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 "audio.h"
  27
  28 #define AUDIO_CAP "alsa"
  29 #include "audio_int.h"
  30
  31 typedef struct ALSAVoiceOut {
  32     HWVoiceOut hw;
  33     void *pcm_buf;
  34     snd_pcm_t *handle;
  35 } ALSAVoiceOut;
  36
  37 typedef struct ALSAVoiceIn {
  38     HWVoiceIn hw;
  39     snd_pcm_t *handle;
  40     void *pcm_buf;
  41 } ALSAVoiceIn;
  42
  43 static struct {
  44     int size_in_usec_in;
  45     int size_in_usec_out;
  46     const char *pcm_name_in;
  47     const char *pcm_name_out;
  48     unsigned int buffer_size_in;
  49     unsigned int period_size_in;
  50     unsigned int buffer_size_out;
  51     unsigned int period_size_out;
  52     unsigned int threshold;
  53
  54     int buffer_size_in_overridden;
  55     int period_size_in_overridden;
  56
  57     int buffer_size_out_overridden;
  58     int period_size_out_overridden;
  59     int verbose;
  60 } conf = {
  61     .buffer_size_out = 1024,
  62     .pcm_name_out = "default",
  63     .pcm_name_in = "default",
  64 };
  65
  66 struct alsa_params_req {
  67     int freq;
  68     snd_pcm_format_t fmt;
  69     int nchannels;
  70     int size_in_usec;
  71     int override_mask;
  72     unsigned int buffer_size;
  73     unsigned int period_size;
  74 };
  75
  76 struct alsa_params_obt {
  77     int freq;
  78     audfmt_e fmt;
  79     int endianness;
  80     int nchannels;
  81     snd_pcm_uframes_t samples;
  82 };
  83
  84 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
  85 {
  86     va_list ap;
  87
  88     va_start (ap, fmt);
  89     AUD_vlog (AUDIO_CAP, fmt, ap);
  90     va_end (ap);
  91
  92     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
  93 }
  94
  95 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
  96     int err,
  97     const char *typ,
  98     const char *fmt,
  99     ...
 100     )
 101 {
 102     va_list ap;
 103
 104     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
 105
 106     va_start (ap, fmt);
 107     AUD_vlog (AUDIO_CAP, fmt, ap);
 108     va_end (ap);
 109
 110     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 111 }
 112
 113 static void alsa_anal_close (snd_pcm_t **handlep)
 114 {
 115     int err = snd_pcm_close (*handlep);
 116     if (err) {
 117         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 118     }
 119     *handlep = NULL;
 120 }
 121
 122 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
 123 {
 124     return audio_pcm_sw_write (sw, buf, len);
 125 }
 126
 127 static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
 128 {
 129     switch (fmt) {
 130     case AUD_FMT_S8:
 131         return SND_PCM_FORMAT_S8;
 132
 133     case AUD_FMT_U8:
 134         return SND_PCM_FORMAT_U8;
 135
 136     case AUD_FMT_S16:
 137         return SND_PCM_FORMAT_S16_LE;
 138
 139     case AUD_FMT_U16:
 140         return SND_PCM_FORMAT_U16_LE;
 141
 142     case AUD_FMT_S32:
 143         return SND_PCM_FORMAT_S32_LE;
 144
 145     case AUD_FMT_U32:
 146         return SND_PCM_FORMAT_U32_LE;
 147
 148     default:
 149         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 150 #ifdef DEBUG_AUDIO
 151         abort ();
 152 #endif
 153         return SND_PCM_FORMAT_U8;
 154     }
 155 }
 156
 157 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
 158                            int *endianness)
 159 {
 160     switch (alsafmt) {
 161     case SND_PCM_FORMAT_S8:
 162         *endianness = 0;
 163         *fmt = AUD_FMT_S8;
 164         break;
 165
 166     case SND_PCM_FORMAT_U8:
 167         *endianness = 0;
 168         *fmt = AUD_FMT_U8;
 169         break;
 170
 171     case SND_PCM_FORMAT_S16_LE:
 172         *endianness = 0;
 173         *fmt = AUD_FMT_S16;
 174         break;
 175
 176     case SND_PCM_FORMAT_U16_LE:
 177         *endianness = 0;
 178         *fmt = AUD_FMT_U16;
 179         break;
 180
 181     case SND_PCM_FORMAT_S16_BE:
 182         *endianness = 1;
 183         *fmt = AUD_FMT_S16;
 184         break;
 185
 186     case SND_PCM_FORMAT_U16_BE:
 187         *endianness = 1;
 188         *fmt = AUD_FMT_U16;
 189         break;
 190
 191     case SND_PCM_FORMAT_S32_LE:
 192         *endianness = 0;
 193         *fmt = AUD_FMT_S32;
 194         break;
 195
 196     case SND_PCM_FORMAT_U32_LE:
 197         *endianness = 0;
 198         *fmt = AUD_FMT_U32;
 199         break;
 200
 201     case SND_PCM_FORMAT_S32_BE:
 202         *endianness = 1;
 203         *fmt = AUD_FMT_S32;
 204         break;
 205
 206     case SND_PCM_FORMAT_U32_BE:
 207         *endianness = 1;
 208         *fmt = AUD_FMT_U32;
 209         break;
 210
 211     default:
 212         dolog ("Unrecognized audio format %d\n", alsafmt);
 213         return -1;
 214     }
 215
 216     return 0;
 217 }
 218
 219 static void alsa_dump_info (struct alsa_params_req *req,
 220                             struct alsa_params_obt *obt)
 221 {
 222     dolog ("parameter | requested value | obtained value\n");
 223     dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
 224     dolog ("channels  |      %10d |     %10d\n",
 225            req->nchannels, obt->nchannels);
 226     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 227     dolog ("============================================\n");
 228     dolog ("requested: buffer size %d period size %d\n",
 229            req->buffer_size, req->period_size);
 230     dolog ("obtained: samples %ld\n", obt->samples);
 231 }
 232
 233 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 234 {
 235     int err;
 236     snd_pcm_sw_params_t *sw_params;
 237
 238     snd_pcm_sw_params_alloca (&sw_params);
 239
 240     err = snd_pcm_sw_params_current (handle, sw_params);
 241     if (err < 0) {
 242         dolog ("Could not fully initialize DAC\n");
 243         alsa_logerr (err, "Failed to get current software parameters\n");
 244         return;
 245     }
 246
 247     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 248     if (err < 0) {
 249         dolog ("Could not fully initialize DAC\n");
 250         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 251                      threshold);
 252         return;
 253     }
 254
 255     err = snd_pcm_sw_params (handle, sw_params);
 256     if (err < 0) {
 257         dolog ("Could not fully initialize DAC\n");
 258         alsa_logerr (err, "Failed to set software parameters\n");
 259         return;
 260     }
 261 }
 262
 263 static int alsa_open (int in, struct alsa_params_req *req,
 264                       struct alsa_params_obt *obt, snd_pcm_t **handlep)
 265 {
 266     snd_pcm_t *handle;
 267     snd_pcm_hw_params_t *hw_params;
 268     int err;
 269     int size_in_usec;
 270     unsigned int freq, nchannels;
 271     const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
 272     snd_pcm_uframes_t obt_buffer_size;
 273     const char *typ = in ? "ADC" : "DAC";
 274     snd_pcm_format_t obtfmt;
 275
 276     freq = req->freq;
 277     nchannels = req->nchannels;
 278     size_in_usec = req->size_in_usec;
 279
 280     snd_pcm_hw_params_alloca (&hw_params);
 281
 282     err = snd_pcm_open (
 283         &handle,
 284         pcm_name,
 285         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 286         SND_PCM_NONBLOCK
 287         );
 288     if (err < 0) {
 289         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 290         return -1;
 291     }
 292
 293     err = snd_pcm_hw_params_any (handle, hw_params);
 294     if (err < 0) {
 295         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 296         goto err;
 297     }
 298
 299     err = snd_pcm_hw_params_set_access (
 300         handle,
 301         hw_params,
 302         SND_PCM_ACCESS_RW_INTERLEAVED
 303         );
 304     if (err < 0) {
 305         alsa_logerr2 (err, typ, "Failed to set access type\n");
 306         goto err;
 307     }
 308
 309     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 310     if (err < 0 && conf.verbose) {
 311         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 312     }
 313
 314     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 315     if (err < 0) {
 316         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 317         goto err;
 318     }
 319
 320     err = snd_pcm_hw_params_set_channels_near (
 321         handle,
 322         hw_params,
 323         &nchannels
 324         );
 325     if (err < 0) {
 326         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 327                       req->nchannels);
 328         goto err;
 329     }
 330
 331     if (nchannels != 1 && nchannels != 2) {
 332         alsa_logerr2 (err, typ,
 333                       "Can not handle obtained number of channels %d\n",
 334                       nchannels);
 335         goto err;
 336     }
 337
 338     if (req->buffer_size) {
 339         unsigned long obt;
 340
 341         if (size_in_usec) {
 342             int dir = 0;
 343             unsigned int btime = req->buffer_size;
 344
 345             err = snd_pcm_hw_params_set_buffer_time_near (
 346                 handle,
 347                 hw_params,
 348                 &btime,
 349                 &dir
 350                 );
 351             obt = btime;
 352         }
 353         else {
 354             snd_pcm_uframes_t bsize = req->buffer_size;
 355
 356             err = snd_pcm_hw_params_set_buffer_size_near (
 357                 handle,
 358                 hw_params,
 359                 &bsize
 360                 );
 361             obt = bsize;
 362         }
 363         if (err < 0) {
 364             alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
 365                           size_in_usec ? "time" : "size", req->buffer_size);
 366             goto err;
 367         }
 368
 369         if ((req->override_mask & 2) && (obt - req->buffer_size))
 370             dolog ("Requested buffer %s %u was rejected, using %lu\n",
 371                    size_in_usec ? "time" : "size", req->buffer_size, obt);
 372     }
 373
 374     if (req->period_size) {
 375         unsigned long obt;
 376
 377         if (size_in_usec) {
 378             int dir = 0;
 379             unsigned int ptime = req->period_size;
 380
 381             err = snd_pcm_hw_params_set_period_time_near (
 382                 handle,
 383                 hw_params,
 384                 &ptime,
 385                 &dir
 386                 );
 387             obt = ptime;
 388         }
 389         else {
 390             int dir = 0;
 391             snd_pcm_uframes_t psize = req->period_size;
 392
 393             err = snd_pcm_hw_params_set_period_size_near (
 394                 handle,
 395                 hw_params,
 396                 &psize,
 397                 &dir
 398                 );
 399             obt = psize;
 400         }
 401
 402         if (err < 0) {
 403             alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
 404                           size_in_usec ? "time" : "size", req->period_size);
 405             goto err;
 406         }
 407
 408         if ((req->override_mask & 1) && (obt - req->period_size))
 409             dolog ("Requested period %s %u was rejected, using %lu\n",
 410                    size_in_usec ? "time" : "size", req->period_size, obt);
 411     }
 412
 413     err = snd_pcm_hw_params (handle, hw_params);
 414     if (err < 0) {
 415         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 416         goto err;
 417     }
 418
 419     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 420     if (err < 0) {
 421         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 422         goto err;
 423     }
 424
 425     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 426     if (err < 0) {
 427         alsa_logerr2 (err, typ, "Failed to get format\n");
 428         goto err;
 429     }
 430
 431     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 432         dolog ("Invalid format was returned %d\n", obtfmt);
 433         goto err;
 434     }
 435
 436     err = snd_pcm_prepare (handle);
 437     if (err < 0) {
 438         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 439         goto err;
 440     }
 441
 442     if (!in && conf.threshold) {
 443         snd_pcm_uframes_t threshold;
 444         int bytes_per_sec;
 445
 446         bytes_per_sec = freq << (nchannels == 2);
 447
 448         switch (obt->fmt) {
 449         case AUD_FMT_S8:
 450         case AUD_FMT_U8:
 451             break;
 452
 453         case AUD_FMT_S16:
 454         case AUD_FMT_U16:
 455             bytes_per_sec <<= 1;
 456             break;
 457
 458         case AUD_FMT_S32:
 459         case AUD_FMT_U32:
 460             bytes_per_sec <<= 2;
 461             break;
 462         }
 463
 464         threshold = (conf.threshold * bytes_per_sec) / 1000;
 465         alsa_set_threshold (handle, threshold);
 466     }
 467
 468     obt->nchannels = nchannels;
 469     obt->freq = freq;
 470     obt->samples = obt_buffer_size;
 471
 472     *handlep = handle;
 473
 474     if (conf.verbose &&
 475         (obt->fmt != req->fmt ||
 476          obt->nchannels != req->nchannels ||
 477          obt->freq != req->freq)) {
 478         dolog ("Audio paramters for %s\n", typ);
 479         alsa_dump_info (req, obt);
 480     }
 481
 482 #ifdef DEBUG
 483     alsa_dump_info (req, obt);
 484 #endif
 485     return 0;
 486
 487  err:
 488     alsa_anal_close (&handle);
 489     return -1;
 490 }
 491
 492 static int alsa_recover (snd_pcm_t *handle)
 493 {
 494     int err = snd_pcm_prepare (handle);
 495     if (err < 0) {
 496         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 497         return -1;
 498     }
 499     return 0;
 500 }
 501
 502 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 503 {
 504     snd_pcm_sframes_t avail;
 505
 506     avail = snd_pcm_avail_update (handle);
 507     if (avail < 0) {
 508         if (avail == -EPIPE) {
 509             if (!alsa_recover (handle)) {
 510                 avail = snd_pcm_avail_update (handle);
 511             }
 512         }
 513
 514         if (avail < 0) {
 515             alsa_logerr (avail,
 516                          "Could not obtain number of available frames\n");
 517             return -1;
 518         }
 519     }
 520
 521     return avail;
 522 }
 523
 524 static int alsa_run_out (HWVoiceOut *hw)
 525 {
 526     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 527     int rpos, live, decr;
 528     int samples;
 529     uint8_t *dst;
 530     st_sample_t *src;
 531     snd_pcm_sframes_t avail;
 532
 533     live = audio_pcm_hw_get_live_out (hw);
 534     if (!live) {
 535         return 0;
 536     }
 537
 538     avail = alsa_get_avail (alsa->handle);
 539     if (avail < 0) {
 540         dolog ("Could not get number of available playback frames\n");
 541         return 0;
 542     }
 543
 544     decr = audio_MIN (live, avail);
 545     samples = decr;
 546     rpos = hw->rpos;
 547     while (samples) {
 548         int left_till_end_samples = hw->samples - rpos;
 549         int len = audio_MIN (samples, left_till_end_samples);
 550         snd_pcm_sframes_t written;
 551
 552         src = hw->mix_buf + rpos;
 553         dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
 554
 555         hw->clip (dst, src, len);
 556
 557         while (len) {
 558             written = snd_pcm_writei (alsa->handle, dst, len);
 559
 560             if (written <= 0) {
 561                 switch (written) {
 562                 case 0:
 563                     if (conf.verbose) {
 564                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 565                     }
 566                     goto exit;
 567
 568                 case -EPIPE:
 569                     if (alsa_recover (alsa->handle)) {
 570                         alsa_logerr (written, "Failed to write %d frames\n",
 571                                      len);
 572                         goto exit;
 573                     }
 574                     if (conf.verbose) {
 575                         dolog ("Recovering from playback xrun\n");
 576                     }
 577                     continue;
 578
 579                 case -EAGAIN:
 580                     goto exit;
 581
 582                 default:
 583                     alsa_logerr (written, "Failed to write %d frames to %p\n",
 584                                  len, dst);
 585                     goto exit;
 586                 }
 587             }
 588
 589             rpos = (rpos + written) % hw->samples;
 590             samples -= written;
 591             len -= written;
 592             dst = advance (dst, written << hw->info.shift);
 593             src += written;
 594         }
 595     }
 596
 597  exit:
 598     hw->rpos = rpos;
 599     return decr;
 600 }
 601
 602 static void alsa_fini_out (HWVoiceOut *hw)
 603 {
 604     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 605
 606     ldebug ("alsa_fini\n");
 607     alsa_anal_close (&alsa->handle);
 608
 609     if (alsa->pcm_buf) {
 610         qemu_free (alsa->pcm_buf);
 611         alsa->pcm_buf = NULL;
 612     }
 613 }
 614
 615 static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
 616 {
 617     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 618     struct alsa_params_req req;
 619     struct alsa_params_obt obt;
 620     snd_pcm_t *handle;
 621     audsettings_t obt_as;
 622
 623     req.fmt = aud_to_alsafmt (as->fmt);
 624     req.freq = as->freq;
 625     req.nchannels = as->nchannels;
 626     req.period_size = conf.period_size_out;
 627     req.buffer_size = conf.buffer_size_out;
 628     req.size_in_usec = conf.size_in_usec_out;
 629     req.override_mask = !!conf.period_size_out_overridden
 630         | (!!conf.buffer_size_out_overridden << 1);
 631
 632     if (alsa_open (0, &req, &obt, &handle)) {
 633         return -1;
 634     }
 635
 636     obt_as.freq = obt.freq;
 637     obt_as.nchannels = obt.nchannels;
 638     obt_as.fmt = obt.fmt;
 639     obt_as.endianness = obt.endianness;
 640
 641     audio_pcm_init_info (&hw->info, &obt_as);
 642     hw->samples = obt.samples;
 643
 644     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 645     if (!alsa->pcm_buf) {
 646         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 647                hw->samples, 1 << hw->info.shift);
 648         alsa_anal_close (&handle);
 649         return -1;
 650     }
 651
 652     alsa->handle = handle;
 653     return 0;
 654 }
 655
 656 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
 657 {
 658     int err;
 659
 660     if (pause) {
 661         err = snd_pcm_drop (handle);
 662         if (err < 0) {
 663             alsa_logerr (err, "Could not stop %s\n", typ);
 664             return -1;
 665         }
 666     }
 667     else {
 668         err = snd_pcm_prepare (handle);
 669         if (err < 0) {
 670             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 671             return -1;
 672         }
 673     }
 674
 675     return 0;
 676 }
 677
 678 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 679 {
 680     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 681
 682     switch (cmd) {
 683     case VOICE_ENABLE:
 684         ldebug ("enabling voice\n");
 685         return alsa_voice_ctl (alsa->handle, "playback", 0);
 686
 687     case VOICE_DISABLE:
 688         ldebug ("disabling voice\n");
 689         return alsa_voice_ctl (alsa->handle, "playback", 1);
 690     }
 691
 692     return -1;
 693 }
 694
 695 static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
 696 {
 697     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 698     struct alsa_params_req req;
 699     struct alsa_params_obt obt;
 700     snd_pcm_t *handle;
 701     audsettings_t obt_as;
 702
 703     req.fmt = aud_to_alsafmt (as->fmt);
 704     req.freq = as->freq;
 705     req.nchannels = as->nchannels;
 706     req.period_size = conf.period_size_in;
 707     req.buffer_size = conf.buffer_size_in;
 708     req.size_in_usec = conf.size_in_usec_in;
 709     req.override_mask = !!conf.period_size_in_overridden
 710         | (!!conf.buffer_size_in_overridden << 1);
 711
 712     if (alsa_open (1, &req, &obt, &handle)) {
 713         return -1;
 714     }
 715
 716     obt_as.freq = obt.freq;
 717     obt_as.nchannels = obt.nchannels;
 718     obt_as.fmt = obt.fmt;
 719     obt_as.endianness = obt.endianness;
 720
 721     audio_pcm_init_info (&hw->info, &obt_as);
 722     hw->samples = obt.samples;
 723
 724     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 725     if (!alsa->pcm_buf) {
 726         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 727                hw->samples, 1 << hw->info.shift);
 728         alsa_anal_close (&handle);
 729         return -1;
 730     }
 731
 732     alsa->handle = handle;
 733     return 0;
 734 }
 735
 736 static void alsa_fini_in (HWVoiceIn *hw)
 737 {
 738     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 739
 740     alsa_anal_close (&alsa->handle);
 741
 742     if (alsa->pcm_buf) {
 743         qemu_free (alsa->pcm_buf);
 744         alsa->pcm_buf = NULL;
 745     }
 746 }
 747
 748 static int alsa_run_in (HWVoiceIn *hw)
 749 {
 750     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 751     int hwshift = hw->info.shift;
 752     int i;
 753     int live = audio_pcm_hw_get_live_in (hw);
 754     int dead = hw->samples - live;
 755     int decr;
 756     struct {
 757         int add;
 758         int len;
 759     } bufs[2] = {
 760         { hw->wpos, 0 },
 761         { 0, 0 }
 762     };
 763     snd_pcm_sframes_t avail;
 764     snd_pcm_uframes_t read_samples = 0;
 765
 766     if (!dead) {
 767         return 0;
 768     }
 769
 770     avail = alsa_get_avail (alsa->handle);
 771     if (avail < 0) {
 772         dolog ("Could not get number of captured frames\n");
 773         return 0;
 774     }
 775
 776     if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
 777         avail = hw->samples;
 778     }
 779
 780     decr = audio_MIN (dead, avail);
 781     if (!decr) {
 782         return 0;
 783     }
 784
 785     if (hw->wpos + decr > hw->samples) {
 786         bufs[0].len = (hw->samples - hw->wpos);
 787         bufs[1].len = (decr - (hw->samples - hw->wpos));
 788     }
 789     else {
 790         bufs[0].len = decr;
 791     }
 792
 793     for (i = 0; i < 2; ++i) {
 794         void *src;
 795         st_sample_t *dst;
 796         snd_pcm_sframes_t nread;
 797         snd_pcm_uframes_t len;
 798
 799         len = bufs[i].len;
 800
 801         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
 802         dst = hw->conv_buf + bufs[i].add;
 803
 804         while (len) {
 805             nread = snd_pcm_readi (alsa->handle, src, len);
 806
 807             if (nread <= 0) {
 808                 switch (nread) {
 809                 case 0:
 810                     if (conf.verbose) {
 811                         dolog ("Failed to read %ld frames (read zero)\n", len);
 812                     }
 813                     goto exit;
 814
 815                 case -EPIPE:
 816                     if (alsa_recover (alsa->handle)) {
 817                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
 818                         goto exit;
 819                     }
 820                     if (conf.verbose) {
 821                         dolog ("Recovering from capture xrun\n");
 822                     }
 823                     continue;
 824
 825                 case -EAGAIN:
 826                     goto exit;
 827
 828                 default:
 829                     alsa_logerr (
 830                         nread,
 831                         "Failed to read %ld frames from %p\n",
 832                         len,
 833                         src
 834                         );
 835                     goto exit;
 836                 }
 837             }
 838
 839             hw->conv (dst, src, nread, &nominal_volume);
 840
 841             src = advance (src, nread << hwshift);
 842             dst += nread;
 843
 844             read_samples += nread;
 845             len -= nread;
 846         }
 847     }
 848
 849  exit:
 850     hw->wpos = (hw->wpos + read_samples) % hw->samples;
 851     return read_samples;
 852 }
 853
 854 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
 855 {
 856     return audio_pcm_sw_read (sw, buf, size);
 857 }
 858
 859 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 860 {
 861     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 862
 863     switch (cmd) {
 864     case VOICE_ENABLE:
 865         ldebug ("enabling voice\n");
 866         return alsa_voice_ctl (alsa->handle, "capture", 0);
 867
 868     case VOICE_DISABLE:
 869         ldebug ("disabling voice\n");
 870         return alsa_voice_ctl (alsa->handle, "capture", 1);
 871     }
 872
 873     return -1;
 874 }
 875
 876 static void *alsa_audio_init (void)
 877 {
 878     return &conf;
 879 }
 880
 881 static void alsa_audio_fini (void *opaque)
 882 {
 883     (void) opaque;
 884 }
 885
 886 static struct audio_option alsa_options[] = {
 887     {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
 888      "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 889     {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
 890      "DAC period size (0 to go with system default)",
 891      &conf.period_size_out_overridden, 0},
 892     {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
 893      "DAC buffer size (0 to go with system default)",
 894      &conf.buffer_size_out_overridden, 0},
 895
 896     {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
 897      "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 898     {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
 899      "ADC period size (0 to go with system default)",
 900      &conf.period_size_in_overridden, 0},
 901     {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
 902      "ADC buffer size (0 to go with system default)",
 903      &conf.buffer_size_in_overridden, 0},
 904
 905     {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
 906      "(undocumented)", NULL, 0},
 907
 908     {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
 909      "DAC device name (for instance dmix)", NULL, 0},
 910
 911     {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
 912      "ADC device name", NULL, 0},
 913
 914     {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
 915      "Behave in a more verbose way", NULL, 0},
 916
 917     {NULL, 0, NULL, NULL, NULL, 0}
 918 };
 919
 920 static struct audio_pcm_ops alsa_pcm_ops = {
 921     alsa_init_out,
 922     alsa_fini_out,
 923     alsa_run_out,
 924     alsa_write,
 925     alsa_ctl_out,
 926
 927     alsa_init_in,
 928     alsa_fini_in,
 929     alsa_run_in,
 930     alsa_read,
 931     alsa_ctl_in
 932 };
 933
 934 struct audio_driver alsa_audio_driver = {
 935     INIT_FIELD (name           = ) "alsa",
 936     INIT_FIELD (descr          = ) "ALSA http://www.alsa-project.org",
 937     INIT_FIELD (options        = ) alsa_options,
 938     INIT_FIELD (init           = ) alsa_audio_init,
 939     INIT_FIELD (fini           = ) alsa_audio_fini,
 940     INIT_FIELD (pcm_ops        = ) &alsa_pcm_ops,
 941     INIT_FIELD (can_be_default = ) 1,
 942     INIT_FIELD (max_voices_out = ) INT_MAX,
 943     INIT_FIELD (max_voices_in  = ) INT_MAX,
 944     INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
 945     INIT_FIELD (voice_size_in  = ) sizeof (ALSAVoiceIn)
 946 };