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 "vl.h"
  26
  27 #define AUDIO_CAP "alsa"
  28 #include "audio_int.h"
  29
  30 typedef struct ALSAVoiceOut {
  31     HWVoiceOut hw;
  32     void *pcm_buf;
  33     snd_pcm_t *handle;
  34 } ALSAVoiceOut;
  35
  36 typedef struct ALSAVoiceIn {
  37     HWVoiceIn hw;
  38     snd_pcm_t *handle;
  39     void *pcm_buf;
  40 } ALSAVoiceIn;
  41
  42 static struct {
  43     int size_in_usec_in;
  44     int size_in_usec_out;
  45     const char *pcm_name_in;
  46     const char *pcm_name_out;
  47     unsigned int buffer_size_in;
  48     unsigned int period_size_in;
  49     unsigned int buffer_size_out;
  50     unsigned int period_size_out;
  51     unsigned int threshold;
  52
  53     int buffer_size_in_overriden;
  54     int period_size_in_overriden;
  55
  56     int buffer_size_out_overriden;
  57     int period_size_out_overriden;
  58     int verbose;
  59 } conf = {
  60 #ifdef HIGH_LATENCY
  61     .size_in_usec_in = 1,
  62     .size_in_usec_out = 1,
  63 #endif
  64     .pcm_name_out = "hw:0,0",
  65     .pcm_name_in = "hw:0,0",
  66 #ifdef HIGH_LATENCY
  67     .buffer_size_in = 400000,
  68     .period_size_in = 400000 / 4,
  69     .buffer_size_out = 400000,
  70     .period_size_out = 400000 / 4,
  71 #else
  72 #define DEFAULT_BUFFER_SIZE 1024
  73 #define DEFAULT_PERIOD_SIZE 256
  74     .buffer_size_in = DEFAULT_BUFFER_SIZE * 4,
  75     .period_size_in = DEFAULT_PERIOD_SIZE * 4,
  76     .buffer_size_out = DEFAULT_BUFFER_SIZE,
  77     .period_size_out = DEFAULT_PERIOD_SIZE,
  78     .buffer_size_in_overriden = 0,
  79     .buffer_size_out_overriden = 0,
  80     .period_size_in_overriden = 0,
  81     .period_size_out_overriden = 0,
  82 #endif
  83     .threshold = 0,
  84     .verbose = 0
  85 };
  86
  87 struct alsa_params_req {
  88     int freq;
  89     audfmt_e fmt;
  90     int nchannels;
  91     unsigned int buffer_size;
  92     unsigned int period_size;
  93 };
  94
  95 struct alsa_params_obt {
  96     int freq;
  97     audfmt_e fmt;
  98     int nchannels;
  99     snd_pcm_uframes_t samples;
 100 };
 101
 102 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
 103 {
 104     va_list ap;
 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 GCC_FMT_ATTR (3, 4) alsa_logerr2 (
 114     int err,
 115     const char *typ,
 116     const char *fmt,
 117     ...
 118     )
 119 {
 120     va_list ap;
 121
 122     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
 123
 124     va_start (ap, fmt);
 125     AUD_vlog (AUDIO_CAP, fmt, ap);
 126     va_end (ap);
 127
 128     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 129 }
 130
 131 static void alsa_anal_close (snd_pcm_t **handlep)
 132 {
 133     int err = snd_pcm_close (*handlep);
 134     if (err) {
 135         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 136     }
 137     *handlep = NULL;
 138 }
 139
 140 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
 141 {
 142     return audio_pcm_sw_write (sw, buf, len);
 143 }
 144
 145 static int aud_to_alsafmt (audfmt_e fmt)
 146 {
 147     switch (fmt) {
 148     case AUD_FMT_S8:
 149         return SND_PCM_FORMAT_S8;
 150
 151     case AUD_FMT_U8:
 152         return SND_PCM_FORMAT_U8;
 153
 154     case AUD_FMT_S16:
 155         return SND_PCM_FORMAT_S16_LE;
 156
 157     case AUD_FMT_U16:
 158         return SND_PCM_FORMAT_U16_LE;
 159
 160     default:
 161         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 162 #ifdef DEBUG_AUDIO
 163         abort ();
 164 #endif
 165         return SND_PCM_FORMAT_U8;
 166     }
 167 }
 168
 169 static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
 170 {
 171     switch (alsafmt) {
 172     case SND_PCM_FORMAT_S8:
 173         *endianness = 0;
 174         *fmt = AUD_FMT_S8;
 175         break;
 176
 177     case SND_PCM_FORMAT_U8:
 178         *endianness = 0;
 179         *fmt = AUD_FMT_U8;
 180         break;
 181
 182     case SND_PCM_FORMAT_S16_LE:
 183         *endianness = 0;
 184         *fmt = AUD_FMT_S16;
 185         break;
 186
 187     case SND_PCM_FORMAT_U16_LE:
 188         *endianness = 0;
 189         *fmt = AUD_FMT_U16;
 190         break;
 191
 192     case SND_PCM_FORMAT_S16_BE:
 193         *endianness = 1;
 194         *fmt = AUD_FMT_S16;
 195         break;
 196
 197     case SND_PCM_FORMAT_U16_BE:
 198         *endianness = 1;
 199         *fmt = AUD_FMT_U16;
 200         break;
 201
 202     default:
 203         dolog ("Unrecognized audio format %d\n", alsafmt);
 204         return -1;
 205     }
 206
 207     return 0;
 208 }
 209
 210 #if defined DEBUG_MISMATCHES || defined DEBUG
 211 static void alsa_dump_info (struct alsa_params_req *req,
 212                             struct alsa_params_obt *obt)
 213 {
 214     dolog ("parameter | requested value | obtained value\n");
 215     dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
 216     dolog ("channels  |      %10d |     %10d\n",
 217            req->nchannels, obt->nchannels);
 218     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 219     dolog ("============================================\n");
 220     dolog ("requested: buffer size %d period size %d\n",
 221            req->buffer_size, req->period_size);
 222     dolog ("obtained: samples %ld\n", obt->samples);
 223 }
 224 #endif
 225
 226 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 227 {
 228     int err;
 229     snd_pcm_sw_params_t *sw_params;
 230
 231     snd_pcm_sw_params_alloca (&sw_params);
 232
 233     err = snd_pcm_sw_params_current (handle, sw_params);
 234     if (err < 0) {
 235         dolog ("Could not fully initialize DAC\n");
 236         alsa_logerr (err, "Failed to get current software parameters\n");
 237         return;
 238     }
 239
 240     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 241     if (err < 0) {
 242         dolog ("Could not fully initialize DAC\n");
 243         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 244                      threshold);
 245         return;
 246     }
 247
 248     err = snd_pcm_sw_params (handle, sw_params);
 249     if (err < 0) {
 250         dolog ("Could not fully initialize DAC\n");
 251         alsa_logerr (err, "Failed to set software parameters\n");
 252         return;
 253     }
 254 }
 255
 256 static int alsa_open (int in, struct alsa_params_req *req,
 257                       struct alsa_params_obt *obt, snd_pcm_t **handlep)
 258 {
 259     snd_pcm_t *handle;
 260     snd_pcm_hw_params_t *hw_params;
 261     int err, freq, nchannels;
 262     const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
 263     unsigned int period_size, buffer_size;
 264     snd_pcm_uframes_t obt_buffer_size;
 265     const char *typ = in ? "ADC" : "DAC";
 266
 267     freq = req->freq;
 268     period_size = req->period_size;
 269     buffer_size = req->buffer_size;
 270     nchannels = req->nchannels;
 271
 272     snd_pcm_hw_params_alloca (&hw_params);
 273
 274     err = snd_pcm_open (
 275         &handle,
 276         pcm_name,
 277         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 278         SND_PCM_NONBLOCK
 279         );
 280     if (err < 0) {
 281         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 282         return -1;
 283     }
 284
 285     err = snd_pcm_hw_params_any (handle, hw_params);
 286     if (err < 0) {
 287         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 288         goto err;
 289     }
 290
 291     err = snd_pcm_hw_params_set_access (
 292         handle,
 293         hw_params,
 294         SND_PCM_ACCESS_RW_INTERLEAVED
 295         );
 296     if (err < 0) {
 297         alsa_logerr2 (err, typ, "Failed to set access type\n");
 298         goto err;
 299     }
 300
 301     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 302     if (err < 0) {
 303         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 304         goto err;
 305     }
 306
 307     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 308     if (err < 0) {
 309         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 310         goto err;
 311     }
 312
 313     err = snd_pcm_hw_params_set_channels_near (
 314         handle,
 315         hw_params,
 316         &nchannels
 317         );
 318     if (err < 0) {
 319         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 320                       req->nchannels);
 321         goto err;
 322     }
 323
 324     if (nchannels != 1 && nchannels != 2) {
 325         alsa_logerr2 (err, typ,
 326                       "Can not handle obtained number of channels %d\n",
 327                       nchannels);
 328         goto err;
 329     }
 330
 331     if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
 332         if (!buffer_size) {
 333             buffer_size = DEFAULT_BUFFER_SIZE;
 334             period_size= DEFAULT_PERIOD_SIZE;
 335         }
 336     }
 337
 338     if (buffer_size) {
 339         if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
 340             if (period_size) {
 341                 err = snd_pcm_hw_params_set_period_time_near (
 342                     handle,
 343                     hw_params,
 344                     &period_size,
 345                     0
 346                     );
 347                 if (err < 0) {
 348                     alsa_logerr2 (err, typ,
 349                                   "Failed to set period time %d\n",
 350                                   req->period_size);
 351                     goto err;
 352                 }
 353             }
 354
 355             err = snd_pcm_hw_params_set_buffer_time_near (
 356                 handle,
 357                 hw_params,
 358                 &buffer_size,
 359                 0
 360                 );
 361
 362             if (err < 0) {
 363                 alsa_logerr2 (err, typ,
 364                               "Failed to set buffer time %d\n",
 365                               req->buffer_size);
 366                 goto err;
 367             }
 368         }
 369         else {
 370             int dir;
 371             snd_pcm_uframes_t minval;
 372
 373             if (period_size) {
 374                 minval = period_size;
 375                 dir = 0;
 376
 377                 err = snd_pcm_hw_params_get_period_size_min (
 378                     hw_params,
 379                     &minval,
 380                     &dir
 381                     );
 382                 if (err < 0) {
 383                     alsa_logerr (
 384                         err,
 385                         "Could not get minmal period size for %s\n",
 386                         typ
 387                         );
 388                 }
 389                 else {
 390                     if (period_size < minval) {
 391                         if ((in && conf.period_size_in_overriden)
 392                             || (!in && conf.period_size_out_overriden)) {
 393                             dolog ("%s period size(%d) is less "
 394                                    "than minmal period size(%ld)\n",
 395                                    typ,
 396                                    period_size,
 397                                    minval);
 398                         }
 399                         period_size = minval;
 400                     }
 401                 }
 402
 403                 err = snd_pcm_hw_params_set_period_size (
 404                     handle,
 405                     hw_params,
 406                     period_size,
 407                     0
 408                     );
 409                 if (err < 0) {
 410                     alsa_logerr2 (err, typ, "Failed to set period size %d\n",
 411                                   req->period_size);
 412                     goto err;
 413                 }
 414             }
 415
 416             minval = buffer_size;
 417             err = snd_pcm_hw_params_get_buffer_size_min (
 418                 hw_params,
 419                 &minval
 420                 );
 421             if (err < 0) {
 422                 alsa_logerr (err, "Could not get minmal buffer size for %s\n",
 423                              typ);
 424             }
 425             else {
 426                 if (buffer_size < minval) {
 427                     if ((in && conf.buffer_size_in_overriden)
 428                         || (!in && conf.buffer_size_out_overriden)) {
 429                         dolog (
 430                             "%s buffer size(%d) is less "
 431                             "than minimal buffer size(%ld)\n",
 432                             typ,
 433                             buffer_size,
 434                             minval
 435                             );
 436                     }
 437                     buffer_size = minval;
 438                 }
 439             }
 440
 441             err = snd_pcm_hw_params_set_buffer_size (
 442                 handle,
 443                 hw_params,
 444                 buffer_size
 445                 );
 446             if (err < 0) {
 447                 alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
 448                               req->buffer_size);
 449                 goto err;
 450             }
 451         }
 452     }
 453     else {
 454         dolog ("warning: Buffer size is not set\n");
 455     }
 456
 457     err = snd_pcm_hw_params (handle, hw_params);
 458     if (err < 0) {
 459         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 460         goto err;
 461     }
 462
 463     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 464     if (err < 0) {
 465         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 466         goto err;
 467     }
 468
 469     err = snd_pcm_prepare (handle);
 470     if (err < 0) {
 471         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 472         goto err;
 473     }
 474
 475     if (!in && conf.threshold) {
 476         snd_pcm_uframes_t threshold;
 477         int bytes_per_sec;
 478
 479         bytes_per_sec = freq
 480             << (nchannels == 2)
 481             << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
 482
 483         threshold = (conf.threshold * bytes_per_sec) / 1000;
 484         alsa_set_threshold (handle, threshold);
 485     }
 486
 487     obt->fmt = req->fmt;
 488     obt->nchannels = nchannels;
 489     obt->freq = freq;
 490     obt->samples = obt_buffer_size;
 491     *handlep = handle;
 492
 493 #if defined DEBUG_MISMATCHES || defined DEBUG
 494     if (obt->fmt != req->fmt ||
 495         obt->nchannels != req->nchannels ||
 496         obt->freq != req->freq) {
 497         dolog ("Audio paramters mismatch for %s\n", typ);
 498         alsa_dump_info (req, obt);
 499     }
 500 #endif
 501
 502 #ifdef DEBUG
 503     alsa_dump_info (req, obt);
 504 #endif
 505     return 0;
 506
 507  err:
 508     alsa_anal_close (&handle);
 509     return -1;
 510 }
 511
 512 static int alsa_recover (snd_pcm_t *handle)
 513 {
 514     int err = snd_pcm_prepare (handle);
 515     if (err < 0) {
 516         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 517         return -1;
 518     }
 519     return 0;
 520 }
 521
 522 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 523 {
 524     snd_pcm_sframes_t avail;
 525
 526     avail = snd_pcm_avail_update (handle);
 527     if (avail < 0) {
 528         if (avail == -EPIPE) {
 529             if (!alsa_recover (handle)) {
 530                 avail = snd_pcm_avail_update (handle);
 531             }
 532         }
 533
 534         if (avail < 0) {
 535             alsa_logerr (avail,
 536                          "Could not obtain number of available frames\n");
 537             return -1;
 538         }
 539     }
 540
 541     return avail;
 542 }
 543
 544 static int alsa_run_out (HWVoiceOut *hw)
 545 {
 546     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 547     int rpos, live, decr;
 548     int samples;
 549     uint8_t *dst;
 550     st_sample_t *src;
 551     snd_pcm_sframes_t avail;
 552
 553     live = audio_pcm_hw_get_live_out (hw);
 554     if (!live) {
 555         return 0;
 556     }
 557
 558     avail = alsa_get_avail (alsa->handle);
 559     if (avail < 0) {
 560         dolog ("Could not get number of available playback frames\n");
 561         return 0;
 562     }
 563
 564     decr = audio_MIN (live, avail);
 565     samples = decr;
 566     rpos = hw->rpos;
 567     while (samples) {
 568         int left_till_end_samples = hw->samples - rpos;
 569         int len = audio_MIN (samples, left_till_end_samples);
 570         snd_pcm_sframes_t written;
 571
 572         src = hw->mix_buf + rpos;
 573         dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
 574
 575         hw->clip (dst, src, len);
 576
 577         while (len) {
 578             written = snd_pcm_writei (alsa->handle, dst, len);
 579
 580             if (written <= 0) {
 581                 switch (written) {
 582                 case 0:
 583                     if (conf.verbose) {
 584                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 585                     }
 586                     goto exit;
 587
 588                 case -EPIPE:
 589                     if (alsa_recover (alsa->handle)) {
 590                         alsa_logerr (written, "Failed to write %d frames\n",
 591                                      len);
 592                         goto exit;
 593                     }
 594                     if (conf.verbose) {
 595                         dolog ("Recovering from playback xrun\n");
 596                     }
 597                     continue;
 598
 599                 case -EAGAIN:
 600                     goto exit;
 601
 602                 default:
 603                     alsa_logerr (written, "Failed to write %d frames to %p\n",
 604                                  len, dst);
 605                     goto exit;
 606                 }
 607             }
 608
 609             mixeng_clear (src, written);
 610             rpos = (rpos + written) % hw->samples;
 611             samples -= written;
 612             len -= written;
 613             dst = advance (dst, written << hw->info.shift);
 614             src += written;
 615         }
 616     }
 617
 618  exit:
 619     hw->rpos = rpos;
 620     return decr;
 621 }
 622
 623 static void alsa_fini_out (HWVoiceOut *hw)
 624 {
 625     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 626
 627     ldebug ("alsa_fini\n");
 628     alsa_anal_close (&alsa->handle);
 629
 630     if (alsa->pcm_buf) {
 631         qemu_free (alsa->pcm_buf);
 632         alsa->pcm_buf = NULL;
 633     }
 634 }
 635
 636 static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
 637 {
 638     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 639     struct alsa_params_req req;
 640     struct alsa_params_obt obt;
 641     audfmt_e effective_fmt;
 642     int endianness;
 643     int err;
 644     snd_pcm_t *handle;
 645     audsettings_t obt_as;
 646
 647     req.fmt = aud_to_alsafmt (as->fmt);
 648     req.freq = as->freq;
 649     req.nchannels = as->nchannels;
 650     req.period_size = conf.period_size_out;
 651     req.buffer_size = conf.buffer_size_out;
 652
 653     if (alsa_open (0, &req, &obt, &handle)) {
 654         return -1;
 655     }
 656
 657     err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
 658     if (err) {
 659         alsa_anal_close (&handle);
 660         return -1;
 661     }
 662
 663     obt_as.freq = obt.freq;
 664     obt_as.nchannels = obt.nchannels;
 665     obt_as.fmt = effective_fmt;
 666
 667     audio_pcm_init_info (
 668         &hw->info,
 669         &obt_as,
 670         audio_need_to_swap_endian (endianness)
 671         );
 672     hw->samples = obt.samples;
 673
 674     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 675     if (!alsa->pcm_buf) {
 676         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 677                hw->samples, 1 << hw->info.shift);
 678         alsa_anal_close (&handle);
 679         return -1;
 680     }
 681
 682     alsa->handle = handle;
 683     return 0;
 684 }
 685
 686 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
 687 {
 688     int err;
 689
 690     if (pause) {
 691         err = snd_pcm_drop (handle);
 692         if (err < 0) {
 693             alsa_logerr (err, "Could not stop %s\n", typ);
 694             return -1;
 695         }
 696     }
 697     else {
 698         err = snd_pcm_prepare (handle);
 699         if (err < 0) {
 700             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 701             return -1;
 702         }
 703     }
 704
 705     return 0;
 706 }
 707
 708 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 709 {
 710     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 711
 712     switch (cmd) {
 713     case VOICE_ENABLE:
 714         ldebug ("enabling voice\n");
 715         return alsa_voice_ctl (alsa->handle, "playback", 0);
 716
 717     case VOICE_DISABLE:
 718         ldebug ("disabling voice\n");
 719         return alsa_voice_ctl (alsa->handle, "playback", 1);
 720     }
 721
 722     return -1;
 723 }
 724
 725 static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
 726 {
 727     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 728     struct alsa_params_req req;
 729     struct alsa_params_obt obt;
 730     int endianness;
 731     int err;
 732     audfmt_e effective_fmt;
 733     snd_pcm_t *handle;
 734     audsettings_t obt_as;
 735
 736     req.fmt = aud_to_alsafmt (as->fmt);
 737     req.freq = as->freq;
 738     req.nchannels = as->nchannels;
 739     req.period_size = conf.period_size_in;
 740     req.buffer_size = conf.buffer_size_in;
 741
 742     if (alsa_open (1, &req, &obt, &handle)) {
 743         return -1;
 744     }
 745
 746     err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
 747     if (err) {
 748         alsa_anal_close (&handle);
 749         return -1;
 750     }
 751
 752     obt_as.freq = obt.freq;
 753     obt_as.nchannels = obt.nchannels;
 754     obt_as.fmt = effective_fmt;
 755
 756     audio_pcm_init_info (
 757         &hw->info,
 758         &obt_as,
 759         audio_need_to_swap_endian (endianness)
 760         );
 761     hw->samples = obt.samples;
 762
 763     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 764     if (!alsa->pcm_buf) {
 765         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 766                hw->samples, 1 << hw->info.shift);
 767         alsa_anal_close (&handle);
 768         return -1;
 769     }
 770
 771     alsa->handle = handle;
 772     return 0;
 773 }
 774
 775 static void alsa_fini_in (HWVoiceIn *hw)
 776 {
 777     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 778
 779     alsa_anal_close (&alsa->handle);
 780
 781     if (alsa->pcm_buf) {
 782         qemu_free (alsa->pcm_buf);
 783         alsa->pcm_buf = NULL;
 784     }
 785 }
 786
 787 static int alsa_run_in (HWVoiceIn *hw)
 788 {
 789     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 790     int hwshift = hw->info.shift;
 791     int i;
 792     int live = audio_pcm_hw_get_live_in (hw);
 793     int dead = hw->samples - live;
 794     int decr;
 795     struct {
 796         int add;
 797         int len;
 798     } bufs[2] = {
 799         { hw->wpos, 0 },
 800         { 0, 0 }
 801     };
 802     snd_pcm_sframes_t avail;
 803     snd_pcm_uframes_t read_samples = 0;
 804
 805     if (!dead) {
 806         return 0;
 807     }
 808
 809     avail = alsa_get_avail (alsa->handle);
 810     if (avail < 0) {
 811         dolog ("Could not get number of captured frames\n");
 812         return 0;
 813     }
 814
 815     if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
 816         avail = hw->samples;
 817     }
 818
 819     decr = audio_MIN (dead, avail);
 820     if (!decr) {
 821         return 0;
 822     }
 823
 824     if (hw->wpos + decr > hw->samples) {
 825         bufs[0].len = (hw->samples - hw->wpos);
 826         bufs[1].len = (decr - (hw->samples - hw->wpos));
 827     }
 828     else {
 829         bufs[0].len = decr;
 830     }
 831
 832     for (i = 0; i < 2; ++i) {
 833         void *src;
 834         st_sample_t *dst;
 835         snd_pcm_sframes_t nread;
 836         snd_pcm_uframes_t len;
 837
 838         len = bufs[i].len;
 839
 840         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
 841         dst = hw->conv_buf + bufs[i].add;
 842
 843         while (len) {
 844             nread = snd_pcm_readi (alsa->handle, src, len);
 845
 846             if (nread <= 0) {
 847                 switch (nread) {
 848                 case 0:
 849                     if (conf.verbose) {
 850                         dolog ("Failed to read %ld frames (read zero)\n", len);
 851                     }
 852                     goto exit;
 853
 854                 case -EPIPE:
 855                     if (alsa_recover (alsa->handle)) {
 856                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
 857                         goto exit;
 858                     }
 859                     if (conf.verbose) {
 860                         dolog ("Recovering from capture xrun\n");
 861                     }
 862                     continue;
 863
 864                 case -EAGAIN:
 865                     goto exit;
 866
 867                 default:
 868                     alsa_logerr (
 869                         nread,
 870                         "Failed to read %ld frames from %p\n",
 871                         len,
 872                         src
 873                         );
 874                     goto exit;
 875                 }
 876             }
 877
 878             hw->conv (dst, src, nread, &nominal_volume);
 879
 880             src = advance (src, nread << hwshift);
 881             dst += nread;
 882
 883             read_samples += nread;
 884             len -= nread;
 885         }
 886     }
 887
 888  exit:
 889     hw->wpos = (hw->wpos + read_samples) % hw->samples;
 890     return read_samples;
 891 }
 892
 893 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
 894 {
 895     return audio_pcm_sw_read (sw, buf, size);
 896 }
 897
 898 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 899 {
 900     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 901
 902     switch (cmd) {
 903     case VOICE_ENABLE:
 904         ldebug ("enabling voice\n");
 905         return alsa_voice_ctl (alsa->handle, "capture", 0);
 906
 907     case VOICE_DISABLE:
 908         ldebug ("disabling voice\n");
 909         return alsa_voice_ctl (alsa->handle, "capture", 1);
 910     }
 911
 912     return -1;
 913 }
 914
 915 static void *alsa_audio_init (void)
 916 {
 917     return &conf;
 918 }
 919
 920 static void alsa_audio_fini (void *opaque)
 921 {
 922     (void) opaque;
 923 }
 924
 925 static struct audio_option alsa_options[] = {
 926     {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
 927      "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 928     {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
 929      "DAC period size", &conf.period_size_out_overriden, 0},
 930     {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
 931      "DAC buffer size", &conf.buffer_size_out_overriden, 0},
 932
 933     {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
 934      "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 935     {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
 936      "ADC period size", &conf.period_size_in_overriden, 0},
 937     {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
 938      "ADC buffer size", &conf.buffer_size_in_overriden, 0},
 939
 940     {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
 941      "(undocumented)", NULL, 0},
 942
 943     {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
 944      "DAC device name (for instance dmix)", NULL, 0},
 945
 946     {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
 947      "ADC device name", NULL, 0},
 948
 949     {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
 950      "Behave in a more verbose way", NULL, 0},
 951
 952     {NULL, 0, NULL, NULL, NULL, 0}
 953 };
 954
 955 static struct audio_pcm_ops alsa_pcm_ops = {
 956     alsa_init_out,
 957     alsa_fini_out,
 958     alsa_run_out,
 959     alsa_write,
 960     alsa_ctl_out,
 961
 962     alsa_init_in,
 963     alsa_fini_in,
 964     alsa_run_in,
 965     alsa_read,
 966     alsa_ctl_in
 967 };
 968
 969 struct audio_driver alsa_audio_driver = {
 970     INIT_FIELD (name           = ) "alsa",
 971     INIT_FIELD (descr          = ) "ALSA http://www.alsa-project.org",
 972     INIT_FIELD (options        = ) alsa_options,
 973     INIT_FIELD (init           = ) alsa_audio_init,
 974     INIT_FIELD (fini           = ) alsa_audio_fini,
 975     INIT_FIELD (pcm_ops        = ) &alsa_pcm_ops,
 976     INIT_FIELD (can_be_default = ) 1,
 977     INIT_FIELD (max_voices_out = ) INT_MAX,
 978     INIT_FIELD (max_voices_in  = ) INT_MAX,
 979     INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
 980     INIT_FIELD (voice_size_in  = ) sizeof (ALSAVoiceIn)
 981 };