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 <alsa/asoundlib.h>
  25 #include "qemu-common.h"
  26 #include "qemu-char.h"
  27 #include "audio.h"
  28
  29 #if QEMU_GNUC_PREREQ(4, 3)
  30 #pragma GCC diagnostic ignored "-Waddress"
  31 #endif
  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 } ALSAVoiceOut;
  51
  52 typedef struct ALSAVoiceIn {
  53     HWVoiceIn hw;
  54     snd_pcm_t *handle;
  55     void *pcm_buf;
  56     struct pollhlp pollhlp;
  57 } ALSAVoiceIn;
  58
  59 static struct {
  60     int size_in_usec_in;
  61     int size_in_usec_out;
  62     const char *pcm_name_in;
  63     const char *pcm_name_out;
  64     unsigned int buffer_size_in;
  65     unsigned int period_size_in;
  66     unsigned int buffer_size_out;
  67     unsigned int period_size_out;
  68     unsigned int threshold;
  69
  70     int buffer_size_in_overridden;
  71     int period_size_in_overridden;
  72
  73     int buffer_size_out_overridden;
  74     int period_size_out_overridden;
  75     int verbose;
  76 } conf = {
  77     .buffer_size_out = 4096,
  78     .period_size_out = 1024,
  79     .pcm_name_out = "default",
  80     .pcm_name_in = "default",
  81 };
  82
  83 struct alsa_params_req {
  84     int freq;
  85     snd_pcm_format_t fmt;
  86     int nchannels;
  87     int size_in_usec;
  88     int override_mask;
  89     unsigned int buffer_size;
  90     unsigned int period_size;
  91 };
  92
  93 struct alsa_params_obt {
  94     int freq;
  95     audfmt_e fmt;
  96     int endianness;
  97     int nchannels;
  98     snd_pcm_uframes_t samples;
  99 };
 100
 101 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
 102 {
 103     va_list ap;
 104
 105     va_start (ap, fmt);
 106     AUD_vlog (AUDIO_CAP, fmt, ap);
 107     va_end (ap);
 108
 109     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 110 }
 111
 112 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
 113     int err,
 114     const char *typ,
 115     const char *fmt,
 116     ...
 117     )
 118 {
 119     va_list ap;
 120
 121     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
 122
 123     va_start (ap, fmt);
 124     AUD_vlog (AUDIO_CAP, fmt, ap);
 125     va_end (ap);
 126
 127     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 128 }
 129
 130 static void alsa_fini_poll (struct pollhlp *hlp)
 131 {
 132     int i;
 133     struct pollfd *pfds = hlp->pfds;
 134
 135     if (pfds) {
 136         for (i = 0; i < hlp->count; ++i) {
 137             qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
 138         }
 139         qemu_free (pfds);
 140     }
 141     hlp->pfds = NULL;
 142     hlp->count = 0;
 143     hlp->handle = NULL;
 144 }
 145
 146 static void alsa_anal_close1 (snd_pcm_t **handlep)
 147 {
 148     int err = snd_pcm_close (*handlep);
 149     if (err) {
 150         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 151     }
 152     *handlep = NULL;
 153 }
 154
 155 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
 156 {
 157     alsa_fini_poll (hlp);
 158     alsa_anal_close1 (handlep);
 159 }
 160
 161 static int alsa_recover (snd_pcm_t *handle)
 162 {
 163     int err = snd_pcm_prepare (handle);
 164     if (err < 0) {
 165         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 166         return -1;
 167     }
 168     return 0;
 169 }
 170
 171 static int alsa_resume (snd_pcm_t *handle)
 172 {
 173     int err = snd_pcm_resume (handle);
 174     if (err < 0) {
 175         alsa_logerr (err, "Failed to resume handle %p\n", handle);
 176         return -1;
 177     }
 178     return 0;
 179 }
 180
 181 static void alsa_poll_handler (void *opaque)
 182 {
 183     int err, count;
 184     snd_pcm_state_t state;
 185     struct pollhlp *hlp = opaque;
 186     unsigned short revents;
 187
 188     count = poll (hlp->pfds, hlp->count, 0);
 189     if (count < 0) {
 190         dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
 191         return;
 192     }
 193
 194     if (!count) {
 195         return;
 196     }
 197
 198     /* XXX: ALSA example uses initial count, not the one returned by
 199        poll, correct? */
 200     err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
 201                                             hlp->count, &revents);
 202     if (err < 0) {
 203         alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
 204         return;
 205     }
 206
 207     if (!(revents & hlp->mask)) {
 208         if (conf.verbose) {
 209             dolog ("revents = %d\n", revents);
 210         }
 211         return;
 212     }
 213
 214     state = snd_pcm_state (hlp->handle);
 215     switch (state) {
 216     case SND_PCM_STATE_SETUP:
 217         alsa_recover (hlp->handle);
 218         break;
 219
 220     case SND_PCM_STATE_XRUN:
 221         alsa_recover (hlp->handle);
 222         break;
 223
 224     case SND_PCM_STATE_SUSPENDED:
 225         alsa_resume (hlp->handle);
 226         break;
 227
 228     case SND_PCM_STATE_PREPARED:
 229         audio_run ("alsa run (prepared)");
 230         break;
 231
 232     case SND_PCM_STATE_RUNNING:
 233         audio_run ("alsa run (running)");
 234         break;
 235
 236     default:
 237         dolog ("Unexpected state %d\n", state);
 238     }
 239 }
 240
 241 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
 242 {
 243     int i, count, err;
 244     struct pollfd *pfds;
 245
 246     count = snd_pcm_poll_descriptors_count (handle);
 247     if (count <= 0) {
 248         dolog ("Could not initialize poll mode\n"
 249                "Invalid number of poll descriptors %d\n", count);
 250         return -1;
 251     }
 252
 253     pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
 254     if (!pfds) {
 255         dolog ("Could not initialize poll mode\n");
 256         return -1;
 257     }
 258
 259     err = snd_pcm_poll_descriptors (handle, pfds, count);
 260     if (err < 0) {
 261         alsa_logerr (err, "Could not initialize poll mode\n"
 262                      "Could not obtain poll descriptors\n");
 263         qemu_free (pfds);
 264         return -1;
 265     }
 266
 267     for (i = 0; i < count; ++i) {
 268         if (pfds[i].events & POLLIN) {
 269             err = qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler,
 270                                        NULL, hlp);
 271         }
 272         if (pfds[i].events & POLLOUT) {
 273             if (conf.verbose) {
 274                 dolog ("POLLOUT %d %d\n", i, pfds[i].fd);
 275             }
 276             err = qemu_set_fd_handler (pfds[i].fd, NULL,
 277                                        alsa_poll_handler, hlp);
 278         }
 279         if (conf.verbose) {
 280             dolog ("Set handler events=%#x index=%d fd=%d err=%d\n",
 281                    pfds[i].events, i, pfds[i].fd, err);
 282         }
 283
 284         if (err) {
 285             dolog ("Failed to set handler events=%#x index=%d fd=%d err=%d\n",
 286                    pfds[i].events, i, pfds[i].fd, err);
 287
 288             while (i--) {
 289                 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
 290             }
 291             qemu_free (pfds);
 292             return -1;
 293         }
 294     }
 295     hlp->pfds = pfds;
 296     hlp->count = count;
 297     hlp->handle = handle;
 298     hlp->mask = mask;
 299     return 0;
 300 }
 301
 302 static int alsa_poll_out (HWVoiceOut *hw)
 303 {
 304     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 305
 306     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
 307 }
 308
 309 static int alsa_poll_in (HWVoiceIn *hw)
 310 {
 311     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 312
 313     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
 314 }
 315
 316 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
 317 {
 318     return audio_pcm_sw_write (sw, buf, len);
 319 }
 320
 321 static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
 322 {
 323     switch (fmt) {
 324     case AUD_FMT_S8:
 325         return SND_PCM_FORMAT_S8;
 326
 327     case AUD_FMT_U8:
 328         return SND_PCM_FORMAT_U8;
 329
 330     case AUD_FMT_S16:
 331         return SND_PCM_FORMAT_S16_LE;
 332
 333     case AUD_FMT_U16:
 334         return SND_PCM_FORMAT_U16_LE;
 335
 336     case AUD_FMT_S32:
 337         return SND_PCM_FORMAT_S32_LE;
 338
 339     case AUD_FMT_U32:
 340         return SND_PCM_FORMAT_U32_LE;
 341
 342     default:
 343         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 344 #ifdef DEBUG_AUDIO
 345         abort ();
 346 #endif
 347         return SND_PCM_FORMAT_U8;
 348     }
 349 }
 350
 351 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
 352                            int *endianness)
 353 {
 354     switch (alsafmt) {
 355     case SND_PCM_FORMAT_S8:
 356         *endianness = 0;
 357         *fmt = AUD_FMT_S8;
 358         break;
 359
 360     case SND_PCM_FORMAT_U8:
 361         *endianness = 0;
 362         *fmt = AUD_FMT_U8;
 363         break;
 364
 365     case SND_PCM_FORMAT_S16_LE:
 366         *endianness = 0;
 367         *fmt = AUD_FMT_S16;
 368         break;
 369
 370     case SND_PCM_FORMAT_U16_LE:
 371         *endianness = 0;
 372         *fmt = AUD_FMT_U16;
 373         break;
 374
 375     case SND_PCM_FORMAT_S16_BE:
 376         *endianness = 1;
 377         *fmt = AUD_FMT_S16;
 378         break;
 379
 380     case SND_PCM_FORMAT_U16_BE:
 381         *endianness = 1;
 382         *fmt = AUD_FMT_U16;
 383         break;
 384
 385     case SND_PCM_FORMAT_S32_LE:
 386         *endianness = 0;
 387         *fmt = AUD_FMT_S32;
 388         break;
 389
 390     case SND_PCM_FORMAT_U32_LE:
 391         *endianness = 0;
 392         *fmt = AUD_FMT_U32;
 393         break;
 394
 395     case SND_PCM_FORMAT_S32_BE:
 396         *endianness = 1;
 397         *fmt = AUD_FMT_S32;
 398         break;
 399
 400     case SND_PCM_FORMAT_U32_BE:
 401         *endianness = 1;
 402         *fmt = AUD_FMT_U32;
 403         break;
 404
 405     default:
 406         dolog ("Unrecognized audio format %d\n", alsafmt);
 407         return -1;
 408     }
 409
 410     return 0;
 411 }
 412
 413 static void alsa_dump_info (struct alsa_params_req *req,
 414                             struct alsa_params_obt *obt)
 415 {
 416     dolog ("parameter | requested value | obtained value\n");
 417     dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
 418     dolog ("channels  |      %10d |     %10d\n",
 419            req->nchannels, obt->nchannels);
 420     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 421     dolog ("============================================\n");
 422     dolog ("requested: buffer size %d period size %d\n",
 423            req->buffer_size, req->period_size);
 424     dolog ("obtained: samples %ld\n", obt->samples);
 425 }
 426
 427 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 428 {
 429     int err;
 430     snd_pcm_sw_params_t *sw_params;
 431
 432     snd_pcm_sw_params_alloca (&sw_params);
 433
 434     err = snd_pcm_sw_params_current (handle, sw_params);
 435     if (err < 0) {
 436         dolog ("Could not fully initialize DAC\n");
 437         alsa_logerr (err, "Failed to get current software parameters\n");
 438         return;
 439     }
 440
 441     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 442     if (err < 0) {
 443         dolog ("Could not fully initialize DAC\n");
 444         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 445                      threshold);
 446         return;
 447     }
 448
 449     err = snd_pcm_sw_params (handle, sw_params);
 450     if (err < 0) {
 451         dolog ("Could not fully initialize DAC\n");
 452         alsa_logerr (err, "Failed to set software parameters\n");
 453         return;
 454     }
 455 }
 456
 457 static int alsa_open (int in, struct alsa_params_req *req,
 458                       struct alsa_params_obt *obt, snd_pcm_t **handlep)
 459 {
 460     snd_pcm_t *handle;
 461     snd_pcm_hw_params_t *hw_params;
 462     int err;
 463     int size_in_usec;
 464     unsigned int freq, nchannels;
 465     const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
 466     snd_pcm_uframes_t obt_buffer_size;
 467     const char *typ = in ? "ADC" : "DAC";
 468     snd_pcm_format_t obtfmt;
 469
 470     freq = req->freq;
 471     nchannels = req->nchannels;
 472     size_in_usec = req->size_in_usec;
 473
 474     snd_pcm_hw_params_alloca (&hw_params);
 475
 476     err = snd_pcm_open (
 477         &handle,
 478         pcm_name,
 479         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 480         SND_PCM_NONBLOCK
 481         );
 482     if (err < 0) {
 483         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 484         return -1;
 485     }
 486
 487     err = snd_pcm_hw_params_any (handle, hw_params);
 488     if (err < 0) {
 489         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 490         goto err;
 491     }
 492
 493     err = snd_pcm_hw_params_set_access (
 494         handle,
 495         hw_params,
 496         SND_PCM_ACCESS_RW_INTERLEAVED
 497         );
 498     if (err < 0) {
 499         alsa_logerr2 (err, typ, "Failed to set access type\n");
 500         goto err;
 501     }
 502
 503     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 504     if (err < 0 && conf.verbose) {
 505         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 506     }
 507
 508     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 509     if (err < 0) {
 510         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 511         goto err;
 512     }
 513
 514     err = snd_pcm_hw_params_set_channels_near (
 515         handle,
 516         hw_params,
 517         &nchannels
 518         );
 519     if (err < 0) {
 520         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 521                       req->nchannels);
 522         goto err;
 523     }
 524
 525     if (nchannels != 1 && nchannels != 2) {
 526         alsa_logerr2 (err, typ,
 527                       "Can not handle obtained number of channels %d\n",
 528                       nchannels);
 529         goto err;
 530     }
 531
 532     if (req->buffer_size) {
 533         unsigned long obt;
 534
 535         if (size_in_usec) {
 536             int dir = 0;
 537             unsigned int btime = req->buffer_size;
 538
 539             err = snd_pcm_hw_params_set_buffer_time_near (
 540                 handle,
 541                 hw_params,
 542                 &btime,
 543                 &dir
 544                 );
 545             obt = btime;
 546         }
 547         else {
 548             snd_pcm_uframes_t bsize = req->buffer_size;
 549
 550             err = snd_pcm_hw_params_set_buffer_size_near (
 551                 handle,
 552                 hw_params,
 553                 &bsize
 554                 );
 555             obt = bsize;
 556         }
 557         if (err < 0) {
 558             alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
 559                           size_in_usec ? "time" : "size", req->buffer_size);
 560             goto err;
 561         }
 562
 563         if ((req->override_mask & 2) && (obt - req->buffer_size))
 564             dolog ("Requested buffer %s %u was rejected, using %lu\n",
 565                    size_in_usec ? "time" : "size", req->buffer_size, obt);
 566     }
 567
 568     if (req->period_size) {
 569         unsigned long obt;
 570
 571         if (size_in_usec) {
 572             int dir = 0;
 573             unsigned int ptime = req->period_size;
 574
 575             err = snd_pcm_hw_params_set_period_time_near (
 576                 handle,
 577                 hw_params,
 578                 &ptime,
 579                 &dir
 580                 );
 581             obt = ptime;
 582         }
 583         else {
 584             int dir = 0;
 585             snd_pcm_uframes_t psize = req->period_size;
 586
 587             err = snd_pcm_hw_params_set_period_size_near (
 588                 handle,
 589                 hw_params,
 590                 &psize,
 591                 &dir
 592                 );
 593             obt = psize;
 594         }
 595
 596         if (err < 0) {
 597             alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
 598                           size_in_usec ? "time" : "size", req->period_size);
 599             goto err;
 600         }
 601
 602         if (((req->override_mask & 1) && (obt - req->period_size)))
 603             dolog ("Requested period %s %u was rejected, using %lu\n",
 604                    size_in_usec ? "time" : "size", req->period_size, obt);
 605     }
 606
 607     err = snd_pcm_hw_params (handle, hw_params);
 608     if (err < 0) {
 609         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 610         goto err;
 611     }
 612
 613     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 614     if (err < 0) {
 615         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 616         goto err;
 617     }
 618
 619     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 620     if (err < 0) {
 621         alsa_logerr2 (err, typ, "Failed to get format\n");
 622         goto err;
 623     }
 624
 625     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 626         dolog ("Invalid format was returned %d\n", obtfmt);
 627         goto err;
 628     }
 629
 630     err = snd_pcm_prepare (handle);
 631     if (err < 0) {
 632         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 633         goto err;
 634     }
 635
 636     if (!in && conf.threshold) {
 637         snd_pcm_uframes_t threshold;
 638         int bytes_per_sec;
 639
 640         bytes_per_sec = freq << (nchannels == 2);
 641
 642         switch (obt->fmt) {
 643         case AUD_FMT_S8:
 644         case AUD_FMT_U8:
 645             break;
 646
 647         case AUD_FMT_S16:
 648         case AUD_FMT_U16:
 649             bytes_per_sec <<= 1;
 650             break;
 651
 652         case AUD_FMT_S32:
 653         case AUD_FMT_U32:
 654             bytes_per_sec <<= 2;
 655             break;
 656         }
 657
 658         threshold = (conf.threshold * bytes_per_sec) / 1000;
 659         alsa_set_threshold (handle, threshold);
 660     }
 661
 662     obt->nchannels = nchannels;
 663     obt->freq = freq;
 664     obt->samples = obt_buffer_size;
 665
 666     *handlep = handle;
 667
 668     if (conf.verbose &&
 669         (obt->fmt != req->fmt ||
 670          obt->nchannels != req->nchannels ||
 671          obt->freq != req->freq)) {
 672         dolog ("Audio parameters for %s\n", typ);
 673         alsa_dump_info (req, obt);
 674     }
 675
 676 #ifdef DEBUG
 677     alsa_dump_info (req, obt);
 678 #endif
 679     return 0;
 680
 681  err:
 682     alsa_anal_close1 (&handle);
 683     return -1;
 684 }
 685
 686 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 687 {
 688     snd_pcm_sframes_t avail;
 689
 690     avail = snd_pcm_avail_update (handle);
 691     if (avail < 0) {
 692         if (avail == -EPIPE) {
 693             if (!alsa_recover (handle)) {
 694                 avail = snd_pcm_avail_update (handle);
 695             }
 696         }
 697
 698         if (avail < 0) {
 699             alsa_logerr (avail,
 700                          "Could not obtain number of available frames\n");
 701             return -1;
 702         }
 703     }
 704
 705     return avail;
 706 }
 707
 708 static void alsa_write_pending (ALSAVoiceOut *alsa)
 709 {
 710     HWVoiceOut *hw = &alsa->hw;
 711
 712     while (alsa->pending) {
 713         int left_till_end_samples = hw->samples - alsa->wpos;
 714         int len = audio_MIN (alsa->pending, left_till_end_samples);
 715         char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
 716
 717         while (len) {
 718             snd_pcm_sframes_t written;
 719
 720             written = snd_pcm_writei (alsa->handle, src, len);
 721
 722             if (written <= 0) {
 723                 switch (written) {
 724                 case 0:
 725                     if (conf.verbose) {
 726                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 727                     }
 728                     return;
 729
 730                 case -EPIPE:
 731                     if (alsa_recover (alsa->handle)) {
 732                         alsa_logerr (written, "Failed to write %d frames\n",
 733                                      len);
 734                         return;
 735                     }
 736                     if (conf.verbose) {
 737                         dolog ("Recovering from playback xrun\n");
 738                     }
 739                     continue;
 740
 741                 case -ESTRPIPE:
 742                     /* stream is suspended and waiting for an
 743                        application recovery */
 744                     if (alsa_resume (alsa->handle)) {
 745                         alsa_logerr (written, "Failed to write %d frames\n",
 746                                      len);
 747                         return;
 748                     }
 749                     if (conf.verbose) {
 750                         dolog ("Resuming suspended output stream\n");
 751                     }
 752                     continue;
 753
 754                 case -EAGAIN:
 755                     return;
 756
 757                 default:
 758                     alsa_logerr (written, "Failed to write %d frames from %p\n",
 759                                  len, src);
 760                     return;
 761                 }
 762             }
 763
 764             alsa->wpos = (alsa->wpos + written) % hw->samples;
 765             alsa->pending -= written;
 766             len -= written;
 767         }
 768     }
 769 }
 770
 771 static int alsa_run_out (HWVoiceOut *hw, int live)
 772 {
 773     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 774     int decr;
 775     snd_pcm_sframes_t avail;
 776
 777     avail = alsa_get_avail (alsa->handle);
 778     if (avail < 0) {
 779         dolog ("Could not get number of available playback frames\n");
 780         return 0;
 781     }
 782
 783     decr = audio_MIN (live, avail);
 784     decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
 785     alsa->pending += decr;
 786     alsa_write_pending (alsa);
 787     return decr;
 788 }
 789
 790 static void alsa_fini_out (HWVoiceOut *hw)
 791 {
 792     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 793
 794     ldebug ("alsa_fini\n");
 795     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 796
 797     if (alsa->pcm_buf) {
 798         qemu_free (alsa->pcm_buf);
 799         alsa->pcm_buf = NULL;
 800     }
 801 }
 802
 803 static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
 804 {
 805     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 806     struct alsa_params_req req;
 807     struct alsa_params_obt obt;
 808     snd_pcm_t *handle;
 809     struct audsettings obt_as;
 810
 811     req.fmt = aud_to_alsafmt (as->fmt);
 812     req.freq = as->freq;
 813     req.nchannels = as->nchannels;
 814     req.period_size = conf.period_size_out;
 815     req.buffer_size = conf.buffer_size_out;
 816     req.size_in_usec = conf.size_in_usec_out;
 817     req.override_mask =
 818         (conf.period_size_out_overridden ? 1 : 0) |
 819         (conf.buffer_size_out_overridden ? 2 : 0);
 820
 821     if (alsa_open (0, &req, &obt, &handle)) {
 822         return -1;
 823     }
 824
 825     obt_as.freq = obt.freq;
 826     obt_as.nchannels = obt.nchannels;
 827     obt_as.fmt = obt.fmt;
 828     obt_as.endianness = obt.endianness;
 829
 830     audio_pcm_init_info (&hw->info, &obt_as);
 831     hw->samples = obt.samples;
 832
 833     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 834     if (!alsa->pcm_buf) {
 835         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 836                hw->samples, 1 << hw->info.shift);
 837         alsa_anal_close1 (&handle);
 838         return -1;
 839     }
 840
 841     alsa->handle = handle;
 842     return 0;
 843 }
 844
 845 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
 846 {
 847     int err;
 848
 849     if (pause) {
 850         err = snd_pcm_drop (handle);
 851         if (err < 0) {
 852             alsa_logerr (err, "Could not stop %s\n", typ);
 853             return -1;
 854         }
 855     }
 856     else {
 857         err = snd_pcm_prepare (handle);
 858         if (err < 0) {
 859             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 860             return -1;
 861         }
 862     }
 863
 864     return 0;
 865 }
 866
 867 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 868 {
 869     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 870
 871     switch (cmd) {
 872     case VOICE_ENABLE:
 873         {
 874             va_list ap;
 875             int poll_mode;
 876
 877             va_start (ap, cmd);
 878             poll_mode = va_arg (ap, int);
 879             va_end (ap);
 880
 881             ldebug ("enabling voice\n");
 882             if (poll_mode && alsa_poll_out (hw)) {
 883                 poll_mode = 0;
 884             }
 885             hw->poll_mode = poll_mode;
 886             return alsa_voice_ctl (alsa->handle, "playback", 0);
 887         }
 888
 889     case VOICE_DISABLE:
 890         ldebug ("disabling voice\n");
 891         return alsa_voice_ctl (alsa->handle, "playback", 1);
 892     }
 893
 894     return -1;
 895 }
 896
 897 static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
 898 {
 899     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 900     struct alsa_params_req req;
 901     struct alsa_params_obt obt;
 902     snd_pcm_t *handle;
 903     struct audsettings obt_as;
 904
 905     req.fmt = aud_to_alsafmt (as->fmt);
 906     req.freq = as->freq;
 907     req.nchannels = as->nchannels;
 908     req.period_size = conf.period_size_in;
 909     req.buffer_size = conf.buffer_size_in;
 910     req.size_in_usec = conf.size_in_usec_in;
 911     req.override_mask =
 912         (conf.period_size_in_overridden ? 1 : 0) |
 913         (conf.buffer_size_in_overridden ? 2 : 0);
 914
 915     if (alsa_open (1, &req, &obt, &handle)) {
 916         return -1;
 917     }
 918
 919     obt_as.freq = obt.freq;
 920     obt_as.nchannels = obt.nchannels;
 921     obt_as.fmt = obt.fmt;
 922     obt_as.endianness = obt.endianness;
 923
 924     audio_pcm_init_info (&hw->info, &obt_as);
 925     hw->samples = obt.samples;
 926
 927     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 928     if (!alsa->pcm_buf) {
 929         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 930                hw->samples, 1 << hw->info.shift);
 931         alsa_anal_close1 (&handle);
 932         return -1;
 933     }
 934
 935     alsa->handle = handle;
 936     return 0;
 937 }
 938
 939 static void alsa_fini_in (HWVoiceIn *hw)
 940 {
 941     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 942
 943     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 944
 945     if (alsa->pcm_buf) {
 946         qemu_free (alsa->pcm_buf);
 947         alsa->pcm_buf = NULL;
 948     }
 949 }
 950
 951 static int alsa_run_in (HWVoiceIn *hw)
 952 {
 953     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 954     int hwshift = hw->info.shift;
 955     int i;
 956     int live = audio_pcm_hw_get_live_in (hw);
 957     int dead = hw->samples - live;
 958     int decr;
 959     struct {
 960         int add;
 961         int len;
 962     } bufs[2] = {
 963         { .add = hw->wpos, .len = 0 },
 964         { .add = 0,        .len = 0 }
 965     };
 966     snd_pcm_sframes_t avail;
 967     snd_pcm_uframes_t read_samples = 0;
 968
 969     if (!dead) {
 970         return 0;
 971     }
 972
 973     avail = alsa_get_avail (alsa->handle);
 974     if (avail < 0) {
 975         dolog ("Could not get number of captured frames\n");
 976         return 0;
 977     }
 978
 979     if (!avail) {
 980         snd_pcm_state_t state;
 981
 982         state = snd_pcm_state (alsa->handle);
 983         switch (state) {
 984         case SND_PCM_STATE_PREPARED:
 985             avail = hw->samples;
 986             break;
 987         case SND_PCM_STATE_SUSPENDED:
 988             /* stream is suspended and waiting for an application recovery */
 989             if (alsa_resume (alsa->handle)) {
 990                 dolog ("Failed to resume suspended input stream\n");
 991                 return 0;
 992             }
 993             if (conf.verbose) {
 994                 dolog ("Resuming suspended input stream\n");
 995             }
 996             break;
 997         default:
 998             if (conf.verbose) {
 999                 dolog ("No frames available and ALSA state is %d\n", state);
1000             }
1001             return 0;
1002         }
1003     }
1004
1005     decr = audio_MIN (dead, avail);
1006     if (!decr) {
1007         return 0;
1008     }
1009
1010     if (hw->wpos + decr > hw->samples) {
1011         bufs[0].len = (hw->samples - hw->wpos);
1012         bufs[1].len = (decr - (hw->samples - hw->wpos));
1013     }
1014     else {
1015         bufs[0].len = decr;
1016     }
1017
1018     for (i = 0; i < 2; ++i) {
1019         void *src;
1020         struct st_sample *dst;
1021         snd_pcm_sframes_t nread;
1022         snd_pcm_uframes_t len;
1023
1024         len = bufs[i].len;
1025
1026         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
1027         dst = hw->conv_buf + bufs[i].add;
1028
1029         while (len) {
1030             nread = snd_pcm_readi (alsa->handle, src, len);
1031
1032             if (nread <= 0) {
1033                 switch (nread) {
1034                 case 0:
1035                     if (conf.verbose) {
1036                         dolog ("Failed to read %ld frames (read zero)\n", len);
1037                     }
1038                     goto exit;
1039
1040                 case -EPIPE:
1041                     if (alsa_recover (alsa->handle)) {
1042                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
1043                         goto exit;
1044                     }
1045                     if (conf.verbose) {
1046                         dolog ("Recovering from capture xrun\n");
1047                     }
1048                     continue;
1049
1050                 case -EAGAIN:
1051                     goto exit;
1052
1053                 default:
1054                     alsa_logerr (
1055                         nread,
1056                         "Failed to read %ld frames from %p\n",
1057                         len,
1058                         src
1059                         );
1060                     goto exit;
1061                 }
1062             }
1063
1064             hw->conv (dst, src, nread, &nominal_volume);
1065
1066             src = advance (src, nread << hwshift);
1067             dst += nread;
1068
1069             read_samples += nread;
1070             len -= nread;
1071         }
1072     }
1073
1074  exit:
1075     hw->wpos = (hw->wpos + read_samples) % hw->samples;
1076     return read_samples;
1077 }
1078
1079 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
1080 {
1081     return audio_pcm_sw_read (sw, buf, size);
1082 }
1083
1084 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
1085 {
1086     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
1087
1088     switch (cmd) {
1089     case VOICE_ENABLE:
1090         {
1091             va_list ap;
1092             int poll_mode;
1093
1094             va_start (ap, cmd);
1095             poll_mode = va_arg (ap, int);
1096             va_end (ap);
1097
1098             ldebug ("enabling voice\n");
1099             if (poll_mode && alsa_poll_in (hw)) {
1100                 poll_mode = 0;
1101             }
1102             hw->poll_mode = poll_mode;
1103
1104             return alsa_voice_ctl (alsa->handle, "capture", 0);
1105         }
1106
1107     case VOICE_DISABLE:
1108         ldebug ("disabling voice\n");
1109         if (hw->poll_mode) {
1110             hw->poll_mode = 0;
1111             alsa_fini_poll (&alsa->pollhlp);
1112         }
1113         return alsa_voice_ctl (alsa->handle, "capture", 1);
1114     }
1115
1116     return -1;
1117 }
1118
1119 static void *alsa_audio_init (void)
1120 {
1121     return &conf;
1122 }
1123
1124 static void alsa_audio_fini (void *opaque)
1125 {
1126     (void) opaque;
1127 }
1128
1129 static struct audio_option alsa_options[] = {
1130     {
1131         .name        = "DAC_SIZE_IN_USEC",
1132         .tag         = AUD_OPT_BOOL,
1133         .valp        = &conf.size_in_usec_out,
1134         .descr       = "DAC period/buffer size in microseconds (otherwise in frames)"
1135     },
1136     {
1137         .name        = "DAC_PERIOD_SIZE",
1138         .tag         = AUD_OPT_INT,
1139         .valp        = &conf.period_size_out,
1140         .descr       = "DAC period size (0 to go with system default)",
1141         .overriddenp = &conf.period_size_out_overridden
1142     },
1143     {
1144         .name        = "DAC_BUFFER_SIZE",
1145         .tag         = AUD_OPT_INT,
1146         .valp        = &conf.buffer_size_out,
1147         .descr       = "DAC buffer size (0 to go with system default)",
1148         .overriddenp = &conf.buffer_size_out_overridden
1149     },
1150     {
1151         .name        = "ADC_SIZE_IN_USEC",
1152         .tag         = AUD_OPT_BOOL,
1153         .valp        = &conf.size_in_usec_in,
1154         .descr       =
1155         "ADC period/buffer size in microseconds (otherwise in frames)"
1156     },
1157     {
1158         .name        = "ADC_PERIOD_SIZE",
1159         .tag         = AUD_OPT_INT,
1160         .valp        = &conf.period_size_in,
1161         .descr       = "ADC period size (0 to go with system default)",
1162         .overriddenp = &conf.period_size_in_overridden
1163     },
1164     {
1165         .name        = "ADC_BUFFER_SIZE",
1166         .tag         = AUD_OPT_INT,
1167         .valp        = &conf.buffer_size_in,
1168         .descr       = "ADC buffer size (0 to go with system default)",
1169         .overriddenp = &conf.buffer_size_in_overridden
1170     },
1171     {
1172         .name        = "THRESHOLD",
1173         .tag         = AUD_OPT_INT,
1174         .valp        = &conf.threshold,
1175         .descr       = "(undocumented)"
1176     },
1177     {
1178         .name        = "DAC_DEV",
1179         .tag         = AUD_OPT_STR,
1180         .valp        = &conf.pcm_name_out,
1181         .descr       = "DAC device name (for instance dmix)"
1182     },
1183     {
1184         .name        = "ADC_DEV",
1185         .tag         = AUD_OPT_STR,
1186         .valp        = &conf.pcm_name_in,
1187         .descr       = "ADC device name"
1188     },
1189     {
1190         .name        = "VERBOSE",
1191         .tag         = AUD_OPT_BOOL,
1192         .valp        = &conf.verbose,
1193         .descr       = "Behave in a more verbose way"
1194     },
1195     { /* End of list */ }
1196 };
1197
1198 static struct audio_pcm_ops alsa_pcm_ops = {
1199     .init_out = alsa_init_out,
1200     .fini_out = alsa_fini_out,
1201     .run_out  = alsa_run_out,
1202     .write    = alsa_write,
1203     .ctl_out  = alsa_ctl_out,
1204
1205     .init_in  = alsa_init_in,
1206     .fini_in  = alsa_fini_in,
1207     .run_in   = alsa_run_in,
1208     .read     = alsa_read,
1209     .ctl_in   = alsa_ctl_in,
1210 };
1211
1212 struct audio_driver alsa_audio_driver = {
1213     .name           = "alsa",
1214     .descr          = "ALSA http://www.alsa-project.org",
1215     .options        = alsa_options,
1216     .init           = alsa_audio_init,
1217     .fini           = alsa_audio_fini,
1218     .pcm_ops        = &alsa_pcm_ops,
1219     .can_be_default = 1,
1220     .max_voices_out = INT_MAX,
1221     .max_voices_in  = INT_MAX,
1222     .voice_size_out = sizeof (ALSAVoiceOut),
1223     .voice_size_in  = sizeof (ALSAVoiceIn)
1224 };