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         unsigned long obt;
 338
 339         if (size_in_usec) {
 340             int dir = 0;
 341             unsigned int btime = req->buffer_size;
 342
 343             err = snd_pcm_hw_params_set_buffer_time_near (
 344                 handle,
 345                 hw_params,
 346                 &btime,
 347                 &dir
 348                 );
 349             obt = btime;
 350         }
 351         else {
 352             snd_pcm_uframes_t bsize = req->buffer_size;
 353
 354             err = snd_pcm_hw_params_set_buffer_size_near (
 355                 handle,
 356                 hw_params,
 357                 &bsize
 358                 );
 359             obt = bsize;
 360         }
 361         if (err < 0) {
 362             alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
 363                           size_in_usec ? "time" : "size", req->buffer_size);
 364             goto err;
 365         }
 366
 367         if (obt - req->buffer_size)
 368             dolog ("Requested buffer %s %u was rejected, using %lu\n",
 369                    size_in_usec ? "time" : "size", req->buffer_size, obt);
 370     }
 371
 372     if (req->period_size) {
 373         unsigned long obt;
 374
 375         if (size_in_usec) {
 376             int dir = 0;
 377             unsigned int ptime = req->period_size;
 378
 379             err = snd_pcm_hw_params_set_period_time_near (
 380                 handle,
 381                 hw_params,
 382                 &ptime,
 383                 &dir
 384                 );
 385             obt = ptime;
 386         }
 387         else {
 388             snd_pcm_uframes_t psize = req->period_size;
 389
 390             err = snd_pcm_hw_params_set_buffer_size_near (
 391                 handle,
 392                 hw_params,
 393                 &psize
 394                 );
 395             obt = psize;
 396         }
 397
 398         if (err < 0) {
 399             alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
 400                           size_in_usec ? "time" : "size", req->period_size);
 401             goto err;
 402         }
 403
 404         if (obt - req->period_size)
 405             dolog ("Requested period %s %u was rejected, using %lu\n",
 406                    size_in_usec ? "time" : "size", req->period_size, obt);
 407     }
 408
 409     err = snd_pcm_hw_params (handle, hw_params);
 410     if (err < 0) {
 411         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 412         goto err;
 413     }
 414
 415     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 416     if (err < 0) {
 417         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 418         goto err;
 419     }
 420
 421     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 422     if (err < 0) {
 423         alsa_logerr2 (err, typ, "Failed to get format\n");
 424         goto err;
 425     }
 426
 427     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 428         dolog ("Invalid format was returned %d\n", obtfmt);
 429         goto err;
 430     }
 431
 432     err = snd_pcm_prepare (handle);
 433     if (err < 0) {
 434         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 435         goto err;
 436     }
 437
 438     if (!in && conf.threshold) {
 439         snd_pcm_uframes_t threshold;
 440         int bytes_per_sec;
 441
 442         bytes_per_sec = freq << (nchannels == 2);
 443
 444         switch (obt->fmt) {
 445         case AUD_FMT_S8:
 446         case AUD_FMT_U8:
 447             break;
 448
 449         case AUD_FMT_S16:
 450         case AUD_FMT_U16:
 451             bytes_per_sec <<= 1;
 452             break;
 453
 454         case AUD_FMT_S32:
 455         case AUD_FMT_U32:
 456             bytes_per_sec <<= 2;
 457             break;
 458         }
 459
 460         threshold = (conf.threshold * bytes_per_sec) / 1000;
 461         alsa_set_threshold (handle, threshold);
 462     }
 463
 464     obt->nchannels = nchannels;
 465     obt->freq = freq;
 466     obt->samples = obt_buffer_size;
 467
 468     *handlep = handle;
 469
 470     if (conf.verbose &&
 471         (obt->fmt != req->fmt ||
 472          obt->nchannels != req->nchannels ||
 473          obt->freq != req->freq)) {
 474         dolog ("Audio paramters for %s\n", typ);
 475         alsa_dump_info (req, obt);
 476     }
 477
 478 #ifdef DEBUG
 479     alsa_dump_info (req, obt);
 480 #endif
 481     return 0;
 482
 483  err:
 484     alsa_anal_close (&handle);
 485     return -1;
 486 }
 487
 488 static int alsa_recover (snd_pcm_t *handle)
 489 {
 490     int err = snd_pcm_prepare (handle);
 491     if (err < 0) {
 492         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 493         return -1;
 494     }
 495     return 0;
 496 }
 497
 498 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 499 {
 500     snd_pcm_sframes_t avail;
 501
 502     avail = snd_pcm_avail_update (handle);
 503     if (avail < 0) {
 504         if (avail == -EPIPE) {
 505             if (!alsa_recover (handle)) {
 506                 avail = snd_pcm_avail_update (handle);
 507             }
 508         }
 509
 510         if (avail < 0) {
 511             alsa_logerr (avail,
 512                          "Could not obtain number of available frames\n");
 513             return -1;
 514         }
 515     }
 516
 517     return avail;
 518 }
 519
 520 static int alsa_run_out (HWVoiceOut *hw)
 521 {
 522     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 523     int rpos, live, decr;
 524     int samples;
 525     uint8_t *dst;
 526     st_sample_t *src;
 527     snd_pcm_sframes_t avail;
 528
 529     live = audio_pcm_hw_get_live_out (hw);
 530     if (!live) {
 531         return 0;
 532     }
 533
 534     avail = alsa_get_avail (alsa->handle);
 535     if (avail < 0) {
 536         dolog ("Could not get number of available playback frames\n");
 537         return 0;
 538     }
 539
 540     decr = audio_MIN (live, avail);
 541     samples = decr;
 542     rpos = hw->rpos;
 543     while (samples) {
 544         int left_till_end_samples = hw->samples - rpos;
 545         int len = audio_MIN (samples, left_till_end_samples);
 546         snd_pcm_sframes_t written;
 547
 548         src = hw->mix_buf + rpos;
 549         dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
 550
 551         hw->clip (dst, src, len);
 552
 553         while (len) {
 554             written = snd_pcm_writei (alsa->handle, dst, len);
 555
 556             if (written <= 0) {
 557                 switch (written) {
 558                 case 0:
 559                     if (conf.verbose) {
 560                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 561                     }
 562                     goto exit;
 563
 564                 case -EPIPE:
 565                     if (alsa_recover (alsa->handle)) {
 566                         alsa_logerr (written, "Failed to write %d frames\n",
 567                                      len);
 568                         goto exit;
 569                     }
 570                     if (conf.verbose) {
 571                         dolog ("Recovering from playback xrun\n");
 572                     }
 573                     continue;
 574
 575                 case -EAGAIN:
 576                     goto exit;
 577
 578                 default:
 579                     alsa_logerr (written, "Failed to write %d frames to %p\n",
 580                                  len, dst);
 581                     goto exit;
 582                 }
 583             }
 584
 585             rpos = (rpos + written) % hw->samples;
 586             samples -= written;
 587             len -= written;
 588             dst = advance (dst, written << hw->info.shift);
 589             src += written;
 590         }
 591     }
 592
 593  exit:
 594     hw->rpos = rpos;
 595     return decr;
 596 }
 597
 598 static void alsa_fini_out (HWVoiceOut *hw)
 599 {
 600     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 601
 602     ldebug ("alsa_fini\n");
 603     alsa_anal_close (&alsa->handle);
 604
 605     if (alsa->pcm_buf) {
 606         qemu_free (alsa->pcm_buf);
 607         alsa->pcm_buf = NULL;
 608     }
 609 }
 610
 611 static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
 612 {
 613     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 614     struct alsa_params_req req;
 615     struct alsa_params_obt obt;
 616     snd_pcm_t *handle;
 617     audsettings_t obt_as;
 618
 619     req.fmt = aud_to_alsafmt (as->fmt);
 620     req.freq = as->freq;
 621     req.nchannels = as->nchannels;
 622     req.period_size = conf.period_size_out;
 623     req.buffer_size = conf.buffer_size_out;
 624     req.size_in_usec = conf.size_in_usec_in;
 625
 626     if (alsa_open (0, &req, &obt, &handle)) {
 627         return -1;
 628     }
 629
 630     obt_as.freq = obt.freq;
 631     obt_as.nchannels = obt.nchannels;
 632     obt_as.fmt = obt.fmt;
 633     obt_as.endianness = obt.endianness;
 634
 635     audio_pcm_init_info (&hw->info, &obt_as);
 636     hw->samples = obt.samples;
 637
 638     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 639     if (!alsa->pcm_buf) {
 640         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 641                hw->samples, 1 << hw->info.shift);
 642         alsa_anal_close (&handle);
 643         return -1;
 644     }
 645
 646     alsa->handle = handle;
 647     return 0;
 648 }
 649
 650 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
 651 {
 652     int err;
 653
 654     if (pause) {
 655         err = snd_pcm_drop (handle);
 656         if (err < 0) {
 657             alsa_logerr (err, "Could not stop %s\n", typ);
 658             return -1;
 659         }
 660     }
 661     else {
 662         err = snd_pcm_prepare (handle);
 663         if (err < 0) {
 664             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 665             return -1;
 666         }
 667     }
 668
 669     return 0;
 670 }
 671
 672 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 673 {
 674     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 675
 676     switch (cmd) {
 677     case VOICE_ENABLE:
 678         ldebug ("enabling voice\n");
 679         return alsa_voice_ctl (alsa->handle, "playback", 0);
 680
 681     case VOICE_DISABLE:
 682         ldebug ("disabling voice\n");
 683         return alsa_voice_ctl (alsa->handle, "playback", 1);
 684     }
 685
 686     return -1;
 687 }
 688
 689 static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
 690 {
 691     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 692     struct alsa_params_req req;
 693     struct alsa_params_obt obt;
 694     snd_pcm_t *handle;
 695     audsettings_t obt_as;
 696
 697     req.fmt = aud_to_alsafmt (as->fmt);
 698     req.freq = as->freq;
 699     req.nchannels = as->nchannels;
 700     req.period_size = conf.period_size_in;
 701     req.buffer_size = conf.buffer_size_in;
 702     req.size_in_usec = conf.size_in_usec_in;
 703
 704     if (alsa_open (1, &req, &obt, &handle)) {
 705         return -1;
 706     }
 707
 708     obt_as.freq = obt.freq;
 709     obt_as.nchannels = obt.nchannels;
 710     obt_as.fmt = obt.fmt;
 711     obt_as.endianness = obt.endianness;
 712
 713     audio_pcm_init_info (&hw->info, &obt_as);
 714     hw->samples = obt.samples;
 715
 716     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 717     if (!alsa->pcm_buf) {
 718         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 719                hw->samples, 1 << hw->info.shift);
 720         alsa_anal_close (&handle);
 721         return -1;
 722     }
 723
 724     alsa->handle = handle;
 725     return 0;
 726 }
 727
 728 static void alsa_fini_in (HWVoiceIn *hw)
 729 {
 730     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 731
 732     alsa_anal_close (&alsa->handle);
 733
 734     if (alsa->pcm_buf) {
 735         qemu_free (alsa->pcm_buf);
 736         alsa->pcm_buf = NULL;
 737     }
 738 }
 739
 740 static int alsa_run_in (HWVoiceIn *hw)
 741 {
 742     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 743     int hwshift = hw->info.shift;
 744     int i;
 745     int live = audio_pcm_hw_get_live_in (hw);
 746     int dead = hw->samples - live;
 747     int decr;
 748     struct {
 749         int add;
 750         int len;
 751     } bufs[2] = {
 752         { hw->wpos, 0 },
 753         { 0, 0 }
 754     };
 755     snd_pcm_sframes_t avail;
 756     snd_pcm_uframes_t read_samples = 0;
 757
 758     if (!dead) {
 759         return 0;
 760     }
 761
 762     avail = alsa_get_avail (alsa->handle);
 763     if (avail < 0) {
 764         dolog ("Could not get number of captured frames\n");
 765         return 0;
 766     }
 767
 768     if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
 769         avail = hw->samples;
 770     }
 771
 772     decr = audio_MIN (dead, avail);
 773     if (!decr) {
 774         return 0;
 775     }
 776
 777     if (hw->wpos + decr > hw->samples) {
 778         bufs[0].len = (hw->samples - hw->wpos);
 779         bufs[1].len = (decr - (hw->samples - hw->wpos));
 780     }
 781     else {
 782         bufs[0].len = decr;
 783     }
 784
 785     for (i = 0; i < 2; ++i) {
 786         void *src;
 787         st_sample_t *dst;
 788         snd_pcm_sframes_t nread;
 789         snd_pcm_uframes_t len;
 790
 791         len = bufs[i].len;
 792
 793         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
 794         dst = hw->conv_buf + bufs[i].add;
 795
 796         while (len) {
 797             nread = snd_pcm_readi (alsa->handle, src, len);
 798
 799             if (nread <= 0) {
 800                 switch (nread) {
 801                 case 0:
 802                     if (conf.verbose) {
 803                         dolog ("Failed to read %ld frames (read zero)\n", len);
 804                     }
 805                     goto exit;
 806
 807                 case -EPIPE:
 808                     if (alsa_recover (alsa->handle)) {
 809                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
 810                         goto exit;
 811                     }
 812                     if (conf.verbose) {
 813                         dolog ("Recovering from capture xrun\n");
 814                     }
 815                     continue;
 816
 817                 case -EAGAIN:
 818                     goto exit;
 819
 820                 default:
 821                     alsa_logerr (
 822                         nread,
 823                         "Failed to read %ld frames from %p\n",
 824                         len,
 825                         src
 826                         );
 827                     goto exit;
 828                 }
 829             }
 830
 831             hw->conv (dst, src, nread, &nominal_volume);
 832
 833             src = advance (src, nread << hwshift);
 834             dst += nread;
 835
 836             read_samples += nread;
 837             len -= nread;
 838         }
 839     }
 840
 841  exit:
 842     hw->wpos = (hw->wpos + read_samples) % hw->samples;
 843     return read_samples;
 844 }
 845
 846 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
 847 {
 848     return audio_pcm_sw_read (sw, buf, size);
 849 }
 850
 851 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 852 {
 853     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 854
 855     switch (cmd) {
 856     case VOICE_ENABLE:
 857         ldebug ("enabling voice\n");
 858         return alsa_voice_ctl (alsa->handle, "capture", 0);
 859
 860     case VOICE_DISABLE:
 861         ldebug ("disabling voice\n");
 862         return alsa_voice_ctl (alsa->handle, "capture", 1);
 863     }
 864
 865     return -1;
 866 }
 867
 868 static void *alsa_audio_init (void)
 869 {
 870     return &conf;
 871 }
 872
 873 static void alsa_audio_fini (void *opaque)
 874 {
 875     (void) opaque;
 876 }
 877
 878 static struct audio_option alsa_options[] = {
 879     {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
 880      "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 881     {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
 882      "DAC period size (0 to go with system default)",
 883      &conf.period_size_out_overridden, 0},
 884     {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
 885      "DAC buffer size (0 to go with system default)",
 886      &conf.buffer_size_out_overridden, 0},
 887
 888     {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
 889      "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 890     {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
 891      "ADC period size (0 to go with system default)",
 892      &conf.period_size_in_overridden, 0},
 893     {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
 894      "ADC buffer size (0 to go with system default)",
 895      &conf.buffer_size_in_overridden, 0},
 896
 897     {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
 898      "(undocumented)", NULL, 0},
 899
 900     {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
 901      "DAC device name (for instance dmix)", NULL, 0},
 902
 903     {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
 904      "ADC device name", NULL, 0},
 905
 906     {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
 907      "Behave in a more verbose way", NULL, 0},
 908
 909     {NULL, 0, NULL, NULL, NULL, 0}
 910 };
 911
 912 static struct audio_pcm_ops alsa_pcm_ops = {
 913     alsa_init_out,
 914     alsa_fini_out,
 915     alsa_run_out,
 916     alsa_write,
 917     alsa_ctl_out,
 918
 919     alsa_init_in,
 920     alsa_fini_in,
 921     alsa_run_in,
 922     alsa_read,
 923     alsa_ctl_in
 924 };
 925
 926 struct audio_driver alsa_audio_driver = {
 927     INIT_FIELD (name           = ) "alsa",
 928     INIT_FIELD (descr          = ) "ALSA http://www.alsa-project.org",
 929     INIT_FIELD (options        = ) alsa_options,
 930     INIT_FIELD (init           = ) alsa_audio_init,
 931     INIT_FIELD (fini           = ) alsa_audio_fini,
 932     INIT_FIELD (pcm_ops        = ) &alsa_pcm_ops,
 933     INIT_FIELD (can_be_default = ) 1,
 934     INIT_FIELD (max_voices_out = ) INT_MAX,
 935     INIT_FIELD (max_voices_in  = ) INT_MAX,
 936     INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
 937     INIT_FIELD (voice_size_in  = ) sizeof (ALSAVoiceIn)
 938 };