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                             snd_pcm_format_t obtfmt)
 416 {
 417     dolog ("parameter | requested value | obtained value\n");
 418     dolog ("format    |      %10d |     %10d\n", req->fmt, obtfmt);
 419     dolog ("channels  |      %10d |     %10d\n",
 420            req->nchannels, obt->nchannels);
 421     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 422     dolog ("============================================\n");
 423     dolog ("requested: buffer size %d period size %d\n",
 424            req->buffer_size, req->period_size);
 425     dolog ("obtained: samples %ld\n", obt->samples);
 426 }
 427
 428 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 429 {
 430     int err;
 431     snd_pcm_sw_params_t *sw_params;
 432
 433     snd_pcm_sw_params_alloca (&sw_params);
 434
 435     err = snd_pcm_sw_params_current (handle, sw_params);
 436     if (err < 0) {
 437         dolog ("Could not fully initialize DAC\n");
 438         alsa_logerr (err, "Failed to get current software parameters\n");
 439         return;
 440     }
 441
 442     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 443     if (err < 0) {
 444         dolog ("Could not fully initialize DAC\n");
 445         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 446                      threshold);
 447         return;
 448     }
 449
 450     err = snd_pcm_sw_params (handle, sw_params);
 451     if (err < 0) {
 452         dolog ("Could not fully initialize DAC\n");
 453         alsa_logerr (err, "Failed to set software parameters\n");
 454         return;
 455     }
 456 }
 457
 458 static int alsa_open (int in, struct alsa_params_req *req,
 459                       struct alsa_params_obt *obt, snd_pcm_t **handlep)
 460 {
 461     snd_pcm_t *handle;
 462     snd_pcm_hw_params_t *hw_params;
 463     int err;
 464     int size_in_usec;
 465     unsigned int freq, nchannels;
 466     const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
 467     snd_pcm_uframes_t obt_buffer_size;
 468     const char *typ = in ? "ADC" : "DAC";
 469     snd_pcm_format_t obtfmt;
 470
 471     freq = req->freq;
 472     nchannels = req->nchannels;
 473     size_in_usec = req->size_in_usec;
 474
 475     snd_pcm_hw_params_alloca (&hw_params);
 476
 477     err = snd_pcm_open (
 478         &handle,
 479         pcm_name,
 480         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 481         SND_PCM_NONBLOCK
 482         );
 483     if (err < 0) {
 484         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 485         return -1;
 486     }
 487
 488     err = snd_pcm_hw_params_any (handle, hw_params);
 489     if (err < 0) {
 490         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 491         goto err;
 492     }
 493
 494     err = snd_pcm_hw_params_set_access (
 495         handle,
 496         hw_params,
 497         SND_PCM_ACCESS_RW_INTERLEAVED
 498         );
 499     if (err < 0) {
 500         alsa_logerr2 (err, typ, "Failed to set access type\n");
 501         goto err;
 502     }
 503
 504     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 505     if (err < 0 && conf.verbose) {
 506         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 507     }
 508
 509     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 510     if (err < 0) {
 511         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 512         goto err;
 513     }
 514
 515     err = snd_pcm_hw_params_set_channels_near (
 516         handle,
 517         hw_params,
 518         &nchannels
 519         );
 520     if (err < 0) {
 521         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 522                       req->nchannels);
 523         goto err;
 524     }
 525
 526     if (nchannels != 1 && nchannels != 2) {
 527         alsa_logerr2 (err, typ,
 528                       "Can not handle obtained number of channels %d\n",
 529                       nchannels);
 530         goto err;
 531     }
 532
 533     if (req->buffer_size) {
 534         unsigned long obt;
 535
 536         if (size_in_usec) {
 537             int dir = 0;
 538             unsigned int btime = req->buffer_size;
 539
 540             err = snd_pcm_hw_params_set_buffer_time_near (
 541                 handle,
 542                 hw_params,
 543                 &btime,
 544                 &dir
 545                 );
 546             obt = btime;
 547         }
 548         else {
 549             snd_pcm_uframes_t bsize = req->buffer_size;
 550
 551             err = snd_pcm_hw_params_set_buffer_size_near (
 552                 handle,
 553                 hw_params,
 554                 &bsize
 555                 );
 556             obt = bsize;
 557         }
 558         if (err < 0) {
 559             alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
 560                           size_in_usec ? "time" : "size", req->buffer_size);
 561             goto err;
 562         }
 563
 564         if ((req->override_mask & 2) && (obt - req->buffer_size))
 565             dolog ("Requested buffer %s %u was rejected, using %lu\n",
 566                    size_in_usec ? "time" : "size", req->buffer_size, obt);
 567     }
 568
 569     if (req->period_size) {
 570         unsigned long obt;
 571
 572         if (size_in_usec) {
 573             int dir = 0;
 574             unsigned int ptime = req->period_size;
 575
 576             err = snd_pcm_hw_params_set_period_time_near (
 577                 handle,
 578                 hw_params,
 579                 &ptime,
 580                 &dir
 581                 );
 582             obt = ptime;
 583         }
 584         else {
 585             int dir = 0;
 586             snd_pcm_uframes_t psize = req->period_size;
 587
 588             err = snd_pcm_hw_params_set_period_size_near (
 589                 handle,
 590                 hw_params,
 591                 &psize,
 592                 &dir
 593                 );
 594             obt = psize;
 595         }
 596
 597         if (err < 0) {
 598             alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
 599                           size_in_usec ? "time" : "size", req->period_size);
 600             goto err;
 601         }
 602
 603         if (((req->override_mask & 1) && (obt - req->period_size)))
 604             dolog ("Requested period %s %u was rejected, using %lu\n",
 605                    size_in_usec ? "time" : "size", req->period_size, obt);
 606     }
 607
 608     err = snd_pcm_hw_params (handle, hw_params);
 609     if (err < 0) {
 610         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 611         goto err;
 612     }
 613
 614     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 615     if (err < 0) {
 616         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 617         goto err;
 618     }
 619
 620     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 621     if (err < 0) {
 622         alsa_logerr2 (err, typ, "Failed to get format\n");
 623         goto err;
 624     }
 625
 626     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 627         dolog ("Invalid format was returned %d\n", obtfmt);
 628         goto err;
 629     }
 630
 631     err = snd_pcm_prepare (handle);
 632     if (err < 0) {
 633         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 634         goto err;
 635     }
 636
 637     if (!in && conf.threshold) {
 638         snd_pcm_uframes_t threshold;
 639         int bytes_per_sec;
 640
 641         bytes_per_sec = freq << (nchannels == 2);
 642
 643         switch (obt->fmt) {
 644         case AUD_FMT_S8:
 645         case AUD_FMT_U8:
 646             break;
 647
 648         case AUD_FMT_S16:
 649         case AUD_FMT_U16:
 650             bytes_per_sec <<= 1;
 651             break;
 652
 653         case AUD_FMT_S32:
 654         case AUD_FMT_U32:
 655             bytes_per_sec <<= 2;
 656             break;
 657         }
 658
 659         threshold = (conf.threshold * bytes_per_sec) / 1000;
 660         alsa_set_threshold (handle, threshold);
 661     }
 662
 663     obt->nchannels = nchannels;
 664     obt->freq = freq;
 665     obt->samples = obt_buffer_size;
 666
 667     *handlep = handle;
 668
 669     if (conf.verbose &&
 670         (obtfmt != req->fmt ||
 671          obt->nchannels != req->nchannels ||
 672          obt->freq != req->freq)) {
 673         dolog ("Audio parameters for %s\n", typ);
 674         alsa_dump_info (req, obt, obtfmt);
 675     }
 676
 677 #ifdef DEBUG
 678     alsa_dump_info (req, obt, obtfmt);
 679 #endif
 680     return 0;
 681
 682  err:
 683     alsa_anal_close1 (&handle);
 684     return -1;
 685 }
 686
 687 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 688 {
 689     snd_pcm_sframes_t avail;
 690
 691     avail = snd_pcm_avail_update (handle);
 692     if (avail < 0) {
 693         if (avail == -EPIPE) {
 694             if (!alsa_recover (handle)) {
 695                 avail = snd_pcm_avail_update (handle);
 696             }
 697         }
 698
 699         if (avail < 0) {
 700             alsa_logerr (avail,
 701                          "Could not obtain number of available frames\n");
 702             return -1;
 703         }
 704     }
 705
 706     return avail;
 707 }
 708
 709 static void alsa_write_pending (ALSAVoiceOut *alsa)
 710 {
 711     HWVoiceOut *hw = &alsa->hw;
 712
 713     while (alsa->pending) {
 714         int left_till_end_samples = hw->samples - alsa->wpos;
 715         int len = audio_MIN (alsa->pending, left_till_end_samples);
 716         char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
 717
 718         while (len) {
 719             snd_pcm_sframes_t written;
 720
 721             written = snd_pcm_writei (alsa->handle, src, len);
 722
 723             if (written <= 0) {
 724                 switch (written) {
 725                 case 0:
 726                     if (conf.verbose) {
 727                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 728                     }
 729                     return;
 730
 731                 case -EPIPE:
 732                     if (alsa_recover (alsa->handle)) {
 733                         alsa_logerr (written, "Failed to write %d frames\n",
 734                                      len);
 735                         return;
 736                     }
 737                     if (conf.verbose) {
 738                         dolog ("Recovering from playback xrun\n");
 739                     }
 740                     continue;
 741
 742                 case -ESTRPIPE:
 743                     /* stream is suspended and waiting for an
 744                        application recovery */
 745                     if (alsa_resume (alsa->handle)) {
 746                         alsa_logerr (written, "Failed to write %d frames\n",
 747                                      len);
 748                         return;
 749                     }
 750                     if (conf.verbose) {
 751                         dolog ("Resuming suspended output stream\n");
 752                     }
 753                     continue;
 754
 755                 case -EAGAIN:
 756                     return;
 757
 758                 default:
 759                     alsa_logerr (written, "Failed to write %d frames from %p\n",
 760                                  len, src);
 761                     return;
 762                 }
 763             }
 764
 765             alsa->wpos = (alsa->wpos + written) % hw->samples;
 766             alsa->pending -= written;
 767             len -= written;
 768         }
 769     }
 770 }
 771
 772 static int alsa_run_out (HWVoiceOut *hw, int live)
 773 {
 774     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 775     int decr;
 776     snd_pcm_sframes_t avail;
 777
 778     avail = alsa_get_avail (alsa->handle);
 779     if (avail < 0) {
 780         dolog ("Could not get number of available playback frames\n");
 781         return 0;
 782     }
 783
 784     decr = audio_MIN (live, avail);
 785     decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
 786     alsa->pending += decr;
 787     alsa_write_pending (alsa);
 788     return decr;
 789 }
 790
 791 static void alsa_fini_out (HWVoiceOut *hw)
 792 {
 793     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 794
 795     ldebug ("alsa_fini\n");
 796     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 797
 798     if (alsa->pcm_buf) {
 799         qemu_free (alsa->pcm_buf);
 800         alsa->pcm_buf = NULL;
 801     }
 802 }
 803
 804 static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
 805 {
 806     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 807     struct alsa_params_req req;
 808     struct alsa_params_obt obt;
 809     snd_pcm_t *handle;
 810     struct audsettings obt_as;
 811
 812     req.fmt = aud_to_alsafmt (as->fmt);
 813     req.freq = as->freq;
 814     req.nchannels = as->nchannels;
 815     req.period_size = conf.period_size_out;
 816     req.buffer_size = conf.buffer_size_out;
 817     req.size_in_usec = conf.size_in_usec_out;
 818     req.override_mask =
 819         (conf.period_size_out_overridden ? 1 : 0) |
 820         (conf.buffer_size_out_overridden ? 2 : 0);
 821
 822     if (alsa_open (0, &req, &obt, &handle)) {
 823         return -1;
 824     }
 825
 826     obt_as.freq = obt.freq;
 827     obt_as.nchannels = obt.nchannels;
 828     obt_as.fmt = obt.fmt;
 829     obt_as.endianness = obt.endianness;
 830
 831     audio_pcm_init_info (&hw->info, &obt_as);
 832     hw->samples = obt.samples;
 833
 834     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 835     if (!alsa->pcm_buf) {
 836         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 837                hw->samples, 1 << hw->info.shift);
 838         alsa_anal_close1 (&handle);
 839         return -1;
 840     }
 841
 842     alsa->handle = handle;
 843     return 0;
 844 }
 845
 846 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
 847 {
 848     int err;
 849
 850     if (pause) {
 851         err = snd_pcm_drop (handle);
 852         if (err < 0) {
 853             alsa_logerr (err, "Could not stop %s\n", typ);
 854             return -1;
 855         }
 856     }
 857     else {
 858         err = snd_pcm_prepare (handle);
 859         if (err < 0) {
 860             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 861             return -1;
 862         }
 863     }
 864
 865     return 0;
 866 }
 867
 868 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 869 {
 870     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 871
 872     switch (cmd) {
 873     case VOICE_ENABLE:
 874         {
 875             va_list ap;
 876             int poll_mode;
 877
 878             va_start (ap, cmd);
 879             poll_mode = va_arg (ap, int);
 880             va_end (ap);
 881
 882             ldebug ("enabling voice\n");
 883             if (poll_mode && alsa_poll_out (hw)) {
 884                 poll_mode = 0;
 885             }
 886             hw->poll_mode = poll_mode;
 887             return alsa_voice_ctl (alsa->handle, "playback", 0);
 888         }
 889
 890     case VOICE_DISABLE:
 891         ldebug ("disabling voice\n");
 892         if (hw->poll_mode) {
 893             hw->poll_mode = 0;
 894             alsa_fini_poll (&alsa->pollhlp);
 895         }
 896         return alsa_voice_ctl (alsa->handle, "playback", 1);
 897     }
 898
 899     return -1;
 900 }
 901
 902 static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
 903 {
 904     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 905     struct alsa_params_req req;
 906     struct alsa_params_obt obt;
 907     snd_pcm_t *handle;
 908     struct audsettings obt_as;
 909
 910     req.fmt = aud_to_alsafmt (as->fmt);
 911     req.freq = as->freq;
 912     req.nchannels = as->nchannels;
 913     req.period_size = conf.period_size_in;
 914     req.buffer_size = conf.buffer_size_in;
 915     req.size_in_usec = conf.size_in_usec_in;
 916     req.override_mask =
 917         (conf.period_size_in_overridden ? 1 : 0) |
 918         (conf.buffer_size_in_overridden ? 2 : 0);
 919
 920     if (alsa_open (1, &req, &obt, &handle)) {
 921         return -1;
 922     }
 923
 924     obt_as.freq = obt.freq;
 925     obt_as.nchannels = obt.nchannels;
 926     obt_as.fmt = obt.fmt;
 927     obt_as.endianness = obt.endianness;
 928
 929     audio_pcm_init_info (&hw->info, &obt_as);
 930     hw->samples = obt.samples;
 931
 932     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 933     if (!alsa->pcm_buf) {
 934         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 935                hw->samples, 1 << hw->info.shift);
 936         alsa_anal_close1 (&handle);
 937         return -1;
 938     }
 939
 940     alsa->handle = handle;
 941     return 0;
 942 }
 943
 944 static void alsa_fini_in (HWVoiceIn *hw)
 945 {
 946     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 947
 948     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 949
 950     if (alsa->pcm_buf) {
 951         qemu_free (alsa->pcm_buf);
 952         alsa->pcm_buf = NULL;
 953     }
 954 }
 955
 956 static int alsa_run_in (HWVoiceIn *hw)
 957 {
 958     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 959     int hwshift = hw->info.shift;
 960     int i;
 961     int live = audio_pcm_hw_get_live_in (hw);
 962     int dead = hw->samples - live;
 963     int decr;
 964     struct {
 965         int add;
 966         int len;
 967     } bufs[2] = {
 968         { .add = hw->wpos, .len = 0 },
 969         { .add = 0,        .len = 0 }
 970     };
 971     snd_pcm_sframes_t avail;
 972     snd_pcm_uframes_t read_samples = 0;
 973
 974     if (!dead) {
 975         return 0;
 976     }
 977
 978     avail = alsa_get_avail (alsa->handle);
 979     if (avail < 0) {
 980         dolog ("Could not get number of captured frames\n");
 981         return 0;
 982     }
 983
 984     if (!avail) {
 985         snd_pcm_state_t state;
 986
 987         state = snd_pcm_state (alsa->handle);
 988         switch (state) {
 989         case SND_PCM_STATE_PREPARED:
 990             avail = hw->samples;
 991             break;
 992         case SND_PCM_STATE_SUSPENDED:
 993             /* stream is suspended and waiting for an application recovery */
 994             if (alsa_resume (alsa->handle)) {
 995                 dolog ("Failed to resume suspended input stream\n");
 996                 return 0;
 997             }
 998             if (conf.verbose) {
 999                 dolog ("Resuming suspended input stream\n");
1000             }
1001             break;
1002         default:
1003             if (conf.verbose) {
1004                 dolog ("No frames available and ALSA state is %d\n", state);
1005             }
1006             return 0;
1007         }
1008     }
1009
1010     decr = audio_MIN (dead, avail);
1011     if (!decr) {
1012         return 0;
1013     }
1014
1015     if (hw->wpos + decr > hw->samples) {
1016         bufs[0].len = (hw->samples - hw->wpos);
1017         bufs[1].len = (decr - (hw->samples - hw->wpos));
1018     }
1019     else {
1020         bufs[0].len = decr;
1021     }
1022
1023     for (i = 0; i < 2; ++i) {
1024         void *src;
1025         struct st_sample *dst;
1026         snd_pcm_sframes_t nread;
1027         snd_pcm_uframes_t len;
1028
1029         len = bufs[i].len;
1030
1031         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
1032         dst = hw->conv_buf + bufs[i].add;
1033
1034         while (len) {
1035             nread = snd_pcm_readi (alsa->handle, src, len);
1036
1037             if (nread <= 0) {
1038                 switch (nread) {
1039                 case 0:
1040                     if (conf.verbose) {
1041                         dolog ("Failed to read %ld frames (read zero)\n", len);
1042                     }
1043                     goto exit;
1044
1045                 case -EPIPE:
1046                     if (alsa_recover (alsa->handle)) {
1047                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
1048                         goto exit;
1049                     }
1050                     if (conf.verbose) {
1051                         dolog ("Recovering from capture xrun\n");
1052                     }
1053                     continue;
1054
1055                 case -EAGAIN:
1056                     goto exit;
1057
1058                 default:
1059                     alsa_logerr (
1060                         nread,
1061                         "Failed to read %ld frames from %p\n",
1062                         len,
1063                         src
1064                         );
1065                     goto exit;
1066                 }
1067             }
1068
1069             hw->conv (dst, src, nread, &nominal_volume);
1070
1071             src = advance (src, nread << hwshift);
1072             dst += nread;
1073
1074             read_samples += nread;
1075             len -= nread;
1076         }
1077     }
1078
1079  exit:
1080     hw->wpos = (hw->wpos + read_samples) % hw->samples;
1081     return read_samples;
1082 }
1083
1084 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
1085 {
1086     return audio_pcm_sw_read (sw, buf, size);
1087 }
1088
1089 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
1090 {
1091     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
1092
1093     switch (cmd) {
1094     case VOICE_ENABLE:
1095         {
1096             va_list ap;
1097             int poll_mode;
1098
1099             va_start (ap, cmd);
1100             poll_mode = va_arg (ap, int);
1101             va_end (ap);
1102
1103             ldebug ("enabling voice\n");
1104             if (poll_mode && alsa_poll_in (hw)) {
1105                 poll_mode = 0;
1106             }
1107             hw->poll_mode = poll_mode;
1108
1109             return alsa_voice_ctl (alsa->handle, "capture", 0);
1110         }
1111
1112     case VOICE_DISABLE:
1113         ldebug ("disabling voice\n");
1114         if (hw->poll_mode) {
1115             hw->poll_mode = 0;
1116             alsa_fini_poll (&alsa->pollhlp);
1117         }
1118         return alsa_voice_ctl (alsa->handle, "capture", 1);
1119     }
1120
1121     return -1;
1122 }
1123
1124 static void *alsa_audio_init (void)
1125 {
1126     return &conf;
1127 }
1128
1129 static void alsa_audio_fini (void *opaque)
1130 {
1131     (void) opaque;
1132 }
1133
1134 static struct audio_option alsa_options[] = {
1135     {
1136         .name        = "DAC_SIZE_IN_USEC",
1137         .tag         = AUD_OPT_BOOL,
1138         .valp        = &conf.size_in_usec_out,
1139         .descr       = "DAC period/buffer size in microseconds (otherwise in frames)"
1140     },
1141     {
1142         .name        = "DAC_PERIOD_SIZE",
1143         .tag         = AUD_OPT_INT,
1144         .valp        = &conf.period_size_out,
1145         .descr       = "DAC period size (0 to go with system default)",
1146         .overriddenp = &conf.period_size_out_overridden
1147     },
1148     {
1149         .name        = "DAC_BUFFER_SIZE",
1150         .tag         = AUD_OPT_INT,
1151         .valp        = &conf.buffer_size_out,
1152         .descr       = "DAC buffer size (0 to go with system default)",
1153         .overriddenp = &conf.buffer_size_out_overridden
1154     },
1155     {
1156         .name        = "ADC_SIZE_IN_USEC",
1157         .tag         = AUD_OPT_BOOL,
1158         .valp        = &conf.size_in_usec_in,
1159         .descr       =
1160         "ADC period/buffer size in microseconds (otherwise in frames)"
1161     },
1162     {
1163         .name        = "ADC_PERIOD_SIZE",
1164         .tag         = AUD_OPT_INT,
1165         .valp        = &conf.period_size_in,
1166         .descr       = "ADC period size (0 to go with system default)",
1167         .overriddenp = &conf.period_size_in_overridden
1168     },
1169     {
1170         .name        = "ADC_BUFFER_SIZE",
1171         .tag         = AUD_OPT_INT,
1172         .valp        = &conf.buffer_size_in,
1173         .descr       = "ADC buffer size (0 to go with system default)",
1174         .overriddenp = &conf.buffer_size_in_overridden
1175     },
1176     {
1177         .name        = "THRESHOLD",
1178         .tag         = AUD_OPT_INT,
1179         .valp        = &conf.threshold,
1180         .descr       = "(undocumented)"
1181     },
1182     {
1183         .name        = "DAC_DEV",
1184         .tag         = AUD_OPT_STR,
1185         .valp        = &conf.pcm_name_out,
1186         .descr       = "DAC device name (for instance dmix)"
1187     },
1188     {
1189         .name        = "ADC_DEV",
1190         .tag         = AUD_OPT_STR,
1191         .valp        = &conf.pcm_name_in,
1192         .descr       = "ADC device name"
1193     },
1194     {
1195         .name        = "VERBOSE",
1196         .tag         = AUD_OPT_BOOL,
1197         .valp        = &conf.verbose,
1198         .descr       = "Behave in a more verbose way"
1199     },
1200     { /* End of list */ }
1201 };
1202
1203 static struct audio_pcm_ops alsa_pcm_ops = {
1204     .init_out = alsa_init_out,
1205     .fini_out = alsa_fini_out,
1206     .run_out  = alsa_run_out,
1207     .write    = alsa_write,
1208     .ctl_out  = alsa_ctl_out,
1209
1210     .init_in  = alsa_init_in,
1211     .fini_in  = alsa_fini_in,
1212     .run_in   = alsa_run_in,
1213     .read     = alsa_read,
1214     .ctl_in   = alsa_ctl_in,
1215 };
1216
1217 struct audio_driver alsa_audio_driver = {
1218     .name           = "alsa",
1219     .descr          = "ALSA http://www.alsa-project.org",
1220     .options        = alsa_options,
1221     .init           = alsa_audio_init,
1222     .fini           = alsa_audio_fini,
1223     .pcm_ops        = &alsa_pcm_ops,
1224     .can_be_default = 1,
1225     .max_voices_out = INT_MAX,
1226     .max_voices_in  = INT_MAX,
1227     .voice_size_out = sizeof (ALSAVoiceOut),
1228     .voice_size_in  = sizeof (ALSAVoiceIn)
1229 };