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 #if QEMU_GNUC_PREREQ(4, 3)
  29 #pragma GCC diagnostic ignored "-Waddress"
  30 #endif
  31
  32 #define AUDIO_CAP "alsa"
  33 #include "audio_int.h"
  34
  35 typedef struct ALSAVoiceOut {
  36     HWVoiceOut hw;
  37     void *pcm_buf;
  38     snd_pcm_t *handle;
  39 } ALSAVoiceOut;
  40
  41 typedef struct ALSAVoiceIn {
  42     HWVoiceIn hw;
  43     snd_pcm_t *handle;
  44     void *pcm_buf;
  45 } ALSAVoiceIn;
  46
  47 static struct {
  48     int size_in_usec_in;
  49     int size_in_usec_out;
  50     const char *pcm_name_in;
  51     const char *pcm_name_out;
  52     unsigned int buffer_size_in;
  53     unsigned int period_size_in;
  54     unsigned int buffer_size_out;
  55     unsigned int period_size_out;
  56     unsigned int threshold;
  57
  58     int buffer_size_in_overridden;
  59     int period_size_in_overridden;
  60
  61     int buffer_size_out_overridden;
  62     int period_size_out_overridden;
  63     int verbose;
  64 } conf = {
  65     .buffer_size_out = 1024,
  66     .pcm_name_out = "default",
  67     .pcm_name_in = "default",
  68 };
  69
  70 struct alsa_params_req {
  71     int freq;
  72     snd_pcm_format_t fmt;
  73     int nchannels;
  74     int size_in_usec;
  75     int override_mask;
  76     unsigned int buffer_size;
  77     unsigned int period_size;
  78 };
  79
  80 struct alsa_params_obt {
  81     int freq;
  82     audfmt_e fmt;
  83     int endianness;
  84     int nchannels;
  85     snd_pcm_uframes_t samples;
  86 };
  87
  88 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
  89 {
  90     va_list ap;
  91
  92     va_start (ap, fmt);
  93     AUD_vlog (AUDIO_CAP, fmt, ap);
  94     va_end (ap);
  95
  96     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
  97 }
  98
  99 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
 100     int err,
 101     const char *typ,
 102     const char *fmt,
 103     ...
 104     )
 105 {
 106     va_list ap;
 107
 108     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
 109
 110     va_start (ap, fmt);
 111     AUD_vlog (AUDIO_CAP, fmt, ap);
 112     va_end (ap);
 113
 114     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 115 }
 116
 117 static void alsa_anal_close (snd_pcm_t **handlep)
 118 {
 119     int err = snd_pcm_close (*handlep);
 120     if (err) {
 121         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 122     }
 123     *handlep = NULL;
 124 }
 125
 126 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
 127 {
 128     return audio_pcm_sw_write (sw, buf, len);
 129 }
 130
 131 static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
 132 {
 133     switch (fmt) {
 134     case AUD_FMT_S8:
 135         return SND_PCM_FORMAT_S8;
 136
 137     case AUD_FMT_U8:
 138         return SND_PCM_FORMAT_U8;
 139
 140     case AUD_FMT_S16:
 141         return SND_PCM_FORMAT_S16_LE;
 142
 143     case AUD_FMT_U16:
 144         return SND_PCM_FORMAT_U16_LE;
 145
 146     case AUD_FMT_S32:
 147         return SND_PCM_FORMAT_S32_LE;
 148
 149     case AUD_FMT_U32:
 150         return SND_PCM_FORMAT_U32_LE;
 151
 152     default:
 153         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 154 #ifdef DEBUG_AUDIO
 155         abort ();
 156 #endif
 157         return SND_PCM_FORMAT_U8;
 158     }
 159 }
 160
 161 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
 162                            int *endianness)
 163 {
 164     switch (alsafmt) {
 165     case SND_PCM_FORMAT_S8:
 166         *endianness = 0;
 167         *fmt = AUD_FMT_S8;
 168         break;
 169
 170     case SND_PCM_FORMAT_U8:
 171         *endianness = 0;
 172         *fmt = AUD_FMT_U8;
 173         break;
 174
 175     case SND_PCM_FORMAT_S16_LE:
 176         *endianness = 0;
 177         *fmt = AUD_FMT_S16;
 178         break;
 179
 180     case SND_PCM_FORMAT_U16_LE:
 181         *endianness = 0;
 182         *fmt = AUD_FMT_U16;
 183         break;
 184
 185     case SND_PCM_FORMAT_S16_BE:
 186         *endianness = 1;
 187         *fmt = AUD_FMT_S16;
 188         break;
 189
 190     case SND_PCM_FORMAT_U16_BE:
 191         *endianness = 1;
 192         *fmt = AUD_FMT_U16;
 193         break;
 194
 195     case SND_PCM_FORMAT_S32_LE:
 196         *endianness = 0;
 197         *fmt = AUD_FMT_S32;
 198         break;
 199
 200     case SND_PCM_FORMAT_U32_LE:
 201         *endianness = 0;
 202         *fmt = AUD_FMT_U32;
 203         break;
 204
 205     case SND_PCM_FORMAT_S32_BE:
 206         *endianness = 1;
 207         *fmt = AUD_FMT_S32;
 208         break;
 209
 210     case SND_PCM_FORMAT_U32_BE:
 211         *endianness = 1;
 212         *fmt = AUD_FMT_U32;
 213         break;
 214
 215     default:
 216         dolog ("Unrecognized audio format %d\n", alsafmt);
 217         return -1;
 218     }
 219
 220     return 0;
 221 }
 222
 223 static void alsa_dump_info (struct alsa_params_req *req,
 224                             struct alsa_params_obt *obt)
 225 {
 226     dolog ("parameter | requested value | obtained value\n");
 227     dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
 228     dolog ("channels  |      %10d |     %10d\n",
 229            req->nchannels, obt->nchannels);
 230     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 231     dolog ("============================================\n");
 232     dolog ("requested: buffer size %d period size %d\n",
 233            req->buffer_size, req->period_size);
 234     dolog ("obtained: samples %ld\n", obt->samples);
 235 }
 236
 237 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 238 {
 239     int err;
 240     snd_pcm_sw_params_t *sw_params;
 241
 242     snd_pcm_sw_params_alloca (&sw_params);
 243
 244     err = snd_pcm_sw_params_current (handle, sw_params);
 245     if (err < 0) {
 246         dolog ("Could not fully initialize DAC\n");
 247         alsa_logerr (err, "Failed to get current software parameters\n");
 248         return;
 249     }
 250
 251     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 252     if (err < 0) {
 253         dolog ("Could not fully initialize DAC\n");
 254         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 255                      threshold);
 256         return;
 257     }
 258
 259     err = snd_pcm_sw_params (handle, sw_params);
 260     if (err < 0) {
 261         dolog ("Could not fully initialize DAC\n");
 262         alsa_logerr (err, "Failed to set software parameters\n");
 263         return;
 264     }
 265 }
 266
 267 static int alsa_open (int in, struct alsa_params_req *req,
 268                       struct alsa_params_obt *obt, snd_pcm_t **handlep)
 269 {
 270     snd_pcm_t *handle;
 271     snd_pcm_hw_params_t *hw_params;
 272     int err;
 273     int size_in_usec;
 274     unsigned int freq, nchannels;
 275     const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
 276     snd_pcm_uframes_t obt_buffer_size;
 277     const char *typ = in ? "ADC" : "DAC";
 278     snd_pcm_format_t obtfmt;
 279
 280     freq = req->freq;
 281     nchannels = req->nchannels;
 282     size_in_usec = req->size_in_usec;
 283
 284     snd_pcm_hw_params_alloca (&hw_params);
 285
 286     err = snd_pcm_open (
 287         &handle,
 288         pcm_name,
 289         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 290         SND_PCM_NONBLOCK
 291         );
 292     if (err < 0) {
 293         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 294         return -1;
 295     }
 296
 297     err = snd_pcm_hw_params_any (handle, hw_params);
 298     if (err < 0) {
 299         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 300         goto err;
 301     }
 302
 303     err = snd_pcm_hw_params_set_access (
 304         handle,
 305         hw_params,
 306         SND_PCM_ACCESS_RW_INTERLEAVED
 307         );
 308     if (err < 0) {
 309         alsa_logerr2 (err, typ, "Failed to set access type\n");
 310         goto err;
 311     }
 312
 313     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 314     if (err < 0 && conf.verbose) {
 315         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 316     }
 317
 318     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 319     if (err < 0) {
 320         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 321         goto err;
 322     }
 323
 324     err = snd_pcm_hw_params_set_channels_near (
 325         handle,
 326         hw_params,
 327         &nchannels
 328         );
 329     if (err < 0) {
 330         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 331                       req->nchannels);
 332         goto err;
 333     }
 334
 335     if (nchannels != 1 && nchannels != 2) {
 336         alsa_logerr2 (err, typ,
 337                       "Can not handle obtained number of channels %d\n",
 338                       nchannels);
 339         goto err;
 340     }
 341
 342     if (req->buffer_size) {
 343         unsigned long obt;
 344
 345         if (size_in_usec) {
 346             int dir = 0;
 347             unsigned int btime = req->buffer_size;
 348
 349             err = snd_pcm_hw_params_set_buffer_time_near (
 350                 handle,
 351                 hw_params,
 352                 &btime,
 353                 &dir
 354                 );
 355             obt = btime;
 356         }
 357         else {
 358             snd_pcm_uframes_t bsize = req->buffer_size;
 359
 360             err = snd_pcm_hw_params_set_buffer_size_near (
 361                 handle,
 362                 hw_params,
 363                 &bsize
 364                 );
 365             obt = bsize;
 366         }
 367         if (err < 0) {
 368             alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
 369                           size_in_usec ? "time" : "size", req->buffer_size);
 370             goto err;
 371         }
 372
 373         if ((req->override_mask & 2) && (obt - req->buffer_size))
 374             dolog ("Requested buffer %s %u was rejected, using %lu\n",
 375                    size_in_usec ? "time" : "size", req->buffer_size, obt);
 376     }
 377
 378     if (req->period_size) {
 379         unsigned long obt;
 380
 381         if (size_in_usec) {
 382             int dir = 0;
 383             unsigned int ptime = req->period_size;
 384
 385             err = snd_pcm_hw_params_set_period_time_near (
 386                 handle,
 387                 hw_params,
 388                 &ptime,
 389                 &dir
 390                 );
 391             obt = ptime;
 392         }
 393         else {
 394             int dir = 0;
 395             snd_pcm_uframes_t psize = req->period_size;
 396
 397             err = snd_pcm_hw_params_set_period_size_near (
 398                 handle,
 399                 hw_params,
 400                 &psize,
 401                 &dir
 402                 );
 403             obt = psize;
 404         }
 405
 406         if (err < 0) {
 407             alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
 408                           size_in_usec ? "time" : "size", req->period_size);
 409             goto err;
 410         }
 411
 412         if ((req->override_mask & 1) && (obt - req->period_size))
 413             dolog ("Requested period %s %u was rejected, using %lu\n",
 414                    size_in_usec ? "time" : "size", req->period_size, obt);
 415     }
 416
 417     err = snd_pcm_hw_params (handle, hw_params);
 418     if (err < 0) {
 419         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 420         goto err;
 421     }
 422
 423     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 424     if (err < 0) {
 425         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 426         goto err;
 427     }
 428
 429     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 430     if (err < 0) {
 431         alsa_logerr2 (err, typ, "Failed to get format\n");
 432         goto err;
 433     }
 434
 435     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 436         dolog ("Invalid format was returned %d\n", obtfmt);
 437         goto err;
 438     }
 439
 440     err = snd_pcm_prepare (handle);
 441     if (err < 0) {
 442         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 443         goto err;
 444     }
 445
 446     if (!in && conf.threshold) {
 447         snd_pcm_uframes_t threshold;
 448         int bytes_per_sec;
 449
 450         bytes_per_sec = freq << (nchannels == 2);
 451
 452         switch (obt->fmt) {
 453         case AUD_FMT_S8:
 454         case AUD_FMT_U8:
 455             break;
 456
 457         case AUD_FMT_S16:
 458         case AUD_FMT_U16:
 459             bytes_per_sec <<= 1;
 460             break;
 461
 462         case AUD_FMT_S32:
 463         case AUD_FMT_U32:
 464             bytes_per_sec <<= 2;
 465             break;
 466         }
 467
 468         threshold = (conf.threshold * bytes_per_sec) / 1000;
 469         alsa_set_threshold (handle, threshold);
 470     }
 471
 472     obt->nchannels = nchannels;
 473     obt->freq = freq;
 474     obt->samples = obt_buffer_size;
 475
 476     *handlep = handle;
 477
 478     if (conf.verbose &&
 479         (obt->fmt != req->fmt ||
 480          obt->nchannels != req->nchannels ||
 481          obt->freq != req->freq)) {
 482         dolog ("Audio paramters for %s\n", typ);
 483         alsa_dump_info (req, obt);
 484     }
 485
 486 #ifdef DEBUG
 487     alsa_dump_info (req, obt);
 488 #endif
 489     return 0;
 490
 491  err:
 492     alsa_anal_close (&handle);
 493     return -1;
 494 }
 495
 496 static int alsa_recover (snd_pcm_t *handle)
 497 {
 498     int err = snd_pcm_prepare (handle);
 499     if (err < 0) {
 500         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 501         return -1;
 502     }
 503     return 0;
 504 }
 505
 506 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 507 {
 508     snd_pcm_sframes_t avail;
 509
 510     avail = snd_pcm_avail_update (handle);
 511     if (avail < 0) {
 512         if (avail == -EPIPE) {
 513             if (!alsa_recover (handle)) {
 514                 avail = snd_pcm_avail_update (handle);
 515             }
 516         }
 517
 518         if (avail < 0) {
 519             alsa_logerr (avail,
 520                          "Could not obtain number of available frames\n");
 521             return -1;
 522         }
 523     }
 524
 525     return avail;
 526 }
 527
 528 static int alsa_run_out (HWVoiceOut *hw)
 529 {
 530     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 531     int rpos, live, decr;
 532     int samples;
 533     uint8_t *dst;
 534     struct st_sample *src;
 535     snd_pcm_sframes_t avail;
 536
 537     live = audio_pcm_hw_get_live_out (hw);
 538     if (!live) {
 539         return 0;
 540     }
 541
 542     avail = alsa_get_avail (alsa->handle);
 543     if (avail < 0) {
 544         dolog ("Could not get number of available playback frames\n");
 545         return 0;
 546     }
 547
 548     decr = audio_MIN (live, avail);
 549     samples = decr;
 550     rpos = hw->rpos;
 551     while (samples) {
 552         int left_till_end_samples = hw->samples - rpos;
 553         int len = audio_MIN (samples, left_till_end_samples);
 554         snd_pcm_sframes_t written;
 555
 556         src = hw->mix_buf + rpos;
 557         dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
 558
 559         hw->clip (dst, src, len);
 560
 561         while (len) {
 562             written = snd_pcm_writei (alsa->handle, dst, len);
 563
 564             if (written <= 0) {
 565                 switch (written) {
 566                 case 0:
 567                     if (conf.verbose) {
 568                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 569                     }
 570                     goto exit;
 571
 572                 case -EPIPE:
 573                     if (alsa_recover (alsa->handle)) {
 574                         alsa_logerr (written, "Failed to write %d frames\n",
 575                                      len);
 576                         goto exit;
 577                     }
 578                     if (conf.verbose) {
 579                         dolog ("Recovering from playback xrun\n");
 580                     }
 581                     continue;
 582
 583                 case -EAGAIN:
 584                     goto exit;
 585
 586                 default:
 587                     alsa_logerr (written, "Failed to write %d frames to %p\n",
 588                                  len, dst);
 589                     goto exit;
 590                 }
 591             }
 592
 593             rpos = (rpos + written) % hw->samples;
 594             samples -= written;
 595             len -= written;
 596             dst = advance (dst, written << hw->info.shift);
 597             src += written;
 598         }
 599     }
 600
 601  exit:
 602     hw->rpos = rpos;
 603     return decr;
 604 }
 605
 606 static void alsa_fini_out (HWVoiceOut *hw)
 607 {
 608     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 609
 610     ldebug ("alsa_fini\n");
 611     alsa_anal_close (&alsa->handle);
 612
 613     if (alsa->pcm_buf) {
 614         qemu_free (alsa->pcm_buf);
 615         alsa->pcm_buf = NULL;
 616     }
 617 }
 618
 619 static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
 620 {
 621     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 622     struct alsa_params_req req;
 623     struct alsa_params_obt obt;
 624     snd_pcm_t *handle;
 625     struct audsettings obt_as;
 626
 627     req.fmt = aud_to_alsafmt (as->fmt);
 628     req.freq = as->freq;
 629     req.nchannels = as->nchannels;
 630     req.period_size = conf.period_size_out;
 631     req.buffer_size = conf.buffer_size_out;
 632     req.size_in_usec = conf.size_in_usec_out;
 633     req.override_mask =
 634         (conf.period_size_out_overridden ? 1 : 0) |
 635         (conf.buffer_size_out_overridden ? 2 : 0);
 636
 637     if (alsa_open (0, &req, &obt, &handle)) {
 638         return -1;
 639     }
 640
 641     obt_as.freq = obt.freq;
 642     obt_as.nchannels = obt.nchannels;
 643     obt_as.fmt = obt.fmt;
 644     obt_as.endianness = obt.endianness;
 645
 646     audio_pcm_init_info (&hw->info, &obt_as);
 647     hw->samples = obt.samples;
 648
 649     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 650     if (!alsa->pcm_buf) {
 651         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 652                hw->samples, 1 << hw->info.shift);
 653         alsa_anal_close (&handle);
 654         return -1;
 655     }
 656
 657     alsa->handle = handle;
 658     return 0;
 659 }
 660
 661 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
 662 {
 663     int err;
 664
 665     if (pause) {
 666         err = snd_pcm_drop (handle);
 667         if (err < 0) {
 668             alsa_logerr (err, "Could not stop %s\n", typ);
 669             return -1;
 670         }
 671     }
 672     else {
 673         err = snd_pcm_prepare (handle);
 674         if (err < 0) {
 675             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 676             return -1;
 677         }
 678     }
 679
 680     return 0;
 681 }
 682
 683 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 684 {
 685     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 686
 687     switch (cmd) {
 688     case VOICE_ENABLE:
 689         ldebug ("enabling voice\n");
 690         return alsa_voice_ctl (alsa->handle, "playback", 0);
 691
 692     case VOICE_DISABLE:
 693         ldebug ("disabling voice\n");
 694         return alsa_voice_ctl (alsa->handle, "playback", 1);
 695     }
 696
 697     return -1;
 698 }
 699
 700 static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
 701 {
 702     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 703     struct alsa_params_req req;
 704     struct alsa_params_obt obt;
 705     snd_pcm_t *handle;
 706     struct audsettings obt_as;
 707
 708     req.fmt = aud_to_alsafmt (as->fmt);
 709     req.freq = as->freq;
 710     req.nchannels = as->nchannels;
 711     req.period_size = conf.period_size_in;
 712     req.buffer_size = conf.buffer_size_in;
 713     req.size_in_usec = conf.size_in_usec_in;
 714     req.override_mask =
 715         (conf.period_size_in_overridden ? 1 : 0) |
 716         (conf.buffer_size_in_overridden ? 2 : 0);
 717
 718     if (alsa_open (1, &req, &obt, &handle)) {
 719         return -1;
 720     }
 721
 722     obt_as.freq = obt.freq;
 723     obt_as.nchannels = obt.nchannels;
 724     obt_as.fmt = obt.fmt;
 725     obt_as.endianness = obt.endianness;
 726
 727     audio_pcm_init_info (&hw->info, &obt_as);
 728     hw->samples = obt.samples;
 729
 730     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 731     if (!alsa->pcm_buf) {
 732         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 733                hw->samples, 1 << hw->info.shift);
 734         alsa_anal_close (&handle);
 735         return -1;
 736     }
 737
 738     alsa->handle = handle;
 739     return 0;
 740 }
 741
 742 static void alsa_fini_in (HWVoiceIn *hw)
 743 {
 744     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 745
 746     alsa_anal_close (&alsa->handle);
 747
 748     if (alsa->pcm_buf) {
 749         qemu_free (alsa->pcm_buf);
 750         alsa->pcm_buf = NULL;
 751     }
 752 }
 753
 754 static int alsa_run_in (HWVoiceIn *hw)
 755 {
 756     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 757     int hwshift = hw->info.shift;
 758     int i;
 759     int live = audio_pcm_hw_get_live_in (hw);
 760     int dead = hw->samples - live;
 761     int decr;
 762     struct {
 763         int add;
 764         int len;
 765     } bufs[2] = {
 766         { hw->wpos, 0 },
 767         { 0, 0 }
 768     };
 769     snd_pcm_sframes_t avail;
 770     snd_pcm_uframes_t read_samples = 0;
 771
 772     if (!dead) {
 773         return 0;
 774     }
 775
 776     avail = alsa_get_avail (alsa->handle);
 777     if (avail < 0) {
 778         dolog ("Could not get number of captured frames\n");
 779         return 0;
 780     }
 781
 782     if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
 783         avail = hw->samples;
 784     }
 785
 786     decr = audio_MIN (dead, avail);
 787     if (!decr) {
 788         return 0;
 789     }
 790
 791     if (hw->wpos + decr > hw->samples) {
 792         bufs[0].len = (hw->samples - hw->wpos);
 793         bufs[1].len = (decr - (hw->samples - hw->wpos));
 794     }
 795     else {
 796         bufs[0].len = decr;
 797     }
 798
 799     for (i = 0; i < 2; ++i) {
 800         void *src;
 801         struct st_sample *dst;
 802         snd_pcm_sframes_t nread;
 803         snd_pcm_uframes_t len;
 804
 805         len = bufs[i].len;
 806
 807         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
 808         dst = hw->conv_buf + bufs[i].add;
 809
 810         while (len) {
 811             nread = snd_pcm_readi (alsa->handle, src, len);
 812
 813             if (nread <= 0) {
 814                 switch (nread) {
 815                 case 0:
 816                     if (conf.verbose) {
 817                         dolog ("Failed to read %ld frames (read zero)\n", len);
 818                     }
 819                     goto exit;
 820
 821                 case -EPIPE:
 822                     if (alsa_recover (alsa->handle)) {
 823                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
 824                         goto exit;
 825                     }
 826                     if (conf.verbose) {
 827                         dolog ("Recovering from capture xrun\n");
 828                     }
 829                     continue;
 830
 831                 case -EAGAIN:
 832                     goto exit;
 833
 834                 default:
 835                     alsa_logerr (
 836                         nread,
 837                         "Failed to read %ld frames from %p\n",
 838                         len,
 839                         src
 840                         );
 841                     goto exit;
 842                 }
 843             }
 844
 845             hw->conv (dst, src, nread, &nominal_volume);
 846
 847             src = advance (src, nread << hwshift);
 848             dst += nread;
 849
 850             read_samples += nread;
 851             len -= nread;
 852         }
 853     }
 854
 855  exit:
 856     hw->wpos = (hw->wpos + read_samples) % hw->samples;
 857     return read_samples;
 858 }
 859
 860 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
 861 {
 862     return audio_pcm_sw_read (sw, buf, size);
 863 }
 864
 865 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 866 {
 867     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 868
 869     switch (cmd) {
 870     case VOICE_ENABLE:
 871         ldebug ("enabling voice\n");
 872         return alsa_voice_ctl (alsa->handle, "capture", 0);
 873
 874     case VOICE_DISABLE:
 875         ldebug ("disabling voice\n");
 876         return alsa_voice_ctl (alsa->handle, "capture", 1);
 877     }
 878
 879     return -1;
 880 }
 881
 882 static void *alsa_audio_init (void)
 883 {
 884     return &conf;
 885 }
 886
 887 static void alsa_audio_fini (void *opaque)
 888 {
 889     (void) opaque;
 890 }
 891
 892 static struct audio_option alsa_options[] = {
 893     {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
 894      "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 895     {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
 896      "DAC period size (0 to go with system default)",
 897      &conf.period_size_out_overridden, 0},
 898     {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
 899      "DAC buffer size (0 to go with system default)",
 900      &conf.buffer_size_out_overridden, 0},
 901
 902     {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
 903      "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 904     {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
 905      "ADC period size (0 to go with system default)",
 906      &conf.period_size_in_overridden, 0},
 907     {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
 908      "ADC buffer size (0 to go with system default)",
 909      &conf.buffer_size_in_overridden, 0},
 910
 911     {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
 912      "(undocumented)", NULL, 0},
 913
 914     {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
 915      "DAC device name (for instance dmix)", NULL, 0},
 916
 917     {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
 918      "ADC device name", NULL, 0},
 919
 920     {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
 921      "Behave in a more verbose way", NULL, 0},
 922
 923     {NULL, 0, NULL, NULL, NULL, 0}
 924 };
 925
 926 static struct audio_pcm_ops alsa_pcm_ops = {
 927     alsa_init_out,
 928     alsa_fini_out,
 929     alsa_run_out,
 930     alsa_write,
 931     alsa_ctl_out,
 932
 933     alsa_init_in,
 934     alsa_fini_in,
 935     alsa_run_in,
 936     alsa_read,
 937     alsa_ctl_in
 938 };
 939
 940 struct audio_driver alsa_audio_driver = {
 941     INIT_FIELD (name           = ) "alsa",
 942     INIT_FIELD (descr          = ) "ALSA http://www.alsa-project.org",
 943     INIT_FIELD (options        = ) alsa_options,
 944     INIT_FIELD (init           = ) alsa_audio_init,
 945     INIT_FIELD (fini           = ) alsa_audio_fini,
 946     INIT_FIELD (pcm_ops        = ) &alsa_pcm_ops,
 947     INIT_FIELD (can_be_default = ) 1,
 948     INIT_FIELD (max_voices_out = ) INT_MAX,
 949     INIT_FIELD (max_voices_in  = ) INT_MAX,
 950     INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
 951     INIT_FIELD (voice_size_in  = ) sizeof (ALSAVoiceIn)
 952 };