audio/alsaaudio.c

   1 /*
   2  * QEMU ALSA audio driver
   3  *
   4  * Copyright (c) 2005 Vassili Karpov (malc)
   5  *
   6  * Permission is hereby granted, free of charge, to any person obtaining a copy
   7  * of this software and associated documentation files (the "Software"), to deal
   8  * in the Software without restriction, including without limitation the rights
   9  * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
  10  * copies of the Software, and to permit persons to whom the Software is
  11  * furnished to do so, subject to the following conditions:
  12  *
  13  * The above copyright notice and this permission notice shall be included in
  14  * all copies or substantial portions of the Software.
  15  *
  16  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
  17  * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
  18  * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
  19  * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
  20  * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
  21  * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
  22  * THE SOFTWARE.
  23  */
  24 #include <alsa/asoundlib.h>
  25 #include "vl.h"
  26
  27 #define AUDIO_CAP "alsa"
  28 #include "audio_int.h"
  29
  30 typedef struct ALSAVoiceOut {
  31     HWVoiceOut hw;
  32     void *pcm_buf;
  33     snd_pcm_t *handle;
  34 } ALSAVoiceOut;
  35
  36 typedef struct ALSAVoiceIn {
  37     HWVoiceIn hw;
  38     snd_pcm_t *handle;
  39     void *pcm_buf;
  40 } ALSAVoiceIn;
  41
  42 static struct {
  43     int size_in_usec_in;
  44     int size_in_usec_out;
  45     const char *pcm_name_in;
  46     const char *pcm_name_out;
  47     unsigned int buffer_size_in;
  48     unsigned int period_size_in;
  49     unsigned int buffer_size_out;
  50     unsigned int period_size_out;
  51     unsigned int threshold;
  52
  53     int buffer_size_in_overriden;
  54     int period_size_in_overriden;
  55
  56     int buffer_size_out_overriden;
  57     int period_size_out_overriden;
  58     int verbose;
  59 } conf = {
  60 #define DEFAULT_BUFFER_SIZE 1024
  61 #define DEFAULT_PERIOD_SIZE 256
  62 #ifdef HIGH_LATENCY
  63     .size_in_usec_in = 1,
  64     .size_in_usec_out = 1,
  65 #endif
  66     .pcm_name_out = "default",
  67     .pcm_name_in = "default",
  68 #ifdef HIGH_LATENCY
  69     .buffer_size_in = 400000,
  70     .period_size_in = 400000 / 4,
  71     .buffer_size_out = 400000,
  72     .period_size_out = 400000 / 4,
  73 #else
  74     .buffer_size_in = DEFAULT_BUFFER_SIZE * 4,
  75     .period_size_in = DEFAULT_PERIOD_SIZE * 4,
  76     .buffer_size_out = DEFAULT_BUFFER_SIZE,
  77     .period_size_out = DEFAULT_PERIOD_SIZE,
  78     .buffer_size_in_overriden = 0,
  79     .buffer_size_out_overriden = 0,
  80     .period_size_in_overriden = 0,
  81     .period_size_out_overriden = 0,
  82 #endif
  83     .threshold = 0,
  84     .verbose = 0
  85 };
  86
  87 struct alsa_params_req {
  88     int freq;
  89     audfmt_e fmt;
  90     int nchannels;
  91     unsigned int buffer_size;
  92     unsigned int period_size;
  93 };
  94
  95 struct alsa_params_obt {
  96     int freq;
  97     audfmt_e fmt;
  98     int nchannels;
  99     snd_pcm_uframes_t samples;
 100 };
 101
 102 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
 103 {
 104     va_list ap;
 105
 106     va_start (ap, fmt);
 107     AUD_vlog (AUDIO_CAP, fmt, ap);
 108     va_end (ap);
 109
 110     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 111 }
 112
 113 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
 114     int err,
 115     const char *typ,
 116     const char *fmt,
 117     ...
 118     )
 119 {
 120     va_list ap;
 121
 122     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
 123
 124     va_start (ap, fmt);
 125     AUD_vlog (AUDIO_CAP, fmt, ap);
 126     va_end (ap);
 127
 128     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 129 }
 130
 131 static void alsa_anal_close (snd_pcm_t **handlep)
 132 {
 133     int err = snd_pcm_close (*handlep);
 134     if (err) {
 135         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 136     }
 137     *handlep = NULL;
 138 }
 139
 140 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
 141 {
 142     return audio_pcm_sw_write (sw, buf, len);
 143 }
 144
 145 static int aud_to_alsafmt (audfmt_e fmt)
 146 {
 147     switch (fmt) {
 148     case AUD_FMT_S8:
 149         return SND_PCM_FORMAT_S8;
 150
 151     case AUD_FMT_U8:
 152         return SND_PCM_FORMAT_U8;
 153
 154     case AUD_FMT_S16:
 155         return SND_PCM_FORMAT_S16_LE;
 156
 157     case AUD_FMT_U16:
 158         return SND_PCM_FORMAT_U16_LE;
 159
 160     case AUD_FMT_S32:
 161         return SND_PCM_FORMAT_S32_LE;
 162
 163     case AUD_FMT_U32:
 164         return SND_PCM_FORMAT_U32_LE;
 165
 166     default:
 167         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 168 #ifdef DEBUG_AUDIO
 169         abort ();
 170 #endif
 171         return SND_PCM_FORMAT_U8;
 172     }
 173 }
 174
 175 static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
 176 {
 177     switch (alsafmt) {
 178     case SND_PCM_FORMAT_S8:
 179         *endianness = 0;
 180         *fmt = AUD_FMT_S8;
 181         break;
 182
 183     case SND_PCM_FORMAT_U8:
 184         *endianness = 0;
 185         *fmt = AUD_FMT_U8;
 186         break;
 187
 188     case SND_PCM_FORMAT_S16_LE:
 189         *endianness = 0;
 190         *fmt = AUD_FMT_S16;
 191         break;
 192
 193     case SND_PCM_FORMAT_U16_LE:
 194         *endianness = 0;
 195         *fmt = AUD_FMT_U16;
 196         break;
 197
 198     case SND_PCM_FORMAT_S16_BE:
 199         *endianness = 1;
 200         *fmt = AUD_FMT_S16;
 201         break;
 202
 203     case SND_PCM_FORMAT_U16_BE:
 204         *endianness = 1;
 205         *fmt = AUD_FMT_U16;
 206         break;
 207
 208     case SND_PCM_FORMAT_S32_LE:
 209         *endianness = 0;
 210         *fmt = AUD_FMT_S32;
 211         break;
 212
 213     case SND_PCM_FORMAT_U32_LE:
 214         *endianness = 0;
 215         *fmt = AUD_FMT_U32;
 216         break;
 217
 218     case SND_PCM_FORMAT_S32_BE:
 219         *endianness = 1;
 220         *fmt = AUD_FMT_S32;
 221         break;
 222
 223     case SND_PCM_FORMAT_U32_BE:
 224         *endianness = 1;
 225         *fmt = AUD_FMT_U32;
 226         break;
 227
 228     default:
 229         dolog ("Unrecognized audio format %d\n", alsafmt);
 230         return -1;
 231     }
 232
 233     return 0;
 234 }
 235
 236 #if defined DEBUG_MISMATCHES || defined DEBUG
 237 static void alsa_dump_info (struct alsa_params_req *req,
 238                             struct alsa_params_obt *obt)
 239 {
 240     dolog ("parameter | requested value | obtained value\n");
 241     dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
 242     dolog ("channels  |      %10d |     %10d\n",
 243            req->nchannels, obt->nchannels);
 244     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 245     dolog ("============================================\n");
 246     dolog ("requested: buffer size %d period size %d\n",
 247            req->buffer_size, req->period_size);
 248     dolog ("obtained: samples %ld\n", obt->samples);
 249 }
 250 #endif
 251
 252 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 253 {
 254     int err;
 255     snd_pcm_sw_params_t *sw_params;
 256
 257     snd_pcm_sw_params_alloca (&sw_params);
 258
 259     err = snd_pcm_sw_params_current (handle, sw_params);
 260     if (err < 0) {
 261         dolog ("Could not fully initialize DAC\n");
 262         alsa_logerr (err, "Failed to get current software parameters\n");
 263         return;
 264     }
 265
 266     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 267     if (err < 0) {
 268         dolog ("Could not fully initialize DAC\n");
 269         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 270                      threshold);
 271         return;
 272     }
 273
 274     err = snd_pcm_sw_params (handle, sw_params);
 275     if (err < 0) {
 276         dolog ("Could not fully initialize DAC\n");
 277         alsa_logerr (err, "Failed to set software parameters\n");
 278         return;
 279     }
 280 }
 281
 282 static int alsa_open (int in, struct alsa_params_req *req,
 283                       struct alsa_params_obt *obt, snd_pcm_t **handlep)
 284 {
 285     snd_pcm_t *handle;
 286     snd_pcm_hw_params_t *hw_params;
 287     int err, freq, nchannels;
 288     const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
 289     unsigned int period_size, buffer_size;
 290     snd_pcm_uframes_t obt_buffer_size;
 291     const char *typ = in ? "ADC" : "DAC";
 292
 293     freq = req->freq;
 294     period_size = req->period_size;
 295     buffer_size = req->buffer_size;
 296     nchannels = req->nchannels;
 297
 298     snd_pcm_hw_params_alloca (&hw_params);
 299
 300     err = snd_pcm_open (
 301         &handle,
 302         pcm_name,
 303         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 304         SND_PCM_NONBLOCK
 305         );
 306     if (err < 0) {
 307         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 308         return -1;
 309     }
 310
 311     err = snd_pcm_hw_params_any (handle, hw_params);
 312     if (err < 0) {
 313         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 314         goto err;
 315     }
 316
 317     err = snd_pcm_hw_params_set_access (
 318         handle,
 319         hw_params,
 320         SND_PCM_ACCESS_RW_INTERLEAVED
 321         );
 322     if (err < 0) {
 323         alsa_logerr2 (err, typ, "Failed to set access type\n");
 324         goto err;
 325     }
 326
 327     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 328     if (err < 0) {
 329         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 330         goto err;
 331     }
 332
 333     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 334     if (err < 0) {
 335         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 336         goto err;
 337     }
 338
 339     err = snd_pcm_hw_params_set_channels_near (
 340         handle,
 341         hw_params,
 342         &nchannels
 343         );
 344     if (err < 0) {
 345         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 346                       req->nchannels);
 347         goto err;
 348     }
 349
 350     if (nchannels != 1 && nchannels != 2) {
 351         alsa_logerr2 (err, typ,
 352                       "Can not handle obtained number of channels %d\n",
 353                       nchannels);
 354         goto err;
 355     }
 356
 357     if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
 358         if (!buffer_size) {
 359             buffer_size = DEFAULT_BUFFER_SIZE;
 360             period_size= DEFAULT_PERIOD_SIZE;
 361         }
 362     }
 363
 364     if (buffer_size) {
 365         if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
 366             if (period_size) {
 367                 err = snd_pcm_hw_params_set_period_time_near (
 368                     handle,
 369                     hw_params,
 370                     &period_size,
 371                     0
 372                     );
 373                 if (err < 0) {
 374                     alsa_logerr2 (err, typ,
 375                                   "Failed to set period time %d\n",
 376                                   req->period_size);
 377                     goto err;
 378                 }
 379             }
 380
 381             err = snd_pcm_hw_params_set_buffer_time_near (
 382                 handle,
 383                 hw_params,
 384                 &buffer_size,
 385                 0
 386                 );
 387
 388             if (err < 0) {
 389                 alsa_logerr2 (err, typ,
 390                               "Failed to set buffer time %d\n",
 391                               req->buffer_size);
 392                 goto err;
 393             }
 394         }
 395         else {
 396             int dir;
 397             snd_pcm_uframes_t minval;
 398
 399             if (period_size) {
 400                 minval = period_size;
 401                 dir = 0;
 402
 403                 err = snd_pcm_hw_params_get_period_size_min (
 404                     hw_params,
 405                     &minval,
 406                     &dir
 407                     );
 408                 if (err < 0) {
 409                     alsa_logerr (
 410                         err,
 411                         "Could not get minmal period size for %s\n",
 412                         typ
 413                         );
 414                 }
 415                 else {
 416                     if (period_size < minval) {
 417                         if ((in && conf.period_size_in_overriden)
 418                             || (!in && conf.period_size_out_overriden)) {
 419                             dolog ("%s period size(%d) is less "
 420                                    "than minmal period size(%ld)\n",
 421                                    typ,
 422                                    period_size,
 423                                    minval);
 424                         }
 425                         period_size = minval;
 426                     }
 427                 }
 428
 429                 err = snd_pcm_hw_params_set_period_size (
 430                     handle,
 431                     hw_params,
 432                     period_size,
 433                     0
 434                     );
 435                 if (err < 0) {
 436                     alsa_logerr2 (err, typ, "Failed to set period size %d\n",
 437                                   req->period_size);
 438                     goto err;
 439                 }
 440             }
 441
 442             minval = buffer_size;
 443             err = snd_pcm_hw_params_get_buffer_size_min (
 444                 hw_params,
 445                 &minval
 446                 );
 447             if (err < 0) {
 448                 alsa_logerr (err, "Could not get minmal buffer size for %s\n",
 449                              typ);
 450             }
 451             else {
 452                 if (buffer_size < minval) {
 453                     if ((in && conf.buffer_size_in_overriden)
 454                         || (!in && conf.buffer_size_out_overriden)) {
 455                         dolog (
 456                             "%s buffer size(%d) is less "
 457                             "than minimal buffer size(%ld)\n",
 458                             typ,
 459                             buffer_size,
 460                             minval
 461                             );
 462                     }
 463                     buffer_size = minval;
 464                 }
 465             }
 466
 467             err = snd_pcm_hw_params_set_buffer_size (
 468                 handle,
 469                 hw_params,
 470                 buffer_size
 471                 );
 472             if (err < 0) {
 473                 alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
 474                               req->buffer_size);
 475                 goto err;
 476             }
 477         }
 478     }
 479     else {
 480         dolog ("warning: Buffer size is not set\n");
 481     }
 482
 483     err = snd_pcm_hw_params (handle, hw_params);
 484     if (err < 0) {
 485         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 486         goto err;
 487     }
 488
 489     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 490     if (err < 0) {
 491         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 492         goto err;
 493     }
 494
 495     err = snd_pcm_prepare (handle);
 496     if (err < 0) {
 497         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 498         goto err;
 499     }
 500
 501     if (!in && conf.threshold) {
 502         snd_pcm_uframes_t threshold;
 503         int bytes_per_sec;
 504
 505         bytes_per_sec = freq
 506             << (nchannels == 2)
 507             << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
 508
 509         threshold = (conf.threshold * bytes_per_sec) / 1000;
 510         alsa_set_threshold (handle, threshold);
 511     }
 512
 513     obt->fmt = req->fmt;
 514     obt->nchannels = nchannels;
 515     obt->freq = freq;
 516     obt->samples = obt_buffer_size;
 517     *handlep = handle;
 518
 519 #if defined DEBUG_MISMATCHES || defined DEBUG
 520     if (obt->fmt != req->fmt ||
 521         obt->nchannels != req->nchannels ||
 522         obt->freq != req->freq) {
 523         dolog ("Audio paramters mismatch for %s\n", typ);
 524         alsa_dump_info (req, obt);
 525     }
 526 #endif
 527
 528 #ifdef DEBUG
 529     alsa_dump_info (req, obt);
 530 #endif
 531     return 0;
 532
 533  err:
 534     alsa_anal_close (&handle);
 535     return -1;
 536 }
 537
 538 static int alsa_recover (snd_pcm_t *handle)
 539 {
 540     int err = snd_pcm_prepare (handle);
 541     if (err < 0) {
 542         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 543         return -1;
 544     }
 545     return 0;
 546 }
 547
 548 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 549 {
 550     snd_pcm_sframes_t avail;
 551
 552     avail = snd_pcm_avail_update (handle);
 553     if (avail < 0) {
 554         if (avail == -EPIPE) {
 555             if (!alsa_recover (handle)) {
 556                 avail = snd_pcm_avail_update (handle);
 557             }
 558         }
 559
 560         if (avail < 0) {
 561             alsa_logerr (avail,
 562                          "Could not obtain number of available frames\n");
 563             return -1;
 564         }
 565     }
 566
 567     return avail;
 568 }
 569
 570 static int alsa_run_out (HWVoiceOut *hw)
 571 {
 572     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 573     int rpos, live, decr;
 574     int samples;
 575     uint8_t *dst;
 576     st_sample_t *src;
 577     snd_pcm_sframes_t avail;
 578
 579     live = audio_pcm_hw_get_live_out (hw);
 580     if (!live) {
 581         return 0;
 582     }
 583
 584     avail = alsa_get_avail (alsa->handle);
 585     if (avail < 0) {
 586         dolog ("Could not get number of available playback frames\n");
 587         return 0;
 588     }
 589
 590     decr = audio_MIN (live, avail);
 591     samples = decr;
 592     rpos = hw->rpos;
 593     while (samples) {
 594         int left_till_end_samples = hw->samples - rpos;
 595         int len = audio_MIN (samples, left_till_end_samples);
 596         snd_pcm_sframes_t written;
 597
 598         src = hw->mix_buf + rpos;
 599         dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
 600
 601         hw->clip (dst, src, len);
 602
 603         while (len) {
 604             written = snd_pcm_writei (alsa->handle, dst, len);
 605
 606             if (written <= 0) {
 607                 switch (written) {
 608                 case 0:
 609                     if (conf.verbose) {
 610                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 611                     }
 612                     goto exit;
 613
 614                 case -EPIPE:
 615                     if (alsa_recover (alsa->handle)) {
 616                         alsa_logerr (written, "Failed to write %d frames\n",
 617                                      len);
 618                         goto exit;
 619                     }
 620                     if (conf.verbose) {
 621                         dolog ("Recovering from playback xrun\n");
 622                     }
 623                     continue;
 624
 625                 case -EAGAIN:
 626                     goto exit;
 627
 628                 default:
 629                     alsa_logerr (written, "Failed to write %d frames to %p\n",
 630                                  len, dst);
 631                     goto exit;
 632                 }
 633             }
 634
 635             rpos = (rpos + written) % hw->samples;
 636             samples -= written;
 637             len -= written;
 638             dst = advance (dst, written << hw->info.shift);
 639             src += written;
 640         }
 641     }
 642
 643  exit:
 644     hw->rpos = rpos;
 645     return decr;
 646 }
 647
 648 static void alsa_fini_out (HWVoiceOut *hw)
 649 {
 650     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 651
 652     ldebug ("alsa_fini\n");
 653     alsa_anal_close (&alsa->handle);
 654
 655     if (alsa->pcm_buf) {
 656         qemu_free (alsa->pcm_buf);
 657         alsa->pcm_buf = NULL;
 658     }
 659 }
 660
 661 static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
 662 {
 663     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 664     struct alsa_params_req req;
 665     struct alsa_params_obt obt;
 666     audfmt_e effective_fmt;
 667     int endianness;
 668     int err;
 669     snd_pcm_t *handle;
 670     audsettings_t obt_as;
 671
 672     req.fmt = aud_to_alsafmt (as->fmt);
 673     req.freq = as->freq;
 674     req.nchannels = as->nchannels;
 675     req.period_size = conf.period_size_out;
 676     req.buffer_size = conf.buffer_size_out;
 677
 678     if (alsa_open (0, &req, &obt, &handle)) {
 679         return -1;
 680     }
 681
 682     err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
 683     if (err) {
 684         alsa_anal_close (&handle);
 685         return -1;
 686     }
 687
 688     obt_as.freq = obt.freq;
 689     obt_as.nchannels = obt.nchannels;
 690     obt_as.fmt = effective_fmt;
 691     obt_as.endianness = endianness;
 692
 693     audio_pcm_init_info (&hw->info, &obt_as);
 694     hw->samples = obt.samples;
 695
 696     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 697     if (!alsa->pcm_buf) {
 698         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 699                hw->samples, 1 << hw->info.shift);
 700         alsa_anal_close (&handle);
 701         return -1;
 702     }
 703
 704     alsa->handle = handle;
 705     return 0;
 706 }
 707
 708 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
 709 {
 710     int err;
 711
 712     if (pause) {
 713         err = snd_pcm_drop (handle);
 714         if (err < 0) {
 715             alsa_logerr (err, "Could not stop %s\n", typ);
 716             return -1;
 717         }
 718     }
 719     else {
 720         err = snd_pcm_prepare (handle);
 721         if (err < 0) {
 722             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 723             return -1;
 724         }
 725     }
 726
 727     return 0;
 728 }
 729
 730 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 731 {
 732     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 733
 734     switch (cmd) {
 735     case VOICE_ENABLE:
 736         ldebug ("enabling voice\n");
 737         return alsa_voice_ctl (alsa->handle, "playback", 0);
 738
 739     case VOICE_DISABLE:
 740         ldebug ("disabling voice\n");
 741         return alsa_voice_ctl (alsa->handle, "playback", 1);
 742     }
 743
 744     return -1;
 745 }
 746
 747 static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
 748 {
 749     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 750     struct alsa_params_req req;
 751     struct alsa_params_obt obt;
 752     int endianness;
 753     int err;
 754     audfmt_e effective_fmt;
 755     snd_pcm_t *handle;
 756     audsettings_t obt_as;
 757
 758     req.fmt = aud_to_alsafmt (as->fmt);
 759     req.freq = as->freq;
 760     req.nchannels = as->nchannels;
 761     req.period_size = conf.period_size_in;
 762     req.buffer_size = conf.buffer_size_in;
 763
 764     if (alsa_open (1, &req, &obt, &handle)) {
 765         return -1;
 766     }
 767
 768     err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
 769     if (err) {
 770         alsa_anal_close (&handle);
 771         return -1;
 772     }
 773
 774     obt_as.freq = obt.freq;
 775     obt_as.nchannels = obt.nchannels;
 776     obt_as.fmt = effective_fmt;
 777     obt_as.endianness = endianness;
 778
 779     audio_pcm_init_info (&hw->info, &obt_as);
 780     hw->samples = obt.samples;
 781
 782     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 783     if (!alsa->pcm_buf) {
 784         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 785                hw->samples, 1 << hw->info.shift);
 786         alsa_anal_close (&handle);
 787         return -1;
 788     }
 789
 790     alsa->handle = handle;
 791     return 0;
 792 }
 793
 794 static void alsa_fini_in (HWVoiceIn *hw)
 795 {
 796     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 797
 798     alsa_anal_close (&alsa->handle);
 799
 800     if (alsa->pcm_buf) {
 801         qemu_free (alsa->pcm_buf);
 802         alsa->pcm_buf = NULL;
 803     }
 804 }
 805
 806 static int alsa_run_in (HWVoiceIn *hw)
 807 {
 808     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 809     int hwshift = hw->info.shift;
 810     int i;
 811     int live = audio_pcm_hw_get_live_in (hw);
 812     int dead = hw->samples - live;
 813     int decr;
 814     struct {
 815         int add;
 816         int len;
 817     } bufs[2] = {
 818         { hw->wpos, 0 },
 819         { 0, 0 }
 820     };
 821     snd_pcm_sframes_t avail;
 822     snd_pcm_uframes_t read_samples = 0;
 823
 824     if (!dead) {
 825         return 0;
 826     }
 827
 828     avail = alsa_get_avail (alsa->handle);
 829     if (avail < 0) {
 830         dolog ("Could not get number of captured frames\n");
 831         return 0;
 832     }
 833
 834     if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
 835         avail = hw->samples;
 836     }
 837
 838     decr = audio_MIN (dead, avail);
 839     if (!decr) {
 840         return 0;
 841     }
 842
 843     if (hw->wpos + decr > hw->samples) {
 844         bufs[0].len = (hw->samples - hw->wpos);
 845         bufs[1].len = (decr - (hw->samples - hw->wpos));
 846     }
 847     else {
 848         bufs[0].len = decr;
 849     }
 850
 851     for (i = 0; i < 2; ++i) {
 852         void *src;
 853         st_sample_t *dst;
 854         snd_pcm_sframes_t nread;
 855         snd_pcm_uframes_t len;
 856
 857         len = bufs[i].len;
 858
 859         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
 860         dst = hw->conv_buf + bufs[i].add;
 861
 862         while (len) {
 863             nread = snd_pcm_readi (alsa->handle, src, len);
 864
 865             if (nread <= 0) {
 866                 switch (nread) {
 867                 case 0:
 868                     if (conf.verbose) {
 869                         dolog ("Failed to read %ld frames (read zero)\n", len);
 870                     }
 871                     goto exit;
 872
 873                 case -EPIPE:
 874                     if (alsa_recover (alsa->handle)) {
 875                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
 876                         goto exit;
 877                     }
 878                     if (conf.verbose) {
 879                         dolog ("Recovering from capture xrun\n");
 880                     }
 881                     continue;
 882
 883                 case -EAGAIN:
 884                     goto exit;
 885
 886                 default:
 887                     alsa_logerr (
 888                         nread,
 889                         "Failed to read %ld frames from %p\n",
 890                         len,
 891                         src
 892                         );
 893                     goto exit;
 894                 }
 895             }
 896
 897             hw->conv (dst, src, nread, &nominal_volume);
 898
 899             src = advance (src, nread << hwshift);
 900             dst += nread;
 901
 902             read_samples += nread;
 903             len -= nread;
 904         }
 905     }
 906
 907  exit:
 908     hw->wpos = (hw->wpos + read_samples) % hw->samples;
 909     return read_samples;
 910 }
 911
 912 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
 913 {
 914     return audio_pcm_sw_read (sw, buf, size);
 915 }
 916
 917 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
 918 {
 919     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 920
 921     switch (cmd) {
 922     case VOICE_ENABLE:
 923         ldebug ("enabling voice\n");
 924         return alsa_voice_ctl (alsa->handle, "capture", 0);
 925
 926     case VOICE_DISABLE:
 927         ldebug ("disabling voice\n");
 928         return alsa_voice_ctl (alsa->handle, "capture", 1);
 929     }
 930
 931     return -1;
 932 }
 933
 934 static void *alsa_audio_init (void)
 935 {
 936     return &conf;
 937 }
 938
 939 static void alsa_audio_fini (void *opaque)
 940 {
 941     (void) opaque;
 942 }
 943
 944 static struct audio_option alsa_options[] = {
 945     {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
 946      "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 947     {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
 948      "DAC period size", &conf.period_size_out_overriden, 0},
 949     {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
 950      "DAC buffer size", &conf.buffer_size_out_overriden, 0},
 951
 952     {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
 953      "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
 954     {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
 955      "ADC period size", &conf.period_size_in_overriden, 0},
 956     {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
 957      "ADC buffer size", &conf.buffer_size_in_overriden, 0},
 958
 959     {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
 960      "(undocumented)", NULL, 0},
 961
 962     {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
 963      "DAC device name (for instance dmix)", NULL, 0},
 964
 965     {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
 966      "ADC device name", NULL, 0},
 967
 968     {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
 969      "Behave in a more verbose way", NULL, 0},
 970
 971     {NULL, 0, NULL, NULL, NULL, 0}
 972 };
 973
 974 static struct audio_pcm_ops alsa_pcm_ops = {
 975     alsa_init_out,
 976     alsa_fini_out,
 977     alsa_run_out,
 978     alsa_write,
 979     alsa_ctl_out,
 980
 981     alsa_init_in,
 982     alsa_fini_in,
 983     alsa_run_in,
 984     alsa_read,
 985     alsa_ctl_in
 986 };
 987
 988 struct audio_driver alsa_audio_driver = {
 989     INIT_FIELD (name           = ) "alsa",
 990     INIT_FIELD (descr          = ) "ALSA http://www.alsa-project.org",
 991     INIT_FIELD (options        = ) alsa_options,
 992     INIT_FIELD (init           = ) alsa_audio_init,
 993     INIT_FIELD (fini           = ) alsa_audio_fini,
 994     INIT_FIELD (pcm_ops        = ) &alsa_pcm_ops,
 995     INIT_FIELD (can_be_default = ) 1,
 996     INIT_FIELD (max_voices_out = ) INT_MAX,
 997     INIT_FIELD (max_voices_in  = ) INT_MAX,
 998     INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
 999     INIT_FIELD (voice_size_in  = ) sizeof (ALSAVoiceIn)
1000 };