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 "qemu-char.h"
  27 #include "audio.h"
  28
  29 #if QEMU_GNUC_PREREQ(4, 3)
  30 #pragma GCC diagnostic ignored "-Waddress"
  31 #endif
  32
  33 #define AUDIO_CAP "alsa"
  34 #include "audio_int.h"
  35
  36 struct pollhlp {
  37     snd_pcm_t *handle;
  38     struct pollfd *pfds;
  39     int count;
  40     int mask;
  41 };
  42
  43 typedef struct ALSAVoiceOut {
  44     HWVoiceOut hw;
  45     int wpos;
  46     int pending;
  47     void *pcm_buf;
  48     snd_pcm_t *handle;
  49     struct pollhlp pollhlp;
  50 } ALSAVoiceOut;
  51
  52 typedef struct ALSAVoiceIn {
  53     HWVoiceIn hw;
  54     snd_pcm_t *handle;
  55     void *pcm_buf;
  56     struct pollhlp pollhlp;
  57 } ALSAVoiceIn;
  58
  59 static struct {
  60     int size_in_usec_in;
  61     int size_in_usec_out;
  62     const char *pcm_name_in;
  63     const char *pcm_name_out;
  64     unsigned int buffer_size_in;
  65     unsigned int period_size_in;
  66     unsigned int buffer_size_out;
  67     unsigned int period_size_out;
  68     unsigned int threshold;
  69
  70     int buffer_size_in_overridden;
  71     int period_size_in_overridden;
  72
  73     int buffer_size_out_overridden;
  74     int period_size_out_overridden;
  75     int verbose;
  76 } conf = {
  77     .buffer_size_out = 4096,
  78     .period_size_out = 1024,
  79     .pcm_name_out = "default",
  80     .pcm_name_in = "default",
  81 };
  82
  83 struct alsa_params_req {
  84     int freq;
  85     snd_pcm_format_t fmt;
  86     int nchannels;
  87     int size_in_usec;
  88     int override_mask;
  89     unsigned int buffer_size;
  90     unsigned int period_size;
  91 };
  92
  93 struct alsa_params_obt {
  94     int freq;
  95     audfmt_e fmt;
  96     int endianness;
  97     int nchannels;
  98     snd_pcm_uframes_t samples;
  99 };
 100
 101 static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
 102 {
 103     va_list ap;
 104
 105     va_start (ap, fmt);
 106     AUD_vlog (AUDIO_CAP, fmt, ap);
 107     va_end (ap);
 108
 109     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 110 }
 111
 112 static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
 113     int err,
 114     const char *typ,
 115     const char *fmt,
 116     ...
 117     )
 118 {
 119     va_list ap;
 120
 121     AUD_log (AUDIO_CAP, "Could not initialize %s\n", typ);
 122
 123     va_start (ap, fmt);
 124     AUD_vlog (AUDIO_CAP, fmt, ap);
 125     va_end (ap);
 126
 127     AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
 128 }
 129
 130 static void alsa_fini_poll (struct pollhlp *hlp)
 131 {
 132     int i;
 133     struct pollfd *pfds = hlp->pfds;
 134
 135     if (pfds) {
 136         for (i = 0; i < hlp->count; ++i) {
 137             qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
 138         }
 139         qemu_free (pfds);
 140     }
 141     hlp->pfds = NULL;
 142     hlp->count = 0;
 143     hlp->handle = NULL;
 144 }
 145
 146 static void alsa_anal_close1 (snd_pcm_t **handlep)
 147 {
 148     int err = snd_pcm_close (*handlep);
 149     if (err) {
 150         alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
 151     }
 152     *handlep = NULL;
 153 }
 154
 155 static void alsa_anal_close (snd_pcm_t **handlep, struct pollhlp *hlp)
 156 {
 157     alsa_fini_poll (hlp);
 158     alsa_anal_close1 (handlep);
 159 }
 160
 161 static int alsa_recover (snd_pcm_t *handle)
 162 {
 163     int err = snd_pcm_prepare (handle);
 164     if (err < 0) {
 165         alsa_logerr (err, "Failed to prepare handle %p\n", handle);
 166         return -1;
 167     }
 168     return 0;
 169 }
 170
 171 static int alsa_resume (snd_pcm_t *handle)
 172 {
 173     int err = snd_pcm_resume (handle);
 174     if (err < 0) {
 175         alsa_logerr (err, "Failed to resume handle %p\n", handle);
 176         return -1;
 177     }
 178     return 0;
 179 }
 180
 181 static void alsa_poll_handler (void *opaque)
 182 {
 183     int err, count;
 184     snd_pcm_state_t state;
 185     struct pollhlp *hlp = opaque;
 186     unsigned short revents;
 187
 188     count = poll (hlp->pfds, hlp->count, 0);
 189     if (count < 0) {
 190         dolog ("alsa_poll_handler: poll %s\n", strerror (errno));
 191         return;
 192     }
 193
 194     if (!count) {
 195         return;
 196     }
 197
 198     /* XXX: ALSA example uses initial count, not the one returned by
 199        poll, correct? */
 200     err = snd_pcm_poll_descriptors_revents (hlp->handle, hlp->pfds,
 201                                             hlp->count, &revents);
 202     if (err < 0) {
 203         alsa_logerr (err, "snd_pcm_poll_descriptors_revents");
 204         return;
 205     }
 206
 207     if (!(revents & hlp->mask)) {
 208         if (conf.verbose) {
 209             dolog ("revents = %d\n", revents);
 210         }
 211         return;
 212     }
 213
 214     state = snd_pcm_state (hlp->handle);
 215     switch (state) {
 216     case SND_PCM_STATE_XRUN:
 217         alsa_recover (hlp->handle);
 218         break;
 219
 220     case SND_PCM_STATE_SUSPENDED:
 221         alsa_resume (hlp->handle);
 222         break;
 223
 224     case SND_PCM_STATE_PREPARED:
 225         audio_run ("alsa run (prepared)");
 226         break;
 227
 228     case SND_PCM_STATE_RUNNING:
 229         audio_run ("alsa run (running)");
 230         break;
 231
 232     default:
 233         dolog ("Unexpected state %d\n", state);
 234     }
 235 }
 236
 237 static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask)
 238 {
 239     int i, count, err;
 240     struct pollfd *pfds;
 241
 242     count = snd_pcm_poll_descriptors_count (handle);
 243     if (count <= 0) {
 244         dolog ("Could not initialize poll mode\n"
 245                "Invalid number of poll descriptors %d\n", count);
 246         return -1;
 247     }
 248
 249     pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds));
 250     if (!pfds) {
 251         dolog ("Could not initialize poll mode\n");
 252         return -1;
 253     }
 254
 255     err = snd_pcm_poll_descriptors (handle, pfds, count);
 256     if (err < 0) {
 257         alsa_logerr (err, "Could not initialize poll mode\n"
 258                      "Could not obtain poll descriptors\n");
 259         qemu_free (pfds);
 260         return -1;
 261     }
 262
 263     for (i = 0; i < count; ++i) {
 264         if (pfds[i].events & POLLIN) {
 265             err = qemu_set_fd_handler (pfds[i].fd, alsa_poll_handler,
 266                                        NULL, hlp);
 267         }
 268         if (pfds[i].events & POLLOUT) {
 269             if (conf.verbose) {
 270                 dolog ("POLLOUT %d %d\n", i, pfds[i].fd);
 271             }
 272             err = qemu_set_fd_handler (pfds[i].fd, NULL,
 273                                        alsa_poll_handler, hlp);
 274         }
 275         if (conf.verbose) {
 276             dolog ("Set handler events=%#x index=%d fd=%d err=%d\n",
 277                    pfds[i].events, i, pfds[i].fd, err);
 278         }
 279
 280         if (err) {
 281             dolog ("Failed to set handler events=%#x index=%d fd=%d err=%d\n",
 282                    pfds[i].events, i, pfds[i].fd, err);
 283
 284             while (i--) {
 285                 qemu_set_fd_handler (pfds[i].fd, NULL, NULL, NULL);
 286             }
 287             qemu_free (pfds);
 288             return -1;
 289         }
 290     }
 291     hlp->pfds = pfds;
 292     hlp->count = count;
 293     hlp->handle = handle;
 294     hlp->mask = mask;
 295     return 0;
 296 }
 297
 298 static int alsa_poll_out (HWVoiceOut *hw)
 299 {
 300     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 301
 302     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLOUT);
 303 }
 304
 305 static int alsa_poll_in (HWVoiceIn *hw)
 306 {
 307     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 308
 309     return alsa_poll_helper (alsa->handle, &alsa->pollhlp, POLLIN);
 310 }
 311
 312 static int alsa_write (SWVoiceOut *sw, void *buf, int len)
 313 {
 314     return audio_pcm_sw_write (sw, buf, len);
 315 }
 316
 317 static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt)
 318 {
 319     switch (fmt) {
 320     case AUD_FMT_S8:
 321         return SND_PCM_FORMAT_S8;
 322
 323     case AUD_FMT_U8:
 324         return SND_PCM_FORMAT_U8;
 325
 326     case AUD_FMT_S16:
 327         return SND_PCM_FORMAT_S16_LE;
 328
 329     case AUD_FMT_U16:
 330         return SND_PCM_FORMAT_U16_LE;
 331
 332     case AUD_FMT_S32:
 333         return SND_PCM_FORMAT_S32_LE;
 334
 335     case AUD_FMT_U32:
 336         return SND_PCM_FORMAT_U32_LE;
 337
 338     default:
 339         dolog ("Internal logic error: Bad audio format %d\n", fmt);
 340 #ifdef DEBUG_AUDIO
 341         abort ();
 342 #endif
 343         return SND_PCM_FORMAT_U8;
 344     }
 345 }
 346
 347 static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt,
 348                            int *endianness)
 349 {
 350     switch (alsafmt) {
 351     case SND_PCM_FORMAT_S8:
 352         *endianness = 0;
 353         *fmt = AUD_FMT_S8;
 354         break;
 355
 356     case SND_PCM_FORMAT_U8:
 357         *endianness = 0;
 358         *fmt = AUD_FMT_U8;
 359         break;
 360
 361     case SND_PCM_FORMAT_S16_LE:
 362         *endianness = 0;
 363         *fmt = AUD_FMT_S16;
 364         break;
 365
 366     case SND_PCM_FORMAT_U16_LE:
 367         *endianness = 0;
 368         *fmt = AUD_FMT_U16;
 369         break;
 370
 371     case SND_PCM_FORMAT_S16_BE:
 372         *endianness = 1;
 373         *fmt = AUD_FMT_S16;
 374         break;
 375
 376     case SND_PCM_FORMAT_U16_BE:
 377         *endianness = 1;
 378         *fmt = AUD_FMT_U16;
 379         break;
 380
 381     case SND_PCM_FORMAT_S32_LE:
 382         *endianness = 0;
 383         *fmt = AUD_FMT_S32;
 384         break;
 385
 386     case SND_PCM_FORMAT_U32_LE:
 387         *endianness = 0;
 388         *fmt = AUD_FMT_U32;
 389         break;
 390
 391     case SND_PCM_FORMAT_S32_BE:
 392         *endianness = 1;
 393         *fmt = AUD_FMT_S32;
 394         break;
 395
 396     case SND_PCM_FORMAT_U32_BE:
 397         *endianness = 1;
 398         *fmt = AUD_FMT_U32;
 399         break;
 400
 401     default:
 402         dolog ("Unrecognized audio format %d\n", alsafmt);
 403         return -1;
 404     }
 405
 406     return 0;
 407 }
 408
 409 static void alsa_dump_info (struct alsa_params_req *req,
 410                             struct alsa_params_obt *obt)
 411 {
 412     dolog ("parameter | requested value | obtained value\n");
 413     dolog ("format    |      %10d |     %10d\n", req->fmt, obt->fmt);
 414     dolog ("channels  |      %10d |     %10d\n",
 415            req->nchannels, obt->nchannels);
 416     dolog ("frequency |      %10d |     %10d\n", req->freq, obt->freq);
 417     dolog ("============================================\n");
 418     dolog ("requested: buffer size %d period size %d\n",
 419            req->buffer_size, req->period_size);
 420     dolog ("obtained: samples %ld\n", obt->samples);
 421 }
 422
 423 static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
 424 {
 425     int err;
 426     snd_pcm_sw_params_t *sw_params;
 427
 428     snd_pcm_sw_params_alloca (&sw_params);
 429
 430     err = snd_pcm_sw_params_current (handle, sw_params);
 431     if (err < 0) {
 432         dolog ("Could not fully initialize DAC\n");
 433         alsa_logerr (err, "Failed to get current software parameters\n");
 434         return;
 435     }
 436
 437     err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
 438     if (err < 0) {
 439         dolog ("Could not fully initialize DAC\n");
 440         alsa_logerr (err, "Failed to set software threshold to %ld\n",
 441                      threshold);
 442         return;
 443     }
 444
 445     err = snd_pcm_sw_params (handle, sw_params);
 446     if (err < 0) {
 447         dolog ("Could not fully initialize DAC\n");
 448         alsa_logerr (err, "Failed to set software parameters\n");
 449         return;
 450     }
 451 }
 452
 453 static int alsa_open (int in, struct alsa_params_req *req,
 454                       struct alsa_params_obt *obt, snd_pcm_t **handlep)
 455 {
 456     snd_pcm_t *handle;
 457     snd_pcm_hw_params_t *hw_params;
 458     int err;
 459     int size_in_usec;
 460     unsigned int freq, nchannels;
 461     const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
 462     snd_pcm_uframes_t obt_buffer_size;
 463     const char *typ = in ? "ADC" : "DAC";
 464     snd_pcm_format_t obtfmt;
 465
 466     freq = req->freq;
 467     nchannels = req->nchannels;
 468     size_in_usec = req->size_in_usec;
 469
 470     snd_pcm_hw_params_alloca (&hw_params);
 471
 472     err = snd_pcm_open (
 473         &handle,
 474         pcm_name,
 475         in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
 476         SND_PCM_NONBLOCK
 477         );
 478     if (err < 0) {
 479         alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
 480         return -1;
 481     }
 482
 483     err = snd_pcm_hw_params_any (handle, hw_params);
 484     if (err < 0) {
 485         alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
 486         goto err;
 487     }
 488
 489     err = snd_pcm_hw_params_set_access (
 490         handle,
 491         hw_params,
 492         SND_PCM_ACCESS_RW_INTERLEAVED
 493         );
 494     if (err < 0) {
 495         alsa_logerr2 (err, typ, "Failed to set access type\n");
 496         goto err;
 497     }
 498
 499     err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
 500     if (err < 0 && conf.verbose) {
 501         alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
 502     }
 503
 504     err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
 505     if (err < 0) {
 506         alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
 507         goto err;
 508     }
 509
 510     err = snd_pcm_hw_params_set_channels_near (
 511         handle,
 512         hw_params,
 513         &nchannels
 514         );
 515     if (err < 0) {
 516         alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
 517                       req->nchannels);
 518         goto err;
 519     }
 520
 521     if (nchannels != 1 && nchannels != 2) {
 522         alsa_logerr2 (err, typ,
 523                       "Can not handle obtained number of channels %d\n",
 524                       nchannels);
 525         goto err;
 526     }
 527
 528     if (req->buffer_size) {
 529         unsigned long obt;
 530
 531         if (size_in_usec) {
 532             int dir = 0;
 533             unsigned int btime = req->buffer_size;
 534
 535             err = snd_pcm_hw_params_set_buffer_time_near (
 536                 handle,
 537                 hw_params,
 538                 &btime,
 539                 &dir
 540                 );
 541             obt = btime;
 542         }
 543         else {
 544             snd_pcm_uframes_t bsize = req->buffer_size;
 545
 546             err = snd_pcm_hw_params_set_buffer_size_near (
 547                 handle,
 548                 hw_params,
 549                 &bsize
 550                 );
 551             obt = bsize;
 552         }
 553         if (err < 0) {
 554             alsa_logerr2 (err, typ, "Failed to set buffer %s to %d\n",
 555                           size_in_usec ? "time" : "size", req->buffer_size);
 556             goto err;
 557         }
 558
 559         if ((req->override_mask & 2) && (obt - req->buffer_size))
 560             dolog ("Requested buffer %s %u was rejected, using %lu\n",
 561                    size_in_usec ? "time" : "size", req->buffer_size, obt);
 562     }
 563
 564     if (req->period_size) {
 565         unsigned long obt;
 566
 567         if (size_in_usec) {
 568             int dir = 0;
 569             unsigned int ptime = req->period_size;
 570
 571             err = snd_pcm_hw_params_set_period_time_near (
 572                 handle,
 573                 hw_params,
 574                 &ptime,
 575                 &dir
 576                 );
 577             obt = ptime;
 578         }
 579         else {
 580             int dir = 0;
 581             snd_pcm_uframes_t psize = req->period_size;
 582
 583             err = snd_pcm_hw_params_set_period_size_near (
 584                 handle,
 585                 hw_params,
 586                 &psize,
 587                 &dir
 588                 );
 589             obt = psize;
 590         }
 591
 592         if (err < 0) {
 593             alsa_logerr2 (err, typ, "Failed to set period %s to %d\n",
 594                           size_in_usec ? "time" : "size", req->period_size);
 595             goto err;
 596         }
 597
 598         if (((req->override_mask & 1) && (obt - req->period_size)))
 599             dolog ("Requested period %s %u was rejected, using %lu\n",
 600                    size_in_usec ? "time" : "size", req->period_size, obt);
 601     }
 602
 603     err = snd_pcm_hw_params (handle, hw_params);
 604     if (err < 0) {
 605         alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
 606         goto err;
 607     }
 608
 609     err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
 610     if (err < 0) {
 611         alsa_logerr2 (err, typ, "Failed to get buffer size\n");
 612         goto err;
 613     }
 614
 615     err = snd_pcm_hw_params_get_format (hw_params, &obtfmt);
 616     if (err < 0) {
 617         alsa_logerr2 (err, typ, "Failed to get format\n");
 618         goto err;
 619     }
 620
 621     if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) {
 622         dolog ("Invalid format was returned %d\n", obtfmt);
 623         goto err;
 624     }
 625
 626     err = snd_pcm_prepare (handle);
 627     if (err < 0) {
 628         alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle);
 629         goto err;
 630     }
 631
 632     if (!in && conf.threshold) {
 633         snd_pcm_uframes_t threshold;
 634         int bytes_per_sec;
 635
 636         bytes_per_sec = freq << (nchannels == 2);
 637
 638         switch (obt->fmt) {
 639         case AUD_FMT_S8:
 640         case AUD_FMT_U8:
 641             break;
 642
 643         case AUD_FMT_S16:
 644         case AUD_FMT_U16:
 645             bytes_per_sec <<= 1;
 646             break;
 647
 648         case AUD_FMT_S32:
 649         case AUD_FMT_U32:
 650             bytes_per_sec <<= 2;
 651             break;
 652         }
 653
 654         threshold = (conf.threshold * bytes_per_sec) / 1000;
 655         alsa_set_threshold (handle, threshold);
 656     }
 657
 658     obt->nchannels = nchannels;
 659     obt->freq = freq;
 660     obt->samples = obt_buffer_size;
 661
 662     *handlep = handle;
 663
 664     if (conf.verbose &&
 665         (obt->fmt != req->fmt ||
 666          obt->nchannels != req->nchannels ||
 667          obt->freq != req->freq)) {
 668         dolog ("Audio parameters for %s\n", typ);
 669         alsa_dump_info (req, obt);
 670     }
 671
 672 #ifdef DEBUG
 673     alsa_dump_info (req, obt);
 674 #endif
 675     return 0;
 676
 677  err:
 678     alsa_anal_close1 (&handle);
 679     return -1;
 680 }
 681
 682 static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
 683 {
 684     snd_pcm_sframes_t avail;
 685
 686     avail = snd_pcm_avail_update (handle);
 687     if (avail < 0) {
 688         if (avail == -EPIPE) {
 689             if (!alsa_recover (handle)) {
 690                 avail = snd_pcm_avail_update (handle);
 691             }
 692         }
 693
 694         if (avail < 0) {
 695             alsa_logerr (avail,
 696                          "Could not obtain number of available frames\n");
 697             return -1;
 698         }
 699     }
 700
 701     return avail;
 702 }
 703
 704 static void alsa_write_pending (ALSAVoiceOut *alsa)
 705 {
 706     HWVoiceOut *hw = &alsa->hw;
 707
 708     while (alsa->pending) {
 709         int left_till_end_samples = hw->samples - alsa->wpos;
 710         int len = audio_MIN (alsa->pending, left_till_end_samples);
 711         char *src = advance (alsa->pcm_buf, alsa->wpos << hw->info.shift);
 712
 713         while (len) {
 714             snd_pcm_sframes_t written;
 715
 716             written = snd_pcm_writei (alsa->handle, src, len);
 717
 718             if (written <= 0) {
 719                 switch (written) {
 720                 case 0:
 721                     if (conf.verbose) {
 722                         dolog ("Failed to write %d frames (wrote zero)\n", len);
 723                     }
 724                     return;
 725
 726                 case -EPIPE:
 727                     if (alsa_recover (alsa->handle)) {
 728                         alsa_logerr (written, "Failed to write %d frames\n",
 729                                      len);
 730                         return;
 731                     }
 732                     if (conf.verbose) {
 733                         dolog ("Recovering from playback xrun\n");
 734                     }
 735                     continue;
 736
 737                 case -ESTRPIPE:
 738                     /* stream is suspended and waiting for an
 739                        application recovery */
 740                     if (alsa_resume (alsa->handle)) {
 741                         alsa_logerr (written, "Failed to write %d frames\n",
 742                                      len);
 743                         return;
 744                     }
 745                     if (conf.verbose) {
 746                         dolog ("Resuming suspended output stream\n");
 747                     }
 748                     continue;
 749
 750                 case -EAGAIN:
 751                     return;
 752
 753                 default:
 754                     alsa_logerr (written, "Failed to write %d frames from %p\n",
 755                                  len, src);
 756                     return;
 757                 }
 758             }
 759
 760             alsa->wpos = (alsa->wpos + written) % hw->samples;
 761             alsa->pending -= written;
 762             len -= written;
 763         }
 764     }
 765 }
 766
 767 static int alsa_run_out (HWVoiceOut *hw, int live)
 768 {
 769     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 770     int decr;
 771     snd_pcm_sframes_t avail;
 772
 773     avail = alsa_get_avail (alsa->handle);
 774     if (avail < 0) {
 775         dolog ("Could not get number of available playback frames\n");
 776         return 0;
 777     }
 778
 779     decr = audio_MIN (live, avail);
 780     decr = audio_pcm_hw_clip_out (hw, alsa->pcm_buf, decr, alsa->pending);
 781     alsa->pending += decr;
 782     alsa_write_pending (alsa);
 783     return decr;
 784 }
 785
 786 static void alsa_fini_out (HWVoiceOut *hw)
 787 {
 788     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 789
 790     ldebug ("alsa_fini\n");
 791     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 792
 793     if (alsa->pcm_buf) {
 794         qemu_free (alsa->pcm_buf);
 795         alsa->pcm_buf = NULL;
 796     }
 797 }
 798
 799 static int alsa_init_out (HWVoiceOut *hw, struct audsettings *as)
 800 {
 801     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 802     struct alsa_params_req req;
 803     struct alsa_params_obt obt;
 804     snd_pcm_t *handle;
 805     struct audsettings obt_as;
 806
 807     req.fmt = aud_to_alsafmt (as->fmt);
 808     req.freq = as->freq;
 809     req.nchannels = as->nchannels;
 810     req.period_size = conf.period_size_out;
 811     req.buffer_size = conf.buffer_size_out;
 812     req.size_in_usec = conf.size_in_usec_out;
 813     req.override_mask =
 814         (conf.period_size_out_overridden ? 1 : 0) |
 815         (conf.buffer_size_out_overridden ? 2 : 0);
 816
 817     if (alsa_open (0, &req, &obt, &handle)) {
 818         return -1;
 819     }
 820
 821     obt_as.freq = obt.freq;
 822     obt_as.nchannels = obt.nchannels;
 823     obt_as.fmt = obt.fmt;
 824     obt_as.endianness = obt.endianness;
 825
 826     audio_pcm_init_info (&hw->info, &obt_as);
 827     hw->samples = obt.samples;
 828
 829     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
 830     if (!alsa->pcm_buf) {
 831         dolog ("Could not allocate DAC buffer (%d samples, each %d bytes)\n",
 832                hw->samples, 1 << hw->info.shift);
 833         alsa_anal_close1 (&handle);
 834         return -1;
 835     }
 836
 837     alsa->handle = handle;
 838     return 0;
 839 }
 840
 841 static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
 842 {
 843     int err;
 844
 845     if (pause) {
 846         err = snd_pcm_drop (handle);
 847         if (err < 0) {
 848             alsa_logerr (err, "Could not stop %s\n", typ);
 849             return -1;
 850         }
 851     }
 852     else {
 853         err = snd_pcm_prepare (handle);
 854         if (err < 0) {
 855             alsa_logerr (err, "Could not prepare handle for %s\n", typ);
 856             return -1;
 857         }
 858     }
 859
 860     return 0;
 861 }
 862
 863 static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
 864 {
 865     ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
 866
 867     switch (cmd) {
 868     case VOICE_ENABLE:
 869         {
 870             va_list ap;
 871             int poll_mode;
 872
 873             va_start (ap, cmd);
 874             poll_mode = va_arg (ap, int);
 875             va_end (ap);
 876
 877             ldebug ("enabling voice\n");
 878             if (poll_mode && alsa_poll_out (hw)) {
 879                 poll_mode = 0;
 880             }
 881             hw->poll_mode = poll_mode;
 882             return alsa_voice_ctl (alsa->handle, "playback", 0);
 883         }
 884
 885     case VOICE_DISABLE:
 886         ldebug ("disabling voice\n");
 887         return alsa_voice_ctl (alsa->handle, "playback", 1);
 888     }
 889
 890     return -1;
 891 }
 892
 893 static int alsa_init_in (HWVoiceIn *hw, struct audsettings *as)
 894 {
 895     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 896     struct alsa_params_req req;
 897     struct alsa_params_obt obt;
 898     snd_pcm_t *handle;
 899     struct audsettings obt_as;
 900
 901     req.fmt = aud_to_alsafmt (as->fmt);
 902     req.freq = as->freq;
 903     req.nchannels = as->nchannels;
 904     req.period_size = conf.period_size_in;
 905     req.buffer_size = conf.buffer_size_in;
 906     req.size_in_usec = conf.size_in_usec_in;
 907     req.override_mask =
 908         (conf.period_size_in_overridden ? 1 : 0) |
 909         (conf.buffer_size_in_overridden ? 2 : 0);
 910
 911     if (alsa_open (1, &req, &obt, &handle)) {
 912         return -1;
 913     }
 914
 915     obt_as.freq = obt.freq;
 916     obt_as.nchannels = obt.nchannels;
 917     obt_as.fmt = obt.fmt;
 918     obt_as.endianness = obt.endianness;
 919
 920     audio_pcm_init_info (&hw->info, &obt_as);
 921     hw->samples = obt.samples;
 922
 923     alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
 924     if (!alsa->pcm_buf) {
 925         dolog ("Could not allocate ADC buffer (%d samples, each %d bytes)\n",
 926                hw->samples, 1 << hw->info.shift);
 927         alsa_anal_close1 (&handle);
 928         return -1;
 929     }
 930
 931     alsa->handle = handle;
 932     return 0;
 933 }
 934
 935 static void alsa_fini_in (HWVoiceIn *hw)
 936 {
 937     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 938
 939     alsa_anal_close (&alsa->handle, &alsa->pollhlp);
 940
 941     if (alsa->pcm_buf) {
 942         qemu_free (alsa->pcm_buf);
 943         alsa->pcm_buf = NULL;
 944     }
 945 }
 946
 947 static int alsa_run_in (HWVoiceIn *hw)
 948 {
 949     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
 950     int hwshift = hw->info.shift;
 951     int i;
 952     int live = audio_pcm_hw_get_live_in (hw);
 953     int dead = hw->samples - live;
 954     int decr;
 955     struct {
 956         int add;
 957         int len;
 958     } bufs[2] = {
 959         { .add = hw->wpos, .len = 0 },
 960         { .add = 0,        .len = 0 }
 961     };
 962     snd_pcm_sframes_t avail;
 963     snd_pcm_uframes_t read_samples = 0;
 964
 965     if (!dead) {
 966         return 0;
 967     }
 968
 969     avail = alsa_get_avail (alsa->handle);
 970     if (avail < 0) {
 971         dolog ("Could not get number of captured frames\n");
 972         return 0;
 973     }
 974
 975     if (!avail) {
 976         snd_pcm_state_t state;
 977
 978         state = snd_pcm_state (alsa->handle);
 979         switch (state) {
 980         case SND_PCM_STATE_PREPARED:
 981             avail = hw->samples;
 982             break;
 983         case SND_PCM_STATE_SUSPENDED:
 984             /* stream is suspended and waiting for an application recovery */
 985             if (alsa_resume (alsa->handle)) {
 986                 dolog ("Failed to resume suspended input stream\n");
 987                 return 0;
 988             }
 989             if (conf.verbose) {
 990                 dolog ("Resuming suspended input stream\n");
 991             }
 992             break;
 993         default:
 994             if (conf.verbose) {
 995                 dolog ("No frames available and ALSA state is %d\n", state);
 996             }
 997             return 0;
 998         }
 999     }
1000
1001     decr = audio_MIN (dead, avail);
1002     if (!decr) {
1003         return 0;
1004     }
1005
1006     if (hw->wpos + decr > hw->samples) {
1007         bufs[0].len = (hw->samples - hw->wpos);
1008         bufs[1].len = (decr - (hw->samples - hw->wpos));
1009     }
1010     else {
1011         bufs[0].len = decr;
1012     }
1013
1014     for (i = 0; i < 2; ++i) {
1015         void *src;
1016         struct st_sample *dst;
1017         snd_pcm_sframes_t nread;
1018         snd_pcm_uframes_t len;
1019
1020         len = bufs[i].len;
1021
1022         src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
1023         dst = hw->conv_buf + bufs[i].add;
1024
1025         while (len) {
1026             nread = snd_pcm_readi (alsa->handle, src, len);
1027
1028             if (nread <= 0) {
1029                 switch (nread) {
1030                 case 0:
1031                     if (conf.verbose) {
1032                         dolog ("Failed to read %ld frames (read zero)\n", len);
1033                     }
1034                     goto exit;
1035
1036                 case -EPIPE:
1037                     if (alsa_recover (alsa->handle)) {
1038                         alsa_logerr (nread, "Failed to read %ld frames\n", len);
1039                         goto exit;
1040                     }
1041                     if (conf.verbose) {
1042                         dolog ("Recovering from capture xrun\n");
1043                     }
1044                     continue;
1045
1046                 case -EAGAIN:
1047                     goto exit;
1048
1049                 default:
1050                     alsa_logerr (
1051                         nread,
1052                         "Failed to read %ld frames from %p\n",
1053                         len,
1054                         src
1055                         );
1056                     goto exit;
1057                 }
1058             }
1059
1060             hw->conv (dst, src, nread, &nominal_volume);
1061
1062             src = advance (src, nread << hwshift);
1063             dst += nread;
1064
1065             read_samples += nread;
1066             len -= nread;
1067         }
1068     }
1069
1070  exit:
1071     hw->wpos = (hw->wpos + read_samples) % hw->samples;
1072     return read_samples;
1073 }
1074
1075 static int alsa_read (SWVoiceIn *sw, void *buf, int size)
1076 {
1077     return audio_pcm_sw_read (sw, buf, size);
1078 }
1079
1080 static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
1081 {
1082     ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
1083
1084     switch (cmd) {
1085     case VOICE_ENABLE:
1086         {
1087             va_list ap;
1088             int poll_mode;
1089
1090             va_start (ap, cmd);
1091             poll_mode = va_arg (ap, int);
1092             va_end (ap);
1093
1094             ldebug ("enabling voice\n");
1095             if (poll_mode && alsa_poll_in (hw)) {
1096                 poll_mode = 0;
1097             }
1098             hw->poll_mode = poll_mode;
1099
1100             return alsa_voice_ctl (alsa->handle, "capture", 0);
1101         }
1102
1103     case VOICE_DISABLE:
1104         ldebug ("disabling voice\n");
1105         if (hw->poll_mode) {
1106             hw->poll_mode = 0;
1107             alsa_fini_poll (&alsa->pollhlp);
1108         }
1109         return alsa_voice_ctl (alsa->handle, "capture", 1);
1110     }
1111
1112     return -1;
1113 }
1114
1115 static void *alsa_audio_init (void)
1116 {
1117     return &conf;
1118 }
1119
1120 static void alsa_audio_fini (void *opaque)
1121 {
1122     (void) opaque;
1123 }
1124
1125 static struct audio_option alsa_options[] = {
1126     {
1127         .name        = "DAC_SIZE_IN_USEC",
1128         .tag         = AUD_OPT_BOOL,
1129         .valp        = &conf.size_in_usec_out,
1130         .descr       = "DAC period/buffer size in microseconds (otherwise in frames)"
1131     },
1132     {
1133         .name        = "DAC_PERIOD_SIZE",
1134         .tag         = AUD_OPT_INT,
1135         .valp        = &conf.period_size_out,
1136         .descr       = "DAC period size (0 to go with system default)",
1137         .overriddenp = &conf.period_size_out_overridden
1138     },
1139     {
1140         .name        = "DAC_BUFFER_SIZE",
1141         .tag         = AUD_OPT_INT,
1142         .valp        = &conf.buffer_size_out,
1143         .descr       = "DAC buffer size (0 to go with system default)",
1144         .overriddenp = &conf.buffer_size_out_overridden
1145     },
1146     {
1147         .name        = "ADC_SIZE_IN_USEC",
1148         .tag         = AUD_OPT_BOOL,
1149         .valp        = &conf.size_in_usec_in,
1150         .descr       =
1151         "ADC period/buffer size in microseconds (otherwise in frames)"
1152     },
1153     {
1154         .name        = "ADC_PERIOD_SIZE",
1155         .tag         = AUD_OPT_INT,
1156         .valp        = &conf.period_size_in,
1157         .descr       = "ADC period size (0 to go with system default)",
1158         .overriddenp = &conf.period_size_in_overridden
1159     },
1160     {
1161         .name        = "ADC_BUFFER_SIZE",
1162         .tag         = AUD_OPT_INT,
1163         .valp        = &conf.buffer_size_in,
1164         .descr       = "ADC buffer size (0 to go with system default)",
1165         .overriddenp = &conf.buffer_size_in_overridden
1166     },
1167     {
1168         .name        = "THRESHOLD",
1169         .tag         = AUD_OPT_INT,
1170         .valp        = &conf.threshold,
1171         .descr       = "(undocumented)"
1172     },
1173     {
1174         .name        = "DAC_DEV",
1175         .tag         = AUD_OPT_STR,
1176         .valp        = &conf.pcm_name_out,
1177         .descr       = "DAC device name (for instance dmix)"
1178     },
1179     {
1180         .name        = "ADC_DEV",
1181         .tag         = AUD_OPT_STR,
1182         .valp        = &conf.pcm_name_in,
1183         .descr       = "ADC device name"
1184     },
1185     {
1186         .name        = "VERBOSE",
1187         .tag         = AUD_OPT_BOOL,
1188         .valp        = &conf.verbose,
1189         .descr       = "Behave in a more verbose way"
1190     },
1191     { /* End of list */ }
1192 };
1193
1194 static struct audio_pcm_ops alsa_pcm_ops = {
1195     .init_out = alsa_init_out,
1196     .fini_out = alsa_fini_out,
1197     .run_out  = alsa_run_out,
1198     .write    = alsa_write,
1199     .ctl_out  = alsa_ctl_out,
1200
1201     .init_in  = alsa_init_in,
1202     .fini_in  = alsa_fini_in,
1203     .run_in   = alsa_run_in,
1204     .read     = alsa_read,
1205     .ctl_in   = alsa_ctl_in,
1206 };
1207
1208 struct audio_driver alsa_audio_driver = {
1209     .name           = "alsa",
1210     .descr          = "ALSA http://www.alsa-project.org",
1211     .options        = alsa_options,
1212     .init           = alsa_audio_init,
1213     .fini           = alsa_audio_fini,
1214     .pcm_ops        = &alsa_pcm_ops,
1215     .can_be_default = 1,
1216     .max_voices_out = INT_MAX,
1217     .max_voices_in  = INT_MAX,
1218     .voice_size_out = sizeof (ALSAVoiceOut),
1219     .voice_size_in  = sizeof (ALSAVoiceIn)
1220 };