audio/alsaaudio.c

   1 /*
   2  * QEMU ALSA audio driver
   3  *
   4  * Copyright (c) 2005 Vassili Karpov (malc)
   5  *
   6  * Permission is hereby granted, free of charge, to any person obtaining a copy
   7  * of this software and associated documentation files (the "Software"), to deal
   8  * in the Software without restriction, including without limitation the rights
   9  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  10  * copies of the Software, and to permit persons to whom the Software is
  11  * furnished to do so, subject to the following conditions:
  12  *
  13  * The above copyright notice and this permission notice shall be included in
  14  * all copies or substantial portions of the Software.
  15  *
  16  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  17  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
  18  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
  19  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
  20  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  21  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  22  * THE SOFTWARE.
  23  */
  24 #include <alsa/asoundlib.h>
  25 #include "qemu-common.h"
  26 #include "audio.h"
  27
  28 #define AUDIO_CAP "alsa"
  29 #include "audio_int.h"
  30
  31 typedef struct ALSAVoiceOut {
  32     HWVoiceOut hw;
  33     void *pcm_buf;
  34     snd_pcm_t *handle;
  35 } ALSAVoiceOut;
  36
  37 typedef struct ALSAVoiceIn {
  38     HWVoiceIn hw;
  39     snd_pcm_t *handle;
  40     void *pcm_buf;
  41 } ALSAVoiceIn;
  42
  43 static struct {
  44     int size_in_usec_in;
  45     int size_in_usec_out;
  46     const char *pcm_name_in;
  47     const char *pcm_name_out;
  48     unsigned int buffer_size_in;
  49     unsigned int period_size_in;
  50     unsigned int buffer_size_out;
  51     unsigned int period_size_out;
  52     unsigned int threshold;
  53
  54     int buffer_size_in_overridden;
  55     int period_size_in_overridden;
  56
  57     int buffer_size_out_overridden;
  58     int period_size_out_overridden;
  59     int verbose;
  60 } conf = {
  61     .pcm_name_out = "default",
  62     .pcm_name_in = "default",
  63 };
  64
  65 struct alsa_params_req {
  66     int freq;
  67     snd_pcm_format_t fmt;
  68     int nchannels;
  69     int size_in_usec;
  70     unsigned int buffer_size;
  71     unsigned int period_size;
  72 };
  73
  74 struct alsa_params_obt {
  75     int freq;
  76     audfmt_e fmt;
  77     int endianness;
  78     int nchannels;
  79     snd_pcm_uframes_t samples;
  80 };
  81
  82 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
  83 {
  84     va_list ap;
  85
  86     va_start (ap, fmt);
  87     AUD_vlog (AUDIO_CAP, fmt, ap);
  88     va_end (ap);
  89
  90     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
  91 }
  92
  93 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
  94     int err,
  95     const char *typ,
  96     const char *fmt,
  97     ...
  98     )
  99 {
 100     va_list ap;
 101
 102     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
 103
 104     va_start (ap, fmt);
 105     AUD_vlog (AUDIO_CAP, fmt, ap);
 106     va_end (ap);
 107
 108     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 109 }
 110
 111 static void alsa_anal_close (snd_pcm_t **handlep)
 112 {
 113     int err = snd_pcm_close (*handlep);
 114     if (err) {
 115         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 116     }
 117     *handlep = NULL;
 118 }
 119
 120 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
 121 {
 122     return audio_pcm_sw_write (sw, buf, len);
 123 }
 124
 125 static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
 126 {
 127     switch (fmt) {
 128     case AUD_FMT_S8:
 129         return SND_PCM_FORMAT_S8;
 130
 131     case AUD_FMT_U8:
 132         return SND_PCM_FORMAT_U8;
 133
 134     case AUD_FMT_S16:
 135         return SND_PCM_FORMAT_S16_LE;
 136
 137     case AUD_FMT_U16:
 138         return SND_PCM_FORMAT_U16_LE;
 139
 140     case AUD_FMT_S32:
 141         return SND_PCM_FORMAT_S32_LE;
 142
 143     case AUD_FMT_U32:
 144         return SND_PCM_FORMAT_U32_LE;
 145
 146     default:
 147         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 148 #ifdef DEBUG_AUDIO
 149         abort ();
 150 #endif
 151         return SND_PCM_FORMAT_U8;
 152     }
 153 }
 154
 155 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
 156                            int *endianness)
 157 {
 158     switch (alsafmt) {
 159     case SND_PCM_FORMAT_S8:
 160         *endianness = 0;
 161         *fmt = AUD_FMT_S8;
 162         break;
 163
 164     case SND_PCM_FORMAT_U8:
 165         *endianness = 0;
 166         *fmt = AUD_FMT_U8;
 167         break;
 168
 169     case SND_PCM_FORMAT_S16_LE:
 170         *endianness = 0;
 171         *fmt = AUD_FMT_S16;
 172         break;
 173
 174     case SND_PCM_FORMAT_U16_LE:
 175         *endianness = 0;
 176         *fmt = AUD_FMT_U16;
 177         break;
 178
 179     case SND_PCM_FORMAT_S16_BE:
 180         *endianness = 1;
 181         *fmt = AUD_FMT_S16;
 182         break;
 183
 184     case SND_PCM_FORMAT_U16_BE:
 185         *endianness = 1;
 186         *fmt = AUD_FMT_U16;
 187         break;
 188
 189     case SND_PCM_FORMAT_S32_LE:
 190         *endianness = 0;
 191         *fmt = AUD_FMT_S32;
 192         break;
 193
 194     case SND_PCM_FORMAT_U32_LE:
 195         *endianness = 0;
 196         *fmt = AUD_FMT_U32;
 197         break;
 198
 199     case SND_PCM_FORMAT_S32_BE:
 200         *endianness = 1;
 201         *fmt = AUD_FMT_S32;
 202         break;
 203
 204     case SND_PCM_FORMAT_U32_BE:
 205         *endianness = 1;
 206         *fmt = AUD_FMT_U32;
 207         break;
 208
 209     default:
 210         dolog ("Unrecognized audio format %d\n", alsafmt);
 211         return -1;
 212     }
 213
 214     return 0;
 215 }
 216
 217 static void alsa_dump_info (struct alsa_params_req *req,
 218                             struct alsa_params_obt *obt)
 219 {
 220     dolog ("parameter | requested value | obtained value\n");
 221     dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
 222     dolog ("channels  |      %10d |     %10d\n",
 223            req->nchannels, obt->nchannels);
 224     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 225     dolog ("============================================\n");
 226     dolog ("requested: buffer size %d period size %d\n",
 227            req->buffer_size, req->period_size);
 228     dolog ("obtained: samples %ld\n", obt->samples);
 229 }
 230
 231 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 232 {
 233     int err;
 234     snd_pcm_sw_params_t *sw_params;
 235
 236     snd_pcm_sw_params_alloca (&sw_params);
 237
 238     err = snd_pcm_sw_params_current (handle, sw_params);
 239     if (err < 0) {
 240         dolog ("Could not fully initialize DAC\n");
 241         alsa_logerr (err, "Failed to get current software parameters\n");
 242         return;
 243     }
 244
 245     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 246     if (err < 0) {
 247         dolog ("Could not fully initialize DAC\n");
 248         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 249                      threshold);
 250         return;
 251     }
 252
 253     err = snd_pcm_sw_params (handle, sw_params);
 254     if (err < 0) {
 255         dolog ("Could not fully initialize DAC\n");
 256         alsa_logerr (err, "Failed to set software parameters\n");
 257         return;
 258     }
 259 }
 260
 261 static int alsa_open (int in, struct alsa_params_req *req,
 262                       struct alsa_params_obt *obt, snd_pcm_t **handlep)
 263 {
 264     snd_pcm_t *handle;
 265     snd_pcm_hw_params_t *hw_params;
 266     int err;
 267     int size_in_usec;
 268     unsigned int freq, nchannels;
 269     const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
 270     snd_pcm_uframes_t obt_buffer_size;
 271     const char *typ = in ? "ADC" : "DAC";
 272     snd_pcm_format_t obtfmt;
 273
 274     freq = req->freq;
 275     nchannels = req->nchannels;
 276     size_in_usec = req->size_in_usec;
 277
 278     snd_pcm_hw_params_alloca (&hw_params);
 279
 280     err = snd_pcm_open (
 281         &handle,
 282         pcm_name,
 283         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 284         SND_PCM_NONBLOCK
 285         );
 286     if (err < 0) {
 287         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 288         return -1;
 289     }
 290
 291     err = snd_pcm_hw_params_any (handle, hw_params);
 292     if (err < 0) {
 293         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 294         goto err;
 295     }
 296
 297     err = snd_pcm_hw_params_set_access (
 298         handle,
 299         hw_params,
 300         SND_PCM_ACCESS_RW_INTERLEAVED
 301         );
 302     if (err < 0) {
 303         alsa_logerr2 (err, typ, "Failed to set access type\n");
 304         goto err;
 305     }
 306
 307     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 308     if (err < 0 && conf.verbose) {
 309         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 310     }
 311
 312     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 313     if (err < 0) {
 314         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 315         goto err;
 316     }
 317
 318     err = snd_pcm_hw_params_set_channels_near (
 319         handle,
 320         hw_params,
 321         &nchannels
 322         );
 323     if (err < 0) {
 324         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 325                       req->nchannels);
 326         goto err;
 327     }
 328
 329     if (nchannels != 1 && nchannels != 2) {
 330         alsa_logerr2 (err, typ,
 331                       "Can not handle obtained number of channels %d\n",
 332                       nchannels);
 333         goto err;
 334     }
 335
 336     if (req->buffer_size) {
 337         if (size_in_usec) {
 338             int dir = 0;
 339             unsigned int btime = req->buffer_size;
 340
 341             err = snd_pcm_hw_params_set_buffer_time_near (
 342                 handle,
 343                 hw_params,
 344                 &btime,
 345                 &dir
 346                 );
 347         }
 348         else {
 349             snd_pcm_uframes_t bsize = req->buffer_size;
 350
 351             err = snd_pcm_hw_params_set_buffer_size_near (
 352                 handle,
 353                 hw_params,
 354                 &bsize
 355                 );
 356         }
 357         if (err < 0) {
 358             alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
 359                           size_in_usec ? "time" : "size", req->buffer_size);
 360             goto err;
 361         }
 362     }
 363
 364     if (req->period_size) {
 365         if (size_in_usec) {
 366             int dir = 0;
 367             unsigned int ptime = req->period_size;
 368
 369             err = snd_pcm_hw_params_set_period_time_near (
 370                 handle,
 371                 hw_params,
 372                 &ptime,
 373                 &dir
 374                 );
 375         }
 376         else {
 377             snd_pcm_uframes_t psize = req->period_size;
 378
 379             err = snd_pcm_hw_params_set_buffer_size_near (
 380                 handle,
 381                 hw_params,
 382                 &psize
 383                 );
 384         }
 385
 386         if (err < 0) {
 387             alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
 388                           size_in_usec ? "time" : "size", req->period_size);
 389             goto err;
 390         }
 391     }
 392
 393     err = snd_pcm_hw_params (handle, hw_params);
 394     if (err < 0) {
 395         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 396         goto err;
 397     }
 398
 399     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 400     if (err < 0) {
 401         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 402         goto err;
 403     }
 404
 405     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 406     if (err < 0) {
 407         alsa_logerr2 (err, typ, "Failed to get format\n");
 408         goto err;
 409     }
 410
 411     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 412         dolog ("Invalid format was returned %d\n", obtfmt);
 413         goto err;
 414     }
 415
 416     err = snd_pcm_prepare (handle);
 417     if (err < 0) {
 418         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 419         goto err;
 420     }
 421
 422     if (!in && conf.threshold) {
 423         snd_pcm_uframes_t threshold;
 424         int bytes_per_sec;
 425
 426         bytes_per_sec = freq << (nchannels == 2);
 427
 428         switch (obt->fmt) {
 429         case AUD_FMT_S8:
 430         case AUD_FMT_U8:
 431             break;
 432
 433         case AUD_FMT_S16:
 434         case AUD_FMT_U16:
 435             bytes_per_sec <<= 1;
 436             break;
 437
 438         case AUD_FMT_S32:
 439         case AUD_FMT_U32:
 440             bytes_per_sec <<= 2;
 441             break;
 442         }
 443
 444         threshold = (conf.threshold * bytes_per_sec) / 1000;
 445         alsa_set_threshold (handle, threshold);
 446     }
 447
 448     obt->nchannels = nchannels;
 449     obt->freq = freq;
 450     obt->samples = obt_buffer_size;
 451
 452     *handlep = handle;
 453
 454     if (conf.verbose &&
 455         (obt->fmt != req->fmt ||
 456          obt->nchannels != req->nchannels ||
 457          obt->freq != req->freq)) {
 458         dolog ("Audio paramters for %s\n", typ);
 459         alsa_dump_info (req, obt);
 460     }
 461
 462 #ifdef DEBUG
 463     alsa_dump_info (req, obt);
 464 #endif
 465     return 0;
 466
 467  err:
 468     alsa_anal_close (&handle);
 469     return -1;
 470 }
 471
 472 static int alsa_recover (snd_pcm_t *handle)
 473 {
 474     int err = snd_pcm_prepare (handle);
 475     if (err < 0) {
 476         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 477         return -1;
 478     }
 479     return 0;
 480 }
 481
 482 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 483 {
 484     snd_pcm_sframes_t avail;
 485
 486     avail = snd_pcm_avail_update (handle);
 487     if (avail < 0) {
 488         if (avail == -EPIPE) {
 489             if (!alsa_recover (handle)) {
 490                 avail = snd_pcm_avail_update (handle);
 491             }
 492         }
 493
 494         if (avail < 0) {
 495             alsa_logerr (avail,
 496                          "Could not obtain number of available frames\n");
 497             return -1;
 498         }
 499     }
 500
 501     return avail;
 502 }
 503
 504 static int alsa_run_out (HWVoiceOut *hw)
 505 {
 506     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 507     int rpos, live, decr;
 508     int samples;
 509     uint8_t *dst;
 510     st_sample_t *src;
 511     snd_pcm_sframes_t avail;
 512
 513     live = audio_pcm_hw_get_live_out (hw);
 514     if (!live) {
 515         return 0;
 516     }
 517
 518     avail = alsa_get_avail (alsa->handle);
 519     if (avail < 0) {
 520         dolog ("Could not get number of available playback frames\n");
 521         return 0;
 522     }
 523
 524     decr = audio_MIN (live, avail);
 525     samples = decr;
 526     rpos = hw->rpos;
 527     while (samples) {
 528         int left_till_end_samples = hw->samples - rpos;
 529         int len = audio_MIN (samples, left_till_end_samples);
 530         snd_pcm_sframes_t written;
 531
 532         src = hw->mix_buf + rpos;
 533         dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
 534
 535         hw->clip (dst, src, len);
 536
 537         while (len) {
 538             written = snd_pcm_writei (alsa->handle, dst, len);
 539
 540             if (written <= 0) {
 541                 switch (written) {
 542                 case 0:
 543                     if (conf.verbose) {
 544                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 545                     }
 546                     goto exit;
 547
 548                 case -EPIPE:
 549                     if (alsa_recover (alsa->handle)) {
 550                         alsa_logerr (written, "Failed to write %d frames\n",
 551                                      len);
 552                         goto exit;
 553                     }
 554                     if (conf.verbose) {
 555                         dolog ("Recovering from playback xrun\n");
 556                     }
 557                     continue;
 558
 559                 case -EAGAIN:
 560                     goto exit;
 561
 562                 default:
 563                     alsa_logerr (written, "Failed to write %d frames to %p\n",
 564                                  len, dst);
 565                     goto exit;
 566                 }
 567             }
 568
 569             rpos = (rpos + written) % hw->samples;
 570             samples -= written;
 571             len -= written;
 572             dst = advance (dst, written << hw->info.shift);
 573             src += written;
 574         }
 575     }
 576
 577  exit:
 578     hw->rpos = rpos;
 579     return decr;
 580 }
 581
 582 static void alsa_fini_out (HWVoiceOut *hw)
 583 {
 584     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 585
 586     ldebug ("alsa_fini\n");
 587     alsa_anal_close (&alsa->handle);
 588
 589     if (alsa->pcm_buf) {
 590         qemu_free (alsa->pcm_buf);
 591         alsa->pcm_buf = NULL;
 592     }
 593 }
 594
 595 static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
 596 {
 597     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 598     struct alsa_params_req req;
 599     struct alsa_params_obt obt;
 600     snd_pcm_t *handle;
 601     audsettings_t obt_as;
 602
 603     req.fmt = aud_to_alsafmt (as->fmt);
 604     req.freq = as->freq;
 605     req.nchannels = as->nchannels;
 606     req.period_size = conf.period_size_out;
 607     req.buffer_size = conf.buffer_size_out;
 608     req.size_in_usec = conf.size_in_usec_in;
 609
 610     if (alsa_open (0, &req, &obt, &handle)) {
 611         return -1;
 612     }
 613
 614     obt_as.freq = obt.freq;
 615     obt_as.nchannels = obt.nchannels;
 616     obt_as.fmt = obt.fmt;
 617     obt_as.endianness = obt.endianness;
 618
 619     audio_pcm_init_info (&hw->info, &obt_as);
 620     hw->samples = obt.samples;
 621
 622     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 623     if (!alsa->pcm_buf) {
 624         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 625                hw->samples, 1 << hw->info.shift);
 626         alsa_anal_close (&handle);
 627         return -1;
 628     }
 629
 630     alsa->handle = handle;
 631     return 0;
 632 }
 633
 634 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
 635 {
 636     int err;
 637
 638     if (pause) {
 639         err = snd_pcm_drop (handle);
 640         if (err < 0) {
 641             alsa_logerr (err, "Could not stop %s\n", typ);
 642             return -1;
 643         }
 644     }
 645     else {
 646         err = snd_pcm_prepare (handle);
 647         if (err < 0) {
 648             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 649             return -1;
 650         }
 651     }
 652
 653     return 0;
 654 }
 655
 656 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 657 {
 658     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 659
 660     switch (cmd) {
 661     case VOICE_ENABLE:
 662         ldebug ("enabling voice\n");
 663         return alsa_voice_ctl (alsa->handle, "playback", 0);
 664
 665     case VOICE_DISABLE:
 666         ldebug ("disabling voice\n");
 667         return alsa_voice_ctl (alsa->handle, "playback", 1);
 668     }
 669
 670     return -1;
 671 }
 672
 673 static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
 674 {
 675     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 676     struct alsa_params_req req;
 677     struct alsa_params_obt obt;
 678     snd_pcm_t *handle;
 679     audsettings_t obt_as;
 680
 681     req.fmt = aud_to_alsafmt (as->fmt);
 682     req.freq = as->freq;
 683     req.nchannels = as->nchannels;
 684     req.period_size = conf.period_size_in;
 685     req.buffer_size = conf.buffer_size_in;
 686     req.size_in_usec = conf.size_in_usec_in;
 687
 688     if (alsa_open (1, &req, &obt, &handle)) {
 689         return -1;
 690     }
 691
 692     obt_as.freq = obt.freq;
 693     obt_as.nchannels = obt.nchannels;
 694     obt_as.fmt = obt.fmt;
 695     obt_as.endianness = obt.endianness;
 696
 697     audio_pcm_init_info (&hw->info, &obt_as);
 698     hw->samples = obt.samples;
 699
 700     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 701     if (!alsa->pcm_buf) {
 702         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 703                hw->samples, 1 << hw->info.shift);
 704         alsa_anal_close (&handle);
 705         return -1;
 706     }
 707
 708     alsa->handle = handle;
 709     return 0;
 710 }
 711
 712 static void alsa_fini_in (HWVoiceIn *hw)
 713 {
 714     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 715
 716     alsa_anal_close (&alsa->handle);
 717
 718     if (alsa->pcm_buf) {
 719         qemu_free (alsa->pcm_buf);
 720         alsa->pcm_buf = NULL;
 721     }
 722 }
 723
 724 static int alsa_run_in (HWVoiceIn *hw)
 725 {
 726     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 727     int hwshift = hw->info.shift;
 728     int i;
 729     int live = audio_pcm_hw_get_live_in (hw);
 730     int dead = hw->samples - live;
 731     int decr;
 732     struct {
 733         int add;
 734         int len;
 735     } bufs[2] = {
 736         { hw->wpos, 0 },
 737         { 0, 0 }
 738     };
 739     snd_pcm_sframes_t avail;
 740     snd_pcm_uframes_t read_samples = 0;
 741
 742     if (!dead) {
 743         return 0;
 744     }
 745
 746     avail = alsa_get_avail (alsa->handle);
 747     if (avail < 0) {
 748         dolog ("Could not get number of captured frames\n");
 749         return 0;
 750     }
 751
 752     if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
 753         avail = hw->samples;
 754     }
 755
 756     decr = audio_MIN (dead, avail);
 757     if (!decr) {
 758         return 0;
 759     }
 760
 761     if (hw->wpos + decr > hw->samples) {
 762         bufs[0].len = (hw->samples - hw->wpos);
 763         bufs[1].len = (decr - (hw->samples - hw->wpos));
 764     }
 765     else {
 766         bufs[0].len = decr;
 767     }
 768
 769     for (i = 0; i < 2; ++i) {
 770         void *src;
 771         st_sample_t *dst;
 772         snd_pcm_sframes_t nread;
 773         snd_pcm_uframes_t len;
 774
 775         len = bufs[i].len;
 776
 777         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
 778         dst = hw->conv_buf + bufs[i].add;
 779
 780         while (len) {
 781             nread = snd_pcm_readi (alsa->handle, src, len);
 782
 783             if (nread <= 0) {
 784                 switch (nread) {
 785                 case 0:
 786                     if (conf.verbose) {
 787                         dolog ("Failed to read %ld frames (read zero)\n", len);
 788                     }
 789                     goto exit;
 790
 791                 case -EPIPE:
 792                     if (alsa_recover (alsa->handle)) {
 793                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
 794                         goto exit;
 795                     }
 796                     if (conf.verbose) {
 797                         dolog ("Recovering from capture xrun\n");
 798                     }
 799                     continue;
 800
 801                 case -EAGAIN:
 802                     goto exit;
 803
 804                 default:
 805                     alsa_logerr (
 806                         nread,
 807                         "Failed to read %ld frames from %p\n",
 808                         len,
 809                         src
 810                         );
 811                     goto exit;
 812                 }
 813             }
 814
 815             hw->conv (dst, src, nread, &nominal_volume);
 816
 817             src = advance (src, nread << hwshift);
 818             dst += nread;
 819
 820             read_samples += nread;
 821             len -= nread;
 822         }
 823     }
 824
 825  exit:
 826     hw->wpos = (hw->wpos + read_samples) % hw->samples;
 827     return read_samples;
 828 }
 829
 830 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
 831 {
 832     return audio_pcm_sw_read (sw, buf, size);
 833 }
 834
 835 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 836 {
 837     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 838
 839     switch (cmd) {
 840     case VOICE_ENABLE:
 841         ldebug ("enabling voice\n");
 842         return alsa_voice_ctl (alsa->handle, "capture", 0);
 843
 844     case VOICE_DISABLE:
 845         ldebug ("disabling voice\n");
 846         return alsa_voice_ctl (alsa->handle, "capture", 1);
 847     }
 848
 849     return -1;
 850 }
 851
 852 static void *alsa_audio_init (void)
 853 {
 854     return &conf;
 855 }
 856
 857 static void alsa_audio_fini (void *opaque)
 858 {
 859     (void) opaque;
 860 }
 861
 862 static struct audio_option alsa_options[] = {
 863     {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
 864      "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 865     {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
 866      "DAC period size (0 to go with system default)",
 867      &conf.period_size_out_overridden, 0},
 868     {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
 869      "DAC buffer size (0 to go with system default)",
 870      &conf.buffer_size_out_overridden, 0},
 871
 872     {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
 873      "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 874     {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
 875      "ADC period size (0 to go with system default)",
 876      &conf.period_size_in_overridden, 0},
 877     {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
 878      "ADC buffer size (0 to go with system default)",
 879      &conf.buffer_size_in_overridden, 0},
 880
 881     {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
 882      "(undocumented)", NULL, 0},
 883
 884     {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
 885      "DAC device name (for instance dmix)", NULL, 0},
 886
 887     {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
 888      "ADC device name", NULL, 0},
 889
 890     {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
 891      "Behave in a more verbose way", NULL, 0},
 892
 893     {NULL, 0, NULL, NULL, NULL, 0}
 894 };
 895
 896 static struct audio_pcm_ops alsa_pcm_ops = {
 897     alsa_init_out,
 898     alsa_fini_out,
 899     alsa_run_out,
 900     alsa_write,
 901     alsa_ctl_out,
 902
 903     alsa_init_in,
 904     alsa_fini_in,
 905     alsa_run_in,
 906     alsa_read,
 907     alsa_ctl_in
 908 };
 909
 910 struct audio_driver alsa_audio_driver = {
 911     INIT_FIELD (name           = ) "alsa",
 912     INIT_FIELD (descr          = ) "ALSA http://www.alsa-project.org",
 913     INIT_FIELD (options        = ) alsa_options,
 914     INIT_FIELD (init           = ) alsa_audio_init,
 915     INIT_FIELD (fini           = ) alsa_audio_fini,
 916     INIT_FIELD (pcm_ops        = ) &alsa_pcm_ops,
 917     INIT_FIELD (can_be_default = ) 1,
 918     INIT_FIELD (max_voices_out = ) INT_MAX,
 919     INIT_FIELD (max_voices_in  = ) INT_MAX,
 920     INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
 921     INIT_FIELD (voice_size_in  = ) sizeof (ALSAVoiceIn)
 922 };