hw)
 261 {
 262     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 263
 264     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
 265 }
 266
 267 static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
 268 {
 269     switch (fmt) {
 270     case AUDIO_FORMAT_S8:
 271         return SND_PCM_FORMAT_S8;
 272
 273     case AUDIO_FORMAT_U8:
 274         return SND_PCM_FORMAT_U8;
 275
 276     case AUDIO_FORMAT_S16:
 277         if (endianness) {
 278             return SND_PCM_FORMAT_S16_BE;
 279         } else {
 280             return SND_PCM_FORMAT_S16_LE;
 281         }
 282
 283     case AUDIO_FORMAT_U16:
 284         if (endianness) {
 285             return SND_PCM_FORMAT_U16_BE;
 286         } else {
 287             return SND_PCM_FORMAT_U16_LE;
 288         }
 289
 290     case AUDIO_FORMAT_S32:
 291         if (endianness) {
 292             return SND_PCM_FORMAT_S32_BE;
 293         } else {
 294             return SND_PCM_FORMAT_S32_LE;
 295         }
 296
 297     case AUDIO_FORMAT_U32:
 298         if (endianness) {
 299             return SND_PCM_FORMAT_U32_BE;
 300         } else {
 301             return SND_PCM_FORMAT_U32_LE;
 302         }
 303
 304     case AUDIO_FORMAT_F32:
 305         if (endianness) {
 306             return SND_PCM_FORMAT_FLOAT_BE;
 307         } else {
 308             return SND_PCM_FORMAT_FLOAT_LE;
 309         }
 310
 311     default:
 312         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 313 #ifdef DEBUG_AUDIO
 314         abort ();
 315 #endif
 316         return SND_PCM_FORMAT_U8;
 317     }
 318 }
 319
 320 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
 321                            int *endianness)
 322 {
 323     switch (alsafmt) {
 324     case SND_PCM_FORMAT_S8:
 325         *endianness = 0;
 326         *fmt = AUDIO_FORMAT_S8;
 327         break;
 328
 329     case SND_PCM_FORMAT_U8:
 330         *endianness = 0;
 331         *fmt = AUDIO_FORMAT_U8;
 332         break;
 333
 334     case SND_PCM_FORMAT_S16_LE:
 335         *endianness = 0;
 336         *fmt = AUDIO_FORMAT_S16;
 337         break;
 338
 339     case SND_PCM_FORMAT_U16_LE:
 340         *endianness = 0;
 341         *fmt = AUDIO_FORMAT_U16;
 342         break;
 343
 344     case SND_PCM_FORMAT_S16_BE:
 345         *endianness = 1;
 346         *fmt = AUDIO_FORMAT_S16;
 347         break;
 348
 349     case SND_PCM_FORMAT_U16_BE:
 350         *endianness = 1;
 351         *fmt = AUDIO_FORMAT_U16;
 352         break;
 353
 354     case SND_PCM_FORMAT_S32_LE:
 355         *endianness = 0;
 356         *fmt = AUDIO_FORMAT_S32;
 357         break;
 358
 359     case SND_PCM_FORMAT_U32_LE:
 360         *endianness = 0;
 361         *fmt = AUDIO_FORMAT_U32;
 362         break;
 363
 364     case SND_PCM_FORMAT_S32_BE:
 365         *endianness = 1;
 366         *fmt = AUDIO_FORMAT_S32;
 367         break;
 368
 369     case SND_PCM_FORMAT_U32_BE:
 370         *endianness = 1;
 371         *fmt = AUDIO_FORMAT_U32;
 372         break;
 373
 374     case SND_PCM_FORMAT_FLOAT_LE:
 375         *endianness = 0;
 376         *fmt = AUDIO_FORMAT_F32;
 377         break;
 378
 379     case SND_PCM_FORMAT_FLOAT_BE:
 380         *endianness = 1;
 381         *fmt = AUDIO_FORMAT_F32;
 382         break;
 383
 384     default:
 385         dolog ("Unrecognized audio format %d\n", alsafmt);
 386         return -1;
 387     }
 388
 389     return 0;
 390 }
 391
 392 static void alsa_dump_info (struct alsa_params_req *req,
 393                             struct alsa_params_obt *obt,
 394                             snd_pcm_format_t obtfmt,
 395                             AudiodevAlsaPerDirectionOptions *apdo)
 396 {
 397     dolog("parameter | requested value | obtained value\n");
 398     dolog("format    |      %10d |     %10d\n", req->fmt, obtfmt);
 399     dolog("channels  |      %10d |     %10d\n",
 400           req->nchannels, obt->nchannels);
 401     dolog("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 402     dolog("============================================\n");
 403     dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
 404           apdo->buffer_length, apdo->period_length);
 405     dolog("obtained: samples %ld\n", obt->samples);
 406 }
 407
 408 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 409 {
 410     int err;
 411     snd_pcm_sw_params_t *sw_params;
 412
 413     snd_pcm_sw_params_alloca (&sw_params);
 414
 415     err = snd_pcm_sw_params_current (handle, sw_params);
 416     if (err < 0) {
 417         dolog ("Could not fully initialize DAC\n");
 418         alsa_logerr (err, "Failed to get current software parameters\n");
 419         return;
 420     }
 421
 422     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 423     if (err < 0) {
 424         dolog ("Could not fully initialize DAC\n");
 425         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 426                      threshold);
 427         return;
 428     }
 429
 430     err = snd_pcm_sw_params (handle, sw_params);
 431     if (err < 0) {
 432         dolog ("Could not fully initialize DAC\n");
 433         alsa_logerr (err, "Failed to set software parameters\n");
 434         return;
 435     }
 436 }
 437
 438 static int alsa_open(bool in, struct alsa_params_req *req,
 439                      struct alsa_params_obt *obt, snd_pcm_t **handlep,
 440                      Audiodev *dev)
 441 {
 442     AudiodevAlsaOptions *aopts = &dev->u.alsa;
 443     AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
 444     snd_pcm_t *handle;
 445     snd_pcm_hw_params_t *hw_params;
 446     int err;
 447     unsigned int freq, nchannels;
 448     const char *pcm_name = apdo->dev ?: "default";
 449     snd_pcm_uframes_t obt_buffer_size;
 450     const char *typ = in ? "ADC" : "DAC";
 451     snd_pcm_format_t obtfmt;
 452
 453     freq = req->freq;
 454     nchannels = req->nchannels;
 455
 456     snd_pcm_hw_params_alloca (&hw_params);
 457
 458     err = snd_pcm_open (
 459         &handle,
 460         pcm_name,
 461         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 462         SND_PCM_NONBLOCK
 463         );
 464     if (err < 0) {
 465         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 466         return -1;
 467     }
 468
 469     err = snd_pcm_hw_params_any (handle, hw_params);
 470     if (err < 0) {
 471         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 472         goto err;
 473     }
 474
 475     err = snd_pcm_hw_params_set_access (
 476         handle,
 477         hw_params,
 478         SND_PCM_ACCESS_RW_INTERLEAVED
 479         );
 480     if (err < 0) {
 481         alsa_logerr2 (err, typ, "Failed to set access type\n");
 482         goto err;
 483     }
 484
 485     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 486     if (err < 0) {
 487         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 488     }
 489
 490     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 491     if (err < 0) {
 492         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 493         goto err;
 494     }
 495
 496     err = snd_pcm_hw_params_set_channels_near (
 497         handle,
 498         hw_params,
 499         &nchannels
 500         );
 501     if (err < 0) {
 502         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 503                       req->nchannels);
 504         goto err;
 505     }
 506
 507     if (apdo->buffer_length) {
 508         int dir = 0;
 509         unsigned int btime = apdo->buffer_length;
 510
 511         err = snd_pcm_hw_params_set_buffer_time_near(
 512             handle, hw_params, &btime, &dir);
 513
 514         if (err < 0) {
 515             alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
 516                          apdo->buffer_length);
 517             goto err;
 518         }
 519
 520         if (apdo->has_buffer_length && btime != apdo->buffer_length) {
 521             dolog("Requested buffer time %" PRId32
 522                   " was rejected, using %u\n", apdo->buffer_length, btime);
 523         }
 524     }
 525
 526     if (apdo->period_length) {
 527         int dir = 0;
 528         unsigned int ptime = apdo->period_length;
 529
 530         err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
 531                                                      &dir);
 532
 533         if (err < 0) {
 534             alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
 535                          apdo->period_length);
 536             goto err;
 537         }
 538
 539         if (apdo->has_period_length && ptime != apdo->period_length) {
 540             dolog("Requested period time %" PRId32 " was rejected, using %d\n",
 541                   apdo->period_length, ptime);
 542         }
 543     }
 544
 545     err = snd_pcm_hw_params (handle, hw_params);
 546     if (err < 0) {
 547         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 548         goto err;
 549     }
 550
 551     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 552     if (err < 0) {
 553         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 554         goto err;
 555     }
 556
 557     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 558     if (err < 0) {
 559         alsa_logerr2 (err, typ, "Failed to get format\n");
 560         goto err;
 561     }
 562
 563     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 564         dolog ("Invalid format was returned %d\n", obtfmt);
 565         goto err;
 566     }
 567
 568     err = snd_pcm_prepare (handle);
 569     if (err < 0) {
 570         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 571         goto err;
 572     }
 573
 574     if (!in && aopts->has_threshold && aopts->threshold) {
 575         struct audsettings as = { .freq = freq };
 576         alsa_set_threshold(
 577             handle,
 578             audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
 579                                 &as, aopts->threshold));
 580     }
 581
 582     obt->nchannels = nchannels;
 583     obt->freq = freq;
 584     obt->samples = obt_buffer_size;
 585
 586     *handlep = handle;
 587
 588     if (DEBUG_ALSA || obtfmt != req->fmt ||
 589         obt->nchannels != req->nchannels || obt->freq != req->freq) {
 590         dolog ("Audio parameters for %s\n", typ);
 591         alsa_dump_info(req, obt, obtfmt, apdo);
 592     }
 593
 594     return 0;
 595
 596  err:
 597     alsa_anal_close1 (&handle);
 598     return -1;
 599 }
 600
 601 static size_t alsa_buffer_get_free(HWVoiceOut *hw)
 602 {
 603     ALSAVoiceOut *alsa = (ALSAVoiceOut *)hw;
 604     snd_pcm_sframes_t avail;
 605     size_t alsa_free, generic_free, generic_in_use;
 606
 607     avail = snd_pcm_avail_update(alsa->handle);
 608     if (avail < 0) {
 609         if (avail == -EPIPE) {
 610             if (!alsa_recover(alsa->handle)) {
 611                 avail = snd_pcm_avail_update(alsa->handle);
 612             }
 613         }
 614         if (avail < 0) {
 615             alsa_logerr(avail,
 616                         "Could not obtain number of available frames\n");
 617             avail = 0;
 618         }
 619     }
 620
 621     alsa_free = avail * hw->info.bytes_per_frame;
 622     generic_free = audio_generic_buffer_get_free(hw);
 623     generic_in_use = hw->samples * hw->info.bytes_per_frame - generic_free;
 624     if (generic_in_use) {
 625         /*
 626          * This code can only be reached in the unlikely case that
 627          * snd_pcm_avail_update() returned a larger number of frames
 628          * than snd_pcm_writei() could write. Make sure that all
 629          * remaining bytes in the generic buffer can be written.
 630          */
 631         alsa_free = alsa_free > generic_in_use ? alsa_free - generic_in_use : 0;
 632     }
 633
 634     return alsa_free;
 635 }
 636
 637 static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len)
 638 {
 639     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 640     size_t pos = 0;
 641     size_t len_frames = len / hw->info.bytes_per_frame;
 642
 643     while (len_frames) {
 644         char *src = advance(buf, pos);
 645         snd_pcm_sframes_t written;
 646
 647         written = snd_pcm_writei(alsa->handle, src, len_frames);
 648
 649         if (written <= 0) {
 650             switch (written) {
 651             case 0:
 652                 trace_alsa_wrote_zero(len_frames);
 653                 return pos;
 654
 655             case -EPIPE:
 656                 if (alsa_recover(alsa->handle)) {
 657                     alsa_logerr(written, "Failed to write %zu frames\n",
 658                                 len_frames);
 659                     return pos;
 660                 }
 661                 trace_alsa_xrun_out();
 662                 continue;
 663
 664             case -ESTRPIPE:
 665                 /*
 666                  * stream is suspended and waiting for an application
 667                  * recovery
 668                  */
 669                 if (alsa_resume(alsa->handle)) {
 670                     alsa_logerr(written, "Failed to write %zu frames\n",
 671                                 len_frames);
 672                     return pos;
 673                 }
 674                 trace_alsa_resume_out();
 675                 continue;
 676
 677             case -EAGAIN:
 678                 return pos;
 679
 680             default:
 681                 alsa_logerr(written, "Failed to write %zu frames from %p\n",
 682                             len, src);
 683                 return pos;
 684             }
 685         }
 686
 687         pos += written * hw->info.bytes_per_frame;
 688         if (written < len_frames) {
 689             break;
 690         }
 691         len_frames -= written;
 692     }
 693
 694     return pos;
 695 }
 696
 697 static void alsa_fini_out (HWVoiceOut *hw)
 698 {
 699     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 700
 701     ldebug ("alsa_fini\n");
 702     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 703 }
 704
 705 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
 706                          void *drv_opaque)
 707 {
 708     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 709     struct alsa_params_req req;
 710     struct alsa_params_obt obt;
 711     snd_pcm_t *handle;
 712     struct audsettings obt_as;
 713     Audiodev *dev = drv_opaque;
 714
 715     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 716     req.freq = as->freq;
 717     req.nchannels = as->nchannels;
 718
 719     if (alsa_open(0, &req, &obt, &handle, dev)) {
 720         return -1;
 721     }
 722
 723     obt_as.freq = obt.freq;
 724     obt_as.nchannels = obt.nchannels;
 725     obt_as.fmt = obt.fmt;
 726     obt_as.endianness = obt.endianness;
 727
 728     audio_pcm_init_info (&hw->info, &obt_as);
 729     hw->samples = obt.samples;
 730
 731     alsa->pollhlp.s = hw->s;
 732     alsa->handle = handle;
 733     alsa->dev = dev;
 734     return 0;
 735 }
 736
 737 #define VOICE_CTL_PAUSE 0
 738 #define VOICE_CTL_PREPARE 1
 739 #define VOICE_CTL_START 2
 740
 741 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
 742 {
 743     int err;
 744
 745     if (ctl == VOICE_CTL_PAUSE) {
 746         err = snd_pcm_drop (handle);
 747         if (err < 0) {
 748             alsa_logerr (err, "Could not stop %s\n", typ);
 749             return -1;
 750         }
 751     } else {
 752         err = snd_pcm_prepare (handle);
 753         if (err < 0) {
 754             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 755             return -1;
 756         }
 757         if (ctl == VOICE_CTL_START) {
 758             err = snd_pcm_start(handle);
 759             if (err < 0) {
 760                 alsa_logerr (err, "Could not start handle for %s\n", typ);
 761                 return -1;
 762             }
 763         }
 764     }
 765
 766     return 0;
 767 }
 768
 769 static void alsa_enable_out(HWVoiceOut *hw, bool enable)
 770 {
 771     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 772     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
 773
 774     if (enable) {
 775         bool poll_mode = apdo->try_poll;
 776
 777         ldebug("enabling voice\n");
 778         if (poll_mode && alsa_poll_out(hw)) {
 779             poll_mode = 0;
 780         }
 781         hw->poll_mode = poll_mode;
 782         alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PREPARE);
 783     } else {
 784         ldebug("disabling voice\n");
 785         if (hw->poll_mode) {
 786             hw->poll_mode = 0;
 787             alsa_fini_poll(&alsa->pollhlp);
 788         }
 789         alsa_voice_ctl(alsa->handle, "playback", VOICE_CTL_PAUSE);
 790     }
 791 }
 792
 793 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 794 {
 795     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 796     struct alsa_params_req req;
 797     struct alsa_params_obt obt;
 798     snd_pcm_t *handle;
 799     struct audsettings obt_as;
 800     Audiodev *dev = drv_opaque;
 801
 802     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 803     req.freq = as->freq;
 804     req.nchannels = as->nchannels;
 805
 806     if (alsa_open(1, &req, &obt, &handle, dev)) {
 807         return -1;
 808     }
 809
 810     obt_as.freq = obt.freq;
 811     obt_as.nchannels = obt.nchannels;
 812     obt_as.fmt = obt.fmt;
 813     obt_as.endianness = obt.endianness;
 814
 815     audio_pcm_init_info (&hw->info, &obt_as);
 816     hw->samples = obt.samples;
 817
 818     alsa->pollhlp.s = hw->s;
 819     alsa->handle = handle;
 820     alsa->dev = dev;
 821     return 0;
 822 }
 823
 824 static void alsa_fini_in (HWVoiceIn *hw)
 825 {
 826     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 827
 828     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 829 }
 830
 831 static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len)
 832 {
 833     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 834     size_t pos = 0;
 835
 836     while (len) {
 837         void *dst = advance(buf, pos);
 838         snd_pcm_sframes_t nread;
 839
 840         nread = snd_pcm_readi(
 841             alsa->handle, dst, len / hw->info.bytes_per_frame);
 842
 843         if (nread <= 0) {
 844             switch (nread) {
 845             case 0:
 846                 trace_alsa_read_zero(len);
 847                 return pos;
 848
 849             case -EPIPE:
 850                 if (alsa_recover(alsa->handle)) {
 851                     alsa_logerr(nread, "Failed to read %zu frames\n", len);
 852                     return pos;
 853                 }
 854                 trace_alsa_xrun_in();
 855                 continue;
 856
 857             case -EAGAIN:
 858                 return pos;
 859
 860             default:
 861                 alsa_logerr(nread, "Failed to read %zu frames to %p\n",
 862                             len, dst);
 863                 return pos;
 864             }
 865         }
 866
 867         pos += nread * hw->info.bytes_per_frame;
 868         len -= nread * hw->info.bytes_per_frame;
 869     }
 870
 871     return pos;
 872 }
 873
 874 static void alsa_enable_in(HWVoiceIn *hw, bool enable)
 875 {
 876     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 877     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
 878
 879     if (enable) {
 880         bool poll_mode = apdo->try_poll;
 881
 882         ldebug("enabling voice\n");
 883         if (poll_mode && alsa_poll_in(hw)) {
 884             poll_mode = 0;
 885         }
 886         hw->poll_mode = poll_mode;
 887
 888         alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_START);
 889     } else {
 890         ldebug ("disabling voice\n");
 891         if (hw->poll_mode) {
 892             hw->poll_mode = 0;
 893             alsa_fini_poll(&alsa->pollhlp);
 894         }
 895         alsa_voice_ctl(alsa->handle, "capture", VOICE_CTL_PAUSE);
 896     }
 897 }
 898
 899 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
 900 {
 901     if (!apdo->has_try_poll) {
 902         apdo->try_poll = true;
 903         apdo->has_try_poll = true;
 904     }
 905 }
 906
 907 static void *alsa_audio_init(Audiodev *dev)
 908 {
 909     AudiodevAlsaOptions *aopts;
 910     assert(dev->driver == AUDIODEV_DRIVER_ALSA);
 911
 912     aopts = &dev->u.alsa;
 913     alsa_init_per_direction(aopts->in);
 914     alsa_init_per_direction(aopts->out);
 915
 916     /* don't set has_* so alsa_open can identify it wasn't set by the user */
 917     if (!dev->u.alsa.out->has_period_length) {
 918         /* 256 frames assuming 44100Hz */
 919         dev->u.alsa.out->period_length = 5805;
 920     }
 921     if (!dev->u.alsa.out->has_buffer_length) {
 922         /* 4096 frames assuming 44100Hz */
 923         dev->u.alsa.out->buffer_length = 92880;
 924     }
 925
 926     if (!dev->u.alsa.in->has_period_length) {
 927         /* 256 frames assuming 44100Hz */
 928         dev->u.alsa.in->period_length = 5805;
 929     }
 930     if (!dev->u.alsa.in->has_buffer_length) {
 931         /* 4096 frames assuming 44100Hz */
 932         dev->u.alsa.in->buffer_length = 92880;
 933     }
 934
 935     return dev;
 936 }
 937
 938 static void alsa_audio_fini (void *opaque)
 939 {
 940 }
 941
 942 static struct audio_pcm_ops alsa_pcm_ops = {
 943     .init_out = alsa_init_out,
 944     .fini_out = alsa_fini_out,
 945     .write    = alsa_write,
 946     .buffer_get_free = alsa_buffer_get_free,
 947     .run_buffer_out = audio_generic_run_buffer_out,
 948     .enable_out = alsa_enable_out,
 949
 950     .init_in  = alsa_init_in,
 951     .fini_in  = alsa_fini_in,
 952     .read     = alsa_read,
 953     .run_buffer_in = audio_generic_run_buffer_in,
 954     .enable_in = alsa_enable_in,
 955 };
 956
 957 static struct audio_driver alsa_audio_driver = {
 958     .name           = "alsa",
 959     .descr          = "ALSA http://www.alsa-project.org",
 960     .init           = alsa_audio_init,
 961     .fini           = alsa_audio_fini,
 962     .pcm_ops        = &alsa_pcm_ops,
 963     .can_be_default = 1,
 964     .max_voices_out = INT_MAX,
 965     .max_voices_in  = INT_MAX,
 966     .voice_size_out = sizeof (ALSAVoiceOut),
 967     .voice_size_in  = sizeof (ALSAVoiceIn)
 968 };
 969
 970 static void register_audio_alsa(void)
 971 {
 972     audio_driver_register(&alsa_audio_driver);
 973 }
 974 type_init(register_audio_alsa);