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 #include "qemu/osdep.h"
  25 #include <alsa/asoundlib.h>
  26 #include "qemu-common.h"
  27 #include "qemu/main-loop.h"
  28 #include "audio.h"
  29 #include "trace.h"
  30
  31 #pragma GCC diagnostic ignored "-Waddress"
  32
  33 #define AUDIO_CAP "alsa"
  34 #include "audio_int.h"
  35
  36 struct pollhlp {
  37     snd_pcm_t *handle;
  38     struct pollfd *pfds;
  39     int count;
  40     int mask;
  41 };
  42
  43 typedef struct ALSAVoiceOut {
  44     HWVoiceOut hw;
  45     int wpos;
  46     int pending;
  47     void *pcm_buf;
  48     snd_pcm_t *handle;
  49     struct pollhlp pollhlp;
  50     Audiodev *dev;
  51 } ALSAVoiceOut;
  52
  53 typedef struct ALSAVoiceIn {
  54     HWVoiceIn hw;
  55     snd_pcm_t *handle;
  56     void *pcm_buf;
  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 GCC_FMT_ATTR (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 GCC_FMT_ATTR (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 ("alsa run (prepared)");
 202         break;
 203
 204     case SND_PCM_STATE_RUNNING:
 205         audio_run ("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 int alsa_write (SWVoiceOut *sw, void *buf, int len)
 272 {
 273     return audio_pcm_sw_write (sw, buf, len);
 274 }
 275
 276 static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness)
 277 {
 278     switch (fmt) {
 279     case AUDIO_FORMAT_S8:
 280         return SND_PCM_FORMAT_S8;
 281
 282     case AUDIO_FORMAT_U8:
 283         return SND_PCM_FORMAT_U8;
 284
 285     case AUDIO_FORMAT_S16:
 286         if (endianness) {
 287             return SND_PCM_FORMAT_S16_BE;
 288         }
 289         else {
 290             return SND_PCM_FORMAT_S16_LE;
 291         }
 292
 293     case AUDIO_FORMAT_U16:
 294         if (endianness) {
 295             return SND_PCM_FORMAT_U16_BE;
 296         }
 297         else {
 298             return SND_PCM_FORMAT_U16_LE;
 299         }
 300
 301     case AUDIO_FORMAT_S32:
 302         if (endianness) {
 303             return SND_PCM_FORMAT_S32_BE;
 304         }
 305         else {
 306             return SND_PCM_FORMAT_S32_LE;
 307         }
 308
 309     case AUDIO_FORMAT_U32:
 310         if (endianness) {
 311             return SND_PCM_FORMAT_U32_BE;
 312         }
 313         else {
 314             return SND_PCM_FORMAT_U32_LE;
 315         }
 316
 317     default:
 318         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 319 #ifdef DEBUG_AUDIO
 320         abort ();
 321 #endif
 322         return SND_PCM_FORMAT_U8;
 323     }
 324 }
 325
 326 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt,
 327                            int *endianness)
 328 {
 329     switch (alsafmt) {
 330     case SND_PCM_FORMAT_S8:
 331         *endianness = 0;
 332         *fmt = AUDIO_FORMAT_S8;
 333         break;
 334
 335     case SND_PCM_FORMAT_U8:
 336         *endianness = 0;
 337         *fmt = AUDIO_FORMAT_U8;
 338         break;
 339
 340     case SND_PCM_FORMAT_S16_LE:
 341         *endianness = 0;
 342         *fmt = AUDIO_FORMAT_S16;
 343         break;
 344
 345     case SND_PCM_FORMAT_U16_LE:
 346         *endianness = 0;
 347         *fmt = AUDIO_FORMAT_U16;
 348         break;
 349
 350     case SND_PCM_FORMAT_S16_BE:
 351         *endianness = 1;
 352         *fmt = AUDIO_FORMAT_S16;
 353         break;
 354
 355     case SND_PCM_FORMAT_U16_BE:
 356         *endianness = 1;
 357         *fmt = AUDIO_FORMAT_U16;
 358         break;
 359
 360     case SND_PCM_FORMAT_S32_LE:
 361         *endianness = 0;
 362         *fmt = AUDIO_FORMAT_S32;
 363         break;
 364
 365     case SND_PCM_FORMAT_U32_LE:
 366         *endianness = 0;
 367         *fmt = AUDIO_FORMAT_U32;
 368         break;
 369
 370     case SND_PCM_FORMAT_S32_BE:
 371         *endianness = 1;
 372         *fmt = AUDIO_FORMAT_S32;
 373         break;
 374
 375     case SND_PCM_FORMAT_U32_BE:
 376         *endianness = 1;
 377         *fmt = AUDIO_FORMAT_U32;
 378         break;
 379
 380     default:
 381         dolog ("Unrecognized audio format %d\n", alsafmt);
 382         return -1;
 383     }
 384
 385     return 0;
 386 }
 387
 388 static void alsa_dump_info (struct alsa_params_req *req,
 389                             struct alsa_params_obt *obt,
 390                             snd_pcm_format_t obtfmt,
 391                             AudiodevAlsaPerDirectionOptions *apdo)
 392 {
 393     dolog("parameter | requested value | obtained value\n");
 394     dolog("format    |      %10d |     %10d\n", req->fmt, obtfmt);
 395     dolog("channels  |      %10d |     %10d\n",
 396           req->nchannels, obt->nchannels);
 397     dolog("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 398     dolog("============================================\n");
 399     dolog("requested: buffer len %" PRId32 " period len %" PRId32 "\n",
 400           apdo->buffer_length, apdo->period_length);
 401     dolog("obtained: samples %ld\n", obt->samples);
 402 }
 403
 404 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 405 {
 406     int err;
 407     snd_pcm_sw_params_t *sw_params;
 408
 409     snd_pcm_sw_params_alloca (&sw_params);
 410
 411     err = snd_pcm_sw_params_current (handle, sw_params);
 412     if (err < 0) {
 413         dolog ("Could not fully initialize DAC\n");
 414         alsa_logerr (err, "Failed to get current software parameters\n");
 415         return;
 416     }
 417
 418     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 419     if (err < 0) {
 420         dolog ("Could not fully initialize DAC\n");
 421         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 422                      threshold);
 423         return;
 424     }
 425
 426     err = snd_pcm_sw_params (handle, sw_params);
 427     if (err < 0) {
 428         dolog ("Could not fully initialize DAC\n");
 429         alsa_logerr (err, "Failed to set software parameters\n");
 430         return;
 431     }
 432 }
 433
 434 static int alsa_open(bool in, struct alsa_params_req *req,
 435                      struct alsa_params_obt *obt, snd_pcm_t **handlep,
 436                      Audiodev *dev)
 437 {
 438     AudiodevAlsaOptions *aopts = &dev->u.alsa;
 439     AudiodevAlsaPerDirectionOptions *apdo = in ? aopts->in : aopts->out;
 440     snd_pcm_t *handle;
 441     snd_pcm_hw_params_t *hw_params;
 442     int err;
 443     unsigned int freq, nchannels;
 444     const char *pcm_name = apdo->has_dev ? apdo->dev : "default";
 445     snd_pcm_uframes_t obt_buffer_size;
 446     const char *typ = in ? "ADC" : "DAC";
 447     snd_pcm_format_t obtfmt;
 448
 449     freq = req->freq;
 450     nchannels = req->nchannels;
 451
 452     snd_pcm_hw_params_alloca (&hw_params);
 453
 454     err = snd_pcm_open (
 455         &handle,
 456         pcm_name,
 457         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 458         SND_PCM_NONBLOCK
 459         );
 460     if (err < 0) {
 461         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 462         return -1;
 463     }
 464
 465     err = snd_pcm_hw_params_any (handle, hw_params);
 466     if (err < 0) {
 467         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 468         goto err;
 469     }
 470
 471     err = snd_pcm_hw_params_set_access (
 472         handle,
 473         hw_params,
 474         SND_PCM_ACCESS_RW_INTERLEAVED
 475         );
 476     if (err < 0) {
 477         alsa_logerr2 (err, typ, "Failed to set access type\n");
 478         goto err;
 479     }
 480
 481     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 482     if (err < 0) {
 483         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 484     }
 485
 486     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 487     if (err < 0) {
 488         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 489         goto err;
 490     }
 491
 492     err = snd_pcm_hw_params_set_channels_near (
 493         handle,
 494         hw_params,
 495         &nchannels
 496         );
 497     if (err < 0) {
 498         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 499                       req->nchannels);
 500         goto err;
 501     }
 502
 503     if (nchannels != 1 && nchannels != 2) {
 504         alsa_logerr2 (err, typ,
 505                       "Can not handle obtained number of channels %d\n",
 506                       nchannels);
 507         goto err;
 508     }
 509
 510     if (apdo->buffer_length) {
 511         int dir = 0;
 512         unsigned int btime = apdo->buffer_length;
 513
 514         err = snd_pcm_hw_params_set_buffer_time_near(
 515             handle, hw_params, &btime, &dir);
 516
 517         if (err < 0) {
 518             alsa_logerr2(err, typ, "Failed to set buffer time to %" PRId32 "\n",
 519                          apdo->buffer_length);
 520             goto err;
 521         }
 522
 523         if (apdo->has_buffer_length && btime != apdo->buffer_length) {
 524             dolog("Requested buffer time %" PRId32
 525                   " was rejected, using %u\n", apdo->buffer_length, btime);
 526         }
 527     }
 528
 529     if (apdo->period_length) {
 530         int dir = 0;
 531         unsigned int ptime = apdo->period_length;
 532
 533         err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, &ptime,
 534                                                      &dir);
 535
 536         if (err < 0) {
 537             alsa_logerr2(err, typ, "Failed to set period time to %" PRId32 "\n",
 538                          apdo->period_length);
 539             goto err;
 540         }
 541
 542         if (apdo->has_period_length && ptime != apdo->period_length) {
 543             dolog("Requested period time %" PRId32 " was rejected, using %d\n",
 544                   apdo->period_length, ptime);
 545         }
 546     }
 547
 548     err = snd_pcm_hw_params (handle, hw_params);
 549     if (err < 0) {
 550         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 551         goto err;
 552     }
 553
 554     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 555     if (err < 0) {
 556         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 557         goto err;
 558     }
 559
 560     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 561     if (err < 0) {
 562         alsa_logerr2 (err, typ, "Failed to get format\n");
 563         goto err;
 564     }
 565
 566     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 567         dolog ("Invalid format was returned %d\n", obtfmt);
 568         goto err;
 569     }
 570
 571     err = snd_pcm_prepare (handle);
 572     if (err < 0) {
 573         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 574         goto err;
 575     }
 576
 577     if (!in && aopts->has_threshold && aopts->threshold) {
 578         struct audsettings as = { .freq = freq };
 579         alsa_set_threshold(
 580             handle,
 581             audio_buffer_frames(qapi_AudiodevAlsaPerDirectionOptions_base(apdo),
 582                                 &as, aopts->threshold));
 583     }
 584
 585     obt->nchannels = nchannels;
 586     obt->freq = freq;
 587     obt->samples = obt_buffer_size;
 588
 589     *handlep = handle;
 590
 591     if (obtfmt != req->fmt ||
 592          obt->nchannels != req->nchannels ||
 593          obt->freq != req->freq) {
 594         dolog ("Audio parameters for %s\n", typ);
 595         alsa_dump_info(req, obt, obtfmt, apdo);
 596     }
 597
 598 #ifdef DEBUG
 599     alsa_dump_info(req, obt, obtfmt, pdo);
 600 #endif
 601     return 0;
 602
 603  err:
 604     alsa_anal_close1 (&handle);
 605     return -1;
 606 }
 607
 608 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 609 {
 610     snd_pcm_sframes_t avail;
 611
 612     avail = snd_pcm_avail_update (handle);
 613     if (avail < 0) {
 614         if (avail == -EPIPE) {
 615             if (!alsa_recover (handle)) {
 616                 avail = snd_pcm_avail_update (handle);
 617             }
 618         }
 619
 620         if (avail < 0) {
 621             alsa_logerr (avail,
 622                          "Could not obtain number of available frames\n");
 623             return -1;
 624         }
 625     }
 626
 627     return avail;
 628 }
 629
 630 static void alsa_write_pending (ALSAVoiceOut *alsa)
 631 {
 632     HWVoiceOut *hw = &alsa->hw;
 633
 634     while (alsa->pending) {
 635         int left_till_end_samples = hw->samples - alsa->wpos;
 636         int len = audio_MIN (alsa->pending, left_till_end_samples);
 637         char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
 638
 639         while (len) {
 640             snd_pcm_sframes_t written;
 641
 642             written = snd_pcm_writei (alsa->handle, src, len);
 643
 644             if (written <= 0) {
 645                 switch (written) {
 646                 case 0:
 647                     trace_alsa_wrote_zero(len);
 648                     return;
 649
 650                 case -EPIPE:
 651                     if (alsa_recover (alsa->handle)) {
 652                         alsa_logerr (written, "Failed to write %d frames\n",
 653                                      len);
 654                         return;
 655                     }
 656                     trace_alsa_xrun_out();
 657                     continue;
 658
 659                 case -ESTRPIPE:
 660                     /* stream is suspended and waiting for an
 661                        application recovery */
 662                     if (alsa_resume (alsa->handle)) {
 663                         alsa_logerr (written, "Failed to write %d frames\n",
 664                                      len);
 665                         return;
 666                     }
 667                     trace_alsa_resume_out();
 668                     continue;
 669
 670                 case -EAGAIN:
 671                     return;
 672
 673                 default:
 674                     alsa_logerr (written, "Failed to write %d frames from %p\n",
 675                                  len, src);
 676                     return;
 677                 }
 678             }
 679
 680             alsa->wpos = (alsa->wpos + written) % hw->samples;
 681             alsa->pending -= written;
 682             len -= written;
 683         }
 684     }
 685 }
 686
 687 static int alsa_run_out (HWVoiceOut *hw, int live)
 688 {
 689     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 690     int decr;
 691     snd_pcm_sframes_t avail;
 692
 693     avail = alsa_get_avail (alsa->handle);
 694     if (avail < 0) {
 695         dolog ("Could not get number of available playback frames\n");
 696         return 0;
 697     }
 698
 699     decr = audio_MIN (live, avail);
 700     decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
 701     alsa->pending += decr;
 702     alsa_write_pending (alsa);
 703     return decr;
 704 }
 705
 706 static void alsa_fini_out (HWVoiceOut *hw)
 707 {
 708     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 709
 710     ldebug ("alsa_fini\n");
 711     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 712
 713     g_free(alsa->pcm_buf);
 714     alsa->pcm_buf = NULL;
 715 }
 716
 717 static int alsa_init_out(HWVoiceOut *hw, struct audsettings *as,
 718                          void *drv_opaque)
 719 {
 720     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 721     struct alsa_params_req req;
 722     struct alsa_params_obt obt;
 723     snd_pcm_t *handle;
 724     struct audsettings obt_as;
 725     Audiodev *dev = drv_opaque;
 726
 727     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 728     req.freq = as->freq;
 729     req.nchannels = as->nchannels;
 730
 731     if (alsa_open(0, &req, &obt, &handle, dev)) {
 732         return -1;
 733     }
 734
 735     obt_as.freq = obt.freq;
 736     obt_as.nchannels = obt.nchannels;
 737     obt_as.fmt = obt.fmt;
 738     obt_as.endianness = obt.endianness;
 739
 740     audio_pcm_init_info (&hw->info, &obt_as);
 741     hw->samples = obt.samples;
 742
 743     alsa->pcm_buf = audio_calloc(__func__, obt.samples, 1 << hw->info.shift);
 744     if (!alsa->pcm_buf) {
 745         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 746                hw->samples, 1 << hw->info.shift);
 747         alsa_anal_close1 (&handle);
 748         return -1;
 749     }
 750
 751     alsa->handle = handle;
 752     alsa->dev = dev;
 753     return 0;
 754 }
 755
 756 #define VOICE_CTL_PAUSE 0
 757 #define VOICE_CTL_PREPARE 1
 758 #define VOICE_CTL_START 2
 759
 760 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int ctl)
 761 {
 762     int err;
 763
 764     if (ctl == VOICE_CTL_PAUSE) {
 765         err = snd_pcm_drop (handle);
 766         if (err < 0) {
 767             alsa_logerr (err, "Could not stop %s\n", typ);
 768             return -1;
 769         }
 770     }
 771     else {
 772         err = snd_pcm_prepare (handle);
 773         if (err < 0) {
 774             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 775             return -1;
 776         }
 777         if (ctl == VOICE_CTL_START) {
 778             err = snd_pcm_start(handle);
 779             if (err < 0) {
 780                 alsa_logerr (err, "Could not start handle for %s\n", typ);
 781                 return -1;
 782             }
 783         }
 784     }
 785
 786     return 0;
 787 }
 788
 789 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 790 {
 791     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 792     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.out;
 793
 794     switch (cmd) {
 795     case VOICE_ENABLE:
 796         {
 797             bool poll_mode = apdo->try_poll;
 798
 799             ldebug ("enabling voice\n");
 800             if (poll_mode && alsa_poll_out (hw)) {
 801                 poll_mode = 0;
 802             }
 803             hw->poll_mode = poll_mode;
 804             return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PREPARE);
 805         }
 806
 807     case VOICE_DISABLE:
 808         ldebug ("disabling voice\n");
 809         if (hw->poll_mode) {
 810             hw->poll_mode = 0;
 811             alsa_fini_poll (&alsa->pollhlp);
 812         }
 813         return alsa_voice_ctl (alsa->handle, "playback", VOICE_CTL_PAUSE);
 814     }
 815
 816     return -1;
 817 }
 818
 819 static int alsa_init_in(HWVoiceIn *hw, struct audsettings *as, void *drv_opaque)
 820 {
 821     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 822     struct alsa_params_req req;
 823     struct alsa_params_obt obt;
 824     snd_pcm_t *handle;
 825     struct audsettings obt_as;
 826     Audiodev *dev = drv_opaque;
 827
 828     req.fmt = aud_to_alsafmt (as->fmt, as->endianness);
 829     req.freq = as->freq;
 830     req.nchannels = as->nchannels;
 831
 832     if (alsa_open(1, &req, &obt, &handle, dev)) {
 833         return -1;
 834     }
 835
 836     obt_as.freq = obt.freq;
 837     obt_as.nchannels = obt.nchannels;
 838     obt_as.fmt = obt.fmt;
 839     obt_as.endianness = obt.endianness;
 840
 841     audio_pcm_init_info (&hw->info, &obt_as);
 842     hw->samples = obt.samples;
 843
 844     alsa->pcm_buf = audio_calloc(__func__, hw->samples, 1 << hw->info.shift);
 845     if (!alsa->pcm_buf) {
 846         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 847                hw->samples, 1 << hw->info.shift);
 848         alsa_anal_close1 (&handle);
 849         return -1;
 850     }
 851
 852     alsa->handle = handle;
 853     alsa->dev = dev;
 854     return 0;
 855 }
 856
 857 static void alsa_fini_in (HWVoiceIn *hw)
 858 {
 859     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 860
 861     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 862
 863     g_free(alsa->pcm_buf);
 864     alsa->pcm_buf = NULL;
 865 }
 866
 867 static int alsa_run_in (HWVoiceIn *hw)
 868 {
 869     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 870     int hwshift = hw->info.shift;
 871     int i;
 872     int live = audio_pcm_hw_get_live_in (hw);
 873     int dead = hw->samples - live;
 874     int decr;
 875     struct {
 876         int add;
 877         int len;
 878     } bufs[2] = {
 879         { .add = hw->wpos, .len = 0 },
 880         { .add = 0,        .len = 0 }
 881     };
 882     snd_pcm_sframes_t avail;
 883     snd_pcm_uframes_t read_samples = 0;
 884
 885     if (!dead) {
 886         return 0;
 887     }
 888
 889     avail = alsa_get_avail (alsa->handle);
 890     if (avail < 0) {
 891         dolog ("Could not get number of captured frames\n");
 892         return 0;
 893     }
 894
 895     if (!avail) {
 896         snd_pcm_state_t state;
 897
 898         state = snd_pcm_state (alsa->handle);
 899         switch (state) {
 900         case SND_PCM_STATE_PREPARED:
 901             avail = hw->samples;
 902             break;
 903         case SND_PCM_STATE_SUSPENDED:
 904             /* stream is suspended and waiting for an application recovery */
 905             if (alsa_resume (alsa->handle)) {
 906                 dolog ("Failed to resume suspended input stream\n");
 907                 return 0;
 908             }
 909             trace_alsa_resume_in();
 910             break;
 911         default:
 912             trace_alsa_no_frames(state);
 913             return 0;
 914         }
 915     }
 916
 917     decr = audio_MIN (dead, avail);
 918     if (!decr) {
 919         return 0;
 920     }
 921
 922     if (hw->wpos + decr > hw->samples) {
 923         bufs[0].len = (hw->samples - hw->wpos);
 924         bufs[1].len = (decr - (hw->samples - hw->wpos));
 925     }
 926     else {
 927         bufs[0].len = decr;
 928     }
 929
 930     for (i = 0; i < 2; ++i) {
 931         void *src;
 932         struct st_sample *dst;
 933         snd_pcm_sframes_t nread;
 934         snd_pcm_uframes_t len;
 935
 936         len = bufs[i].len;
 937
 938         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
 939         dst = hw->conv_buf + bufs[i].add;
 940
 941         while (len) {
 942             nread = snd_pcm_readi (alsa->handle, src, len);
 943
 944             if (nread <= 0) {
 945                 switch (nread) {
 946                 case 0:
 947                     trace_alsa_read_zero(len);
 948                     goto exit;
 949
 950                 case -EPIPE:
 951                     if (alsa_recover (alsa->handle)) {
 952                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
 953                         goto exit;
 954                     }
 955                     trace_alsa_xrun_in();
 956                     continue;
 957
 958                 case -EAGAIN:
 959                     goto exit;
 960
 961                 default:
 962                     alsa_logerr (
 963                         nread,
 964                         "Failed to read %ld frames from %p\n",
 965                         len,
 966                         src
 967                         );
 968                     goto exit;
 969                 }
 970             }
 971
 972             hw->conv (dst, src, nread);
 973
 974             src = advance (src, nread << hwshift);
 975             dst += nread;
 976
 977             read_samples += nread;
 978             len -= nread;
 979         }
 980     }
 981
 982  exit:
 983     hw->wpos = (hw->wpos + read_samples) % hw->samples;
 984     return read_samples;
 985 }
 986
 987 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
 988 {
 989     return audio_pcm_sw_read (sw, buf, size);
 990 }
 991
 992 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 993 {
 994     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 995     AudiodevAlsaPerDirectionOptions *apdo = alsa->dev->u.alsa.in;
 996
 997     switch (cmd) {
 998     case VOICE_ENABLE:
 999         {
1000             bool poll_mode = apdo->try_poll;
1001
1002             ldebug ("enabling voice\n");
1003             if (poll_mode && alsa_poll_in (hw)) {
1004                 poll_mode = 0;
1005             }
1006             hw->poll_mode = poll_mode;
1007
1008             return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_START);
1009         }
1010
1011     case VOICE_DISABLE:
1012         ldebug ("disabling voice\n");
1013         if (hw->poll_mode) {
1014             hw->poll_mode = 0;
1015             alsa_fini_poll (&alsa->pollhlp);
1016         }
1017         return alsa_voice_ctl (alsa->handle, "capture", VOICE_CTL_PAUSE);
1018     }
1019
1020     return -1;
1021 }
1022
1023 static void alsa_init_per_direction(AudiodevAlsaPerDirectionOptions *apdo)
1024 {
1025     if (!apdo->has_try_poll) {
1026         apdo->try_poll = true;
1027         apdo->has_try_poll = true;
1028     }
1029 }
1030
1031 static void *alsa_audio_init(Audiodev *dev)
1032 {
1033     AudiodevAlsaOptions *aopts;
1034     assert(dev->driver == AUDIODEV_DRIVER_ALSA);
1035
1036     aopts = &dev->u.alsa;
1037     alsa_init_per_direction(aopts->in);
1038     alsa_init_per_direction(aopts->out);
1039
1040     /*
1041      * need to define them, as otherwise alsa produces no sound
1042      * doesn't set has_* so alsa_open can identify it wasn't set by the user
1043      */
1044     if (!dev->u.alsa.out->has_period_length) {
1045         /* 1024 frames assuming 44100Hz */
1046         dev->u.alsa.out->period_length = 1024 * 1000000 / 44100;
1047     }
1048     if (!dev->u.alsa.out->has_buffer_length) {
1049         /* 4096 frames assuming 44100Hz */
1050         dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100;
1051     }
1052
1053     /*
1054      * OptsVisitor sets unspecified optional fields to zero, but do not depend
1055      * on it...
1056      */
1057     if (!dev->u.alsa.in->has_period_length) {
1058         dev->u.alsa.in->period_length = 0;
1059     }
1060     if (!dev->u.alsa.in->has_buffer_length) {
1061         dev->u.alsa.in->buffer_length = 0;
1062     }
1063
1064     return dev;
1065 }
1066
1067 static void alsa_audio_fini (void *opaque)
1068 {
1069 }
1070
1071 static struct audio_pcm_ops alsa_pcm_ops = {
1072     .init_out = alsa_init_out,
1073     .fini_out = alsa_fini_out,
1074     .run_out  = alsa_run_out,
1075     .write    = alsa_write,
1076     .ctl_out  = alsa_ctl_out,
1077
1078     .init_in  = alsa_init_in,
1079     .fini_in  = alsa_fini_in,
1080     .run_in   = alsa_run_in,
1081     .read     = alsa_read,
1082     .ctl_in   = alsa_ctl_in,
1083 };
1084
1085 static struct audio_driver alsa_audio_driver = {
1086     .name           = "alsa",
1087     .descr          = "ALSA http://www.alsa-project.org",
1088     .init           = alsa_audio_init,
1089     .fini           = alsa_audio_fini,
1090     .pcm_ops        = &alsa_pcm_ops,
1091     .can_be_default = 1,
1092     .max_voices_out = INT_MAX,
1093     .max_voices_in  = INT_MAX,
1094     .voice_size_out = sizeof (ALSAVoiceOut),
1095     .voice_size_in  = sizeof (ALSAVoiceIn)
1096 };
1097
1098 static void register_audio_alsa(void)
1099 {
1100     audio_driver_register(&alsa_audio_driver);
1101 }
1102 type_init(register_audio_alsa);