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 int alsa_resume (snd_pcm_t *handle)
 507 {
 508     int err = snd_pcm_resume (handle);
 509     if (err < 0) {
 510         alsa_logerr (err, "Failed to resume handle %p\n", handle);
 511         return -1;
 512     }
 513     return 0;
 514 }
 515
 516 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 517 {
 518     snd_pcm_sframes_t avail;
 519
 520     avail = snd_pcm_avail_update (handle);
 521     if (avail < 0) {
 522         if (avail == -EPIPE) {
 523             if (!alsa_recover (handle)) {
 524                 avail = snd_pcm_avail_update (handle);
 525             }
 526         }
 527
 528         if (avail < 0) {
 529             alsa_logerr (avail,
 530                          "Could not obtain number of available frames\n");
 531             return -1;
 532         }
 533     }
 534
 535     return avail;
 536 }
 537
 538 static int alsa_run_out (HWVoiceOut *hw)
 539 {
 540     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 541     int rpos, live, decr;
 542     int samples;
 543     uint8_t *dst;
 544     struct st_sample *src;
 545     snd_pcm_sframes_t avail;
 546
 547     live = audio_pcm_hw_get_live_out (hw);
 548     if (!live) {
 549         return 0;
 550     }
 551
 552     avail = alsa_get_avail (alsa->handle);
 553     if (avail < 0) {
 554         dolog ("Could not get number of available playback frames\n");
 555         return 0;
 556     }
 557
 558     decr = audio_MIN (live, avail);
 559     samples = decr;
 560     rpos = hw->rpos;
 561     while (samples) {
 562         int left_till_end_samples = hw->samples - rpos;
 563         int len = audio_MIN (samples, left_till_end_samples);
 564         snd_pcm_sframes_t written;
 565
 566         src = hw->mix_buf + rpos;
 567         dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
 568
 569         hw->clip (dst, src, len);
 570
 571         while (len) {
 572             written = snd_pcm_writei (alsa->handle, dst, len);
 573
 574             if (written <= 0) {
 575                 switch (written) {
 576                 case 0:
 577                     if (conf.verbose) {
 578                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 579                     }
 580                     goto exit;
 581
 582                 case -EPIPE:
 583                     if (alsa_recover (alsa->handle)) {
 584                         alsa_logerr (written, "Failed to write %d frames\n",
 585                                      len);
 586                         goto exit;
 587                     }
 588                     if (conf.verbose) {
 589                         dolog ("Recovering from playback xrun\n");
 590                     }
 591                     continue;
 592
 593                 case -ESTRPIPE:
 594                     /* stream is suspended and waiting for an
 595                        application recovery */
 596                     if (alsa_resume (alsa->handle)) {
 597                         alsa_logerr (written, "Failed to write %d frames\n",
 598                                      len);
 599                         goto exit;
 600                     }
 601                     if (conf.verbose) {
 602                         dolog ("Resuming suspended output stream\n");
 603                     }
 604                     continue;
 605
 606                 case -EAGAIN:
 607                     goto exit;
 608
 609                 default:
 610                     alsa_logerr (written, "Failed to write %d frames to %p\n",
 611                                  len, dst);
 612                     goto exit;
 613                 }
 614             }
 615
 616             rpos = (rpos + written) % hw->samples;
 617             samples -= written;
 618             len -= written;
 619             dst = advance (dst, written << hw->info.shift);
 620             src += written;
 621         }
 622     }
 623
 624  exit:
 625     hw->rpos = rpos;
 626     return decr;
 627 }
 628
 629 static void alsa_fini_out (HWVoiceOut *hw)
 630 {
 631     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 632
 633     ldebug ("alsa_fini\n");
 634     alsa_anal_close (&alsa->handle);
 635
 636     if (alsa->pcm_buf) {
 637         qemu_free (alsa->pcm_buf);
 638         alsa->pcm_buf = NULL;
 639     }
 640 }
 641
 642 static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
 643 {
 644     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 645     struct alsa_params_req req;
 646     struct alsa_params_obt obt;
 647     snd_pcm_t *handle;
 648     struct audsettings obt_as;
 649
 650     req.fmt = aud_to_alsafmt (as->fmt);
 651     req.freq = as->freq;
 652     req.nchannels = as->nchannels;
 653     req.period_size = conf.period_size_out;
 654     req.buffer_size = conf.buffer_size_out;
 655     req.size_in_usec = conf.size_in_usec_out;
 656     req.override_mask =
 657         (conf.period_size_out_overridden ? 1 : 0) |
 658         (conf.buffer_size_out_overridden ? 2 : 0);
 659
 660     if (alsa_open (0, &req, &obt, &handle)) {
 661         return -1;
 662     }
 663
 664     obt_as.freq = obt.freq;
 665     obt_as.nchannels = obt.nchannels;
 666     obt_as.fmt = obt.fmt;
 667     obt_as.endianness = obt.endianness;
 668
 669     audio_pcm_init_info (&hw->info, &obt_as);
 670     hw->samples = obt.samples;
 671
 672     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 673     if (!alsa->pcm_buf) {
 674         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 675                hw->samples, 1 << hw->info.shift);
 676         alsa_anal_close (&handle);
 677         return -1;
 678     }
 679
 680     alsa->handle = handle;
 681     return 0;
 682 }
 683
 684 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
 685 {
 686     int err;
 687
 688     if (pause) {
 689         err = snd_pcm_drop (handle);
 690         if (err < 0) {
 691             alsa_logerr (err, "Could not stop %s\n", typ);
 692             return -1;
 693         }
 694     }
 695     else {
 696         err = snd_pcm_prepare (handle);
 697         if (err < 0) {
 698             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 699             return -1;
 700         }
 701     }
 702
 703     return 0;
 704 }
 705
 706 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 707 {
 708     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 709
 710     switch (cmd) {
 711     case VOICE_ENABLE:
 712         ldebug ("enabling voice\n");
 713         return alsa_voice_ctl (alsa->handle, "playback", 0);
 714
 715     case VOICE_DISABLE:
 716         ldebug ("disabling voice\n");
 717         return alsa_voice_ctl (alsa->handle, "playback", 1);
 718     }
 719
 720     return -1;
 721 }
 722
 723 static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
 724 {
 725     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 726     struct alsa_params_req req;
 727     struct alsa_params_obt obt;
 728     snd_pcm_t *handle;
 729     struct audsettings obt_as;
 730
 731     req.fmt = aud_to_alsafmt (as->fmt);
 732     req.freq = as->freq;
 733     req.nchannels = as->nchannels;
 734     req.period_size = conf.period_size_in;
 735     req.buffer_size = conf.buffer_size_in;
 736     req.size_in_usec = conf.size_in_usec_in;
 737     req.override_mask =
 738         (conf.period_size_in_overridden ? 1 : 0) |
 739         (conf.buffer_size_in_overridden ? 2 : 0);
 740
 741     if (alsa_open (1, &req, &obt, &handle)) {
 742         return -1;
 743     }
 744
 745     obt_as.freq = obt.freq;
 746     obt_as.nchannels = obt.nchannels;
 747     obt_as.fmt = obt.fmt;
 748     obt_as.endianness = obt.endianness;
 749
 750     audio_pcm_init_info (&hw->info, &obt_as);
 751     hw->samples = obt.samples;
 752
 753     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 754     if (!alsa->pcm_buf) {
 755         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 756                hw->samples, 1 << hw->info.shift);
 757         alsa_anal_close (&handle);
 758         return -1;
 759     }
 760
 761     alsa->handle = handle;
 762     return 0;
 763 }
 764
 765 static void alsa_fini_in (HWVoiceIn *hw)
 766 {
 767     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 768
 769     alsa_anal_close (&alsa->handle);
 770
 771     if (alsa->pcm_buf) {
 772         qemu_free (alsa->pcm_buf);
 773         alsa->pcm_buf = NULL;
 774     }
 775 }
 776
 777 static int alsa_run_in (HWVoiceIn *hw)
 778 {
 779     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 780     int hwshift = hw->info.shift;
 781     int i;
 782     int live = audio_pcm_hw_get_live_in (hw);
 783     int dead = hw->samples - live;
 784     int decr;
 785     struct {
 786         int add;
 787         int len;
 788     } bufs[2] = {
 789         { .add = hw->wpos, .len = 0 },
 790         { .add = 0,        .len = 0 }
 791     };
 792     snd_pcm_sframes_t avail;
 793     snd_pcm_uframes_t read_samples = 0;
 794
 795     if (!dead) {
 796         return 0;
 797     }
 798
 799     avail = alsa_get_avail (alsa->handle);
 800     if (avail < 0) {
 801         dolog ("Could not get number of captured frames\n");
 802         return 0;
 803     }
 804
 805     if (!avail) {
 806         snd_pcm_state_t state;
 807
 808         state = snd_pcm_state (alsa->handle);
 809         switch (state) {
 810         case SND_PCM_STATE_PREPARED:
 811             avail = hw->samples;
 812             break;
 813         case SND_PCM_STATE_SUSPENDED:
 814             /* stream is suspended and waiting for an application recovery */
 815             if (alsa_resume (alsa->handle)) {
 816                 dolog ("Failed to resume suspended input stream\n");
 817                 return 0;
 818             }
 819             if (conf.verbose) {
 820                 dolog ("Resuming suspended input stream\n");
 821             }
 822             break;
 823         default:
 824             if (conf.verbose) {
 825                 dolog ("No frames available and ALSA state is %d\n", state);
 826             }
 827             return 0;
 828         }
 829     }
 830
 831     decr = audio_MIN (dead, avail);
 832     if (!decr) {
 833         return 0;
 834     }
 835
 836     if (hw->wpos + decr > hw->samples) {
 837         bufs[0].len = (hw->samples - hw->wpos);
 838         bufs[1].len = (decr - (hw->samples - hw->wpos));
 839     }
 840     else {
 841         bufs[0].len = decr;
 842     }
 843
 844     for (i = 0; i < 2; ++i) {
 845         void *src;
 846         struct st_sample *dst;
 847         snd_pcm_sframes_t nread;
 848         snd_pcm_uframes_t len;
 849
 850         len = bufs[i].len;
 851
 852         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
 853         dst = hw->conv_buf + bufs[i].add;
 854
 855         while (len) {
 856             nread = snd_pcm_readi (alsa->handle, src, len);
 857
 858             if (nread <= 0) {
 859                 switch (nread) {
 860                 case 0:
 861                     if (conf.verbose) {
 862                         dolog ("Failed to read %ld frames (read zero)\n", len);
 863                     }
 864                     goto exit;
 865
 866                 case -EPIPE:
 867                     if (alsa_recover (alsa->handle)) {
 868                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
 869                         goto exit;
 870                     }
 871                     if (conf.verbose) {
 872                         dolog ("Recovering from capture xrun\n");
 873                     }
 874                     continue;
 875
 876                 case -EAGAIN:
 877                     goto exit;
 878
 879                 default:
 880                     alsa_logerr (
 881                         nread,
 882                         "Failed to read %ld frames from %p\n",
 883                         len,
 884                         src
 885                         );
 886                     goto exit;
 887                 }
 888             }
 889
 890             hw->conv (dst, src, nread, &nominal_volume);
 891
 892             src = advance (src, nread << hwshift);
 893             dst += nread;
 894
 895             read_samples += nread;
 896             len -= nread;
 897         }
 898     }
 899
 900  exit:
 901     hw->wpos = (hw->wpos + read_samples) % hw->samples;
 902     return read_samples;
 903 }
 904
 905 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
 906 {
 907     return audio_pcm_sw_read (sw, buf, size);
 908 }
 909
 910 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 911 {
 912     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 913
 914     switch (cmd) {
 915     case VOICE_ENABLE:
 916         ldebug ("enabling voice\n");
 917         return alsa_voice_ctl (alsa->handle, "capture", 0);
 918
 919     case VOICE_DISABLE:
 920         ldebug ("disabling voice\n");
 921         return alsa_voice_ctl (alsa->handle, "capture", 1);
 922     }
 923
 924     return -1;
 925 }
 926
 927 static void *alsa_audio_init (void)
 928 {
 929     return &conf;
 930 }
 931
 932 static void alsa_audio_fini (void *opaque)
 933 {
 934     (void) opaque;
 935 }
 936
 937 static struct audio_option alsa_options[] = {
 938     {
 939         .name        = "DAC_SIZE_IN_USEC",
 940         .tag         = AUD_OPT_BOOL,
 941         .valp        = &conf.size_in_usec_out,
 942         .descr       = "DAC period/buffer size in microseconds (otherwise in frames)"
 943     },
 944     {
 945         .name        = "DAC_PERIOD_SIZE",
 946         .tag         = AUD_OPT_INT,
 947         .valp        = &conf.period_size_out,
 948         .descr       = "DAC period size (0 to go with system default)",
 949         .overriddenp = &conf.period_size_out_overridden
 950     },
 951     {
 952         .name        = "DAC_BUFFER_SIZE",
 953         .tag         = AUD_OPT_INT,
 954         .valp        = &conf.buffer_size_out,
 955         .descr       = "DAC buffer size (0 to go with system default)",
 956         .overriddenp = &conf.buffer_size_out_overridden
 957     },
 958     {
 959         .name        = "ADC_SIZE_IN_USEC",
 960         .tag         = AUD_OPT_BOOL,
 961         .valp        = &conf.size_in_usec_in,
 962         .descr       =
 963         "ADC period/buffer size in microseconds (otherwise in frames)"
 964     },
 965     {
 966         .name        = "ADC_PERIOD_SIZE",
 967         .tag         = AUD_OPT_INT,
 968         .valp        = &conf.period_size_in,
 969         .descr       = "ADC period size (0 to go with system default)",
 970         .overriddenp = &conf.period_size_in_overridden
 971     },
 972     {
 973         .name        = "ADC_BUFFER_SIZE",
 974         .tag         = AUD_OPT_INT,
 975         .valp        = &conf.buffer_size_in,
 976         .descr       = "ADC buffer size (0 to go with system default)",
 977         .overriddenp = &conf.buffer_size_in_overridden
 978     },
 979     {
 980         .name        = "THRESHOLD",
 981         .tag         = AUD_OPT_INT,
 982         .valp        = &conf.threshold,
 983         .descr       = "(undocumented)"
 984     },
 985     {
 986         .name        = "DAC_DEV",
 987         .tag         = AUD_OPT_STR,
 988         .valp        = &conf.pcm_name_out,
 989         .descr       = "DAC device name (for instance dmix)"
 990     },
 991     {
 992         .name        = "ADC_DEV",
 993         .tag         = AUD_OPT_STR,
 994         .valp        = &conf.pcm_name_in,
 995         .descr       = "ADC device name"
 996     },
 997     {
 998         .name        = "VERBOSE",
 999         .tag         = AUD_OPT_BOOL,
1000         .valp        = &conf.verbose,
1001         .descr       = "Behave in a more verbose way"
1002     },
1003     { /* End of list */ }
1004 };
1005
1006 static struct audio_pcm_ops alsa_pcm_ops = {
1007     .init_out = alsa_init_out,
1008     .fini_out = alsa_fini_out,
1009     .run_out  = alsa_run_out,
1010     .write    = alsa_write,
1011     .ctl_out  = alsa_ctl_out,
1012
1013     .init_in  = alsa_init_in,
1014     .fini_in  = alsa_fini_in,
1015     .run_in   = alsa_run_in,
1016     .read     = alsa_read,
1017     .ctl_in   = alsa_ctl_in
1018 };
1019
1020 struct audio_driver alsa_audio_driver = {
1021     .name           = "alsa",
1022     .descr          = "ALSA http://www.alsa-project.org",
1023     .options        = alsa_options,
1024     .init           = alsa_audio_init,
1025     .fini           = alsa_audio_fini,
1026     .pcm_ops        = &alsa_pcm_ops,
1027     .can_be_default = 1,
1028     .max_voices_out = INT_MAX,
1029     .max_voices_in  = INT_MAX,
1030     .voice_size_out = sizeof (ALSAVoiceOut),
1031     .voice_size_in  = sizeof (ALSAVoiceIn)
1032 };