audio/audio.c

   1 /*
   2  * QEMU Audio subsystem
   3  *
   4  * Copyright (c) 2003-2005 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
  25 #include "qemu/osdep.h"
  26 #include "audio.h"
  27 #include "migration/vmstate.h"
  28 #include "monitor/monitor.h"
  29 #include "qemu/timer.h"
  30 #include "qapi/error.h"
  31 #include "qapi/qobject-input-visitor.h"
  32 #include "qapi/qapi-visit-audio.h"
  33 #include "qemu/cutils.h"
  34 #include "qemu/module.h"
  35 #include "qemu-common.h"
  36 #include "sysemu/replay.h"
  37 #include "sysemu/runstate.h"
  38 #include "ui/qemu-spice.h"
  39 #include "trace.h"
  40
  41 #define AUDIO_CAP "audio"
  42 #include "audio_int.h"
  43
  44 /* #define DEBUG_LIVE */
  45 /* #define DEBUG_OUT */
  46 /* #define DEBUG_CAPTURE */
  47 /* #define DEBUG_POLL */
  48
  49 #define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
  50
  51
  52 /* Order of CONFIG_AUDIO_DRIVERS is import.
  53    The 1st one is the one used by default, that is the reason
  54     that we generate the list.
  55 */
  56 const char *audio_prio_list[] = {
  57     "spice",
  58     CONFIG_AUDIO_DRIVERS
  59     "none",
  60     "wav",
  61     NULL
  62 };
  63
  64 static QLIST_HEAD(, audio_driver) audio_drivers;
  65 static AudiodevListHead audiodevs = QSIMPLEQ_HEAD_INITIALIZER(audiodevs);
  66
  67 void audio_driver_register(audio_driver *drv)
  68 {
  69     QLIST_INSERT_HEAD(&audio_drivers, drv, next);
  70 }
  71
  72 audio_driver *audio_driver_lookup(const char *name)
  73 {
  74     struct audio_driver *d;
  75
  76     QLIST_FOREACH(d, &audio_drivers, next) {
  77         if (strcmp(name, d->name) == 0) {
  78             return d;
  79         }
  80     }
  81
  82     audio_module_load_one(name);
  83     QLIST_FOREACH(d, &audio_drivers, next) {
  84         if (strcmp(name, d->name) == 0) {
  85             return d;
  86         }
  87     }
  88
  89     return NULL;
  90 }
  91
  92 static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
  93     QTAILQ_HEAD_INITIALIZER(audio_states);
  94
  95 const struct mixeng_volume nominal_volume = {
  96     .mute = 0,
  97 #ifdef FLOAT_MIXENG
  98     .r = 1.0,
  99     .l = 1.0,
 100 #else
 101     .r = 1ULL << 32,
 102     .l = 1ULL << 32,
 103 #endif
 104 };
 105
 106 static bool legacy_config = true;
 107
 108 int audio_bug (const char *funcname, int cond)
 109 {
 110     if (cond) {
 111         static int shown;
 112
 113         AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
 114         if (!shown) {
 115             shown = 1;
 116             AUD_log (NULL, "Save all your work and restart without audio\n");
 117             AUD_log (NULL, "I am sorry\n");
 118         }
 119         AUD_log (NULL, "Context:\n");
 120         abort();
 121     }
 122
 123     return cond;
 124 }
 125
 126 static inline int audio_bits_to_index (int bits)
 127 {
 128     switch (bits) {
 129     case 8:
 130         return 0;
 131
 132     case 16:
 133         return 1;
 134
 135     case 32:
 136         return 2;
 137
 138     default:
 139         audio_bug ("bits_to_index", 1);
 140         AUD_log (NULL, "invalid bits %d\n", bits);
 141         return 0;
 142     }
 143 }
 144
 145 void *audio_calloc (const char *funcname, int nmemb, size_t size)
 146 {
 147     int cond;
 148     size_t len;
 149
 150     len = nmemb * size;
 151     cond = !nmemb || !size;
 152     cond |= nmemb < 0;
 153     cond |= len < size;
 154
 155     if (audio_bug ("audio_calloc", cond)) {
 156         AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
 157                  funcname);
 158         AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
 159         return NULL;
 160     }
 161
 162     return g_malloc0 (len);
 163 }
 164
 165 void AUD_vlog (const char *cap, const char *fmt, va_list ap)
 166 {
 167     if (cap) {
 168         fprintf(stderr, "%s: ", cap);
 169     }
 170
 171     vfprintf(stderr, fmt, ap);
 172 }
 173
 174 void AUD_log (const char *cap, const char *fmt, ...)
 175 {
 176     va_list ap;
 177
 178     va_start (ap, fmt);
 179     AUD_vlog (cap, fmt, ap);
 180     va_end (ap);
 181 }
 182
 183 static void audio_print_settings (struct audsettings *as)
 184 {
 185     dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
 186
 187     switch (as->fmt) {
 188     case AUDIO_FORMAT_S8:
 189         AUD_log (NULL, "S8");
 190         break;
 191     case AUDIO_FORMAT_U8:
 192         AUD_log (NULL, "U8");
 193         break;
 194     case AUDIO_FORMAT_S16:
 195         AUD_log (NULL, "S16");
 196         break;
 197     case AUDIO_FORMAT_U16:
 198         AUD_log (NULL, "U16");
 199         break;
 200     case AUDIO_FORMAT_S32:
 201         AUD_log (NULL, "S32");
 202         break;
 203     case AUDIO_FORMAT_U32:
 204         AUD_log (NULL, "U32");
 205         break;
 206     case AUDIO_FORMAT_F32:
 207         AUD_log (NULL, "F32");
 208         break;
 209     default:
 210         AUD_log (NULL, "invalid(%d)", as->fmt);
 211         break;
 212     }
 213
 214     AUD_log (NULL, " endianness=");
 215     switch (as->endianness) {
 216     case 0:
 217         AUD_log (NULL, "little");
 218         break;
 219     case 1:
 220         AUD_log (NULL, "big");
 221         break;
 222     default:
 223         AUD_log (NULL, "invalid");
 224         break;
 225     }
 226     AUD_log (NULL, "\n");
 227 }
 228
 229 static int audio_validate_settings (struct audsettings *as)
 230 {
 231     int invalid;
 232
 233     invalid = as->nchannels < 1;
 234     invalid |= as->endianness != 0 && as->endianness != 1;
 235
 236     switch (as->fmt) {
 237     case AUDIO_FORMAT_S8:
 238     case AUDIO_FORMAT_U8:
 239     case AUDIO_FORMAT_S16:
 240     case AUDIO_FORMAT_U16:
 241     case AUDIO_FORMAT_S32:
 242     case AUDIO_FORMAT_U32:
 243     case AUDIO_FORMAT_F32:
 244         break;
 245     default:
 246         invalid = 1;
 247         break;
 248     }
 249
 250     invalid |= as->freq <= 0;
 251     return invalid ? -1 : 0;
 252 }
 253
 254 static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
 255 {
 256     int bits = 8;
 257     bool is_signed = false, is_float = false;
 258
 259     switch (as->fmt) {
 260     case AUDIO_FORMAT_S8:
 261         is_signed = true;
 262         /* fall through */
 263     case AUDIO_FORMAT_U8:
 264         break;
 265
 266     case AUDIO_FORMAT_S16:
 267         is_signed = true;
 268         /* fall through */
 269     case AUDIO_FORMAT_U16:
 270         bits = 16;
 271         break;
 272
 273     case AUDIO_FORMAT_F32:
 274         is_float = true;
 275         /* fall through */
 276     case AUDIO_FORMAT_S32:
 277         is_signed = true;
 278         /* fall through */
 279     case AUDIO_FORMAT_U32:
 280         bits = 32;
 281         break;
 282
 283     default:
 284         abort();
 285     }
 286     return info->freq == as->freq
 287         && info->nchannels == as->nchannels
 288         && info->is_signed == is_signed
 289         && info->is_float == is_float
 290         && info->bits == bits
 291         && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
 292 }
 293
 294 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
 295 {
 296     int bits = 8, mul;
 297     bool is_signed = false, is_float = false;
 298
 299     switch (as->fmt) {
 300     case AUDIO_FORMAT_S8:
 301         is_signed = true;
 302         /* fall through */
 303     case AUDIO_FORMAT_U8:
 304         mul = 1;
 305         break;
 306
 307     case AUDIO_FORMAT_S16:
 308         is_signed = true;
 309         /* fall through */
 310     case AUDIO_FORMAT_U16:
 311         bits = 16;
 312         mul = 2;
 313         break;
 314
 315     case AUDIO_FORMAT_F32:
 316         is_float = true;
 317         /* fall through */
 318     case AUDIO_FORMAT_S32:
 319         is_signed = true;
 320         /* fall through */
 321     case AUDIO_FORMAT_U32:
 322         bits = 32;
 323         mul = 4;
 324         break;
 325
 326     default:
 327         abort();
 328     }
 329
 330     info->freq = as->freq;
 331     info->bits = bits;
 332     info->is_signed = is_signed;
 333     info->is_float = is_float;
 334     info->nchannels = as->nchannels;
 335     info->bytes_per_frame = as->nchannels * mul;
 336     info->bytes_per_second = info->freq * info->bytes_per_frame;
 337     info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
 338 }
 339
 340 void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
 341 {
 342     if (!len) {
 343         return;
 344     }
 345
 346     if (info->is_signed || info->is_float) {
 347         memset(buf, 0x00, len * info->bytes_per_frame);
 348     } else {
 349         switch (info->bits) {
 350         case 8:
 351             memset(buf, 0x80, len * info->bytes_per_frame);
 352             break;
 353
 354         case 16:
 355             {
 356                 int i;
 357                 uint16_t *p = buf;
 358                 short s = INT16_MAX;
 359
 360                 if (info->swap_endianness) {
 361                     s = bswap16 (s);
 362                 }
 363
 364                 for (i = 0; i < len * info->nchannels; i++) {
 365                     p[i] = s;
 366                 }
 367             }
 368             break;
 369
 370         case 32:
 371             {
 372                 int i;
 373                 uint32_t *p = buf;
 374                 int32_t s = INT32_MAX;
 375
 376                 if (info->swap_endianness) {
 377                     s = bswap32 (s);
 378                 }
 379
 380                 for (i = 0; i < len * info->nchannels; i++) {
 381                     p[i] = s;
 382                 }
 383             }
 384             break;
 385
 386         default:
 387             AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
 388                      info->bits);
 389             break;
 390         }
 391     }
 392 }
 393
 394 /*
 395  * Capture
 396  */
 397 static void noop_conv (struct st_sample *dst, const void *src, int samples)
 398 {
 399     (void) src;
 400     (void) dst;
 401     (void) samples;
 402 }
 403
 404 static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
 405                                                         struct audsettings *as)
 406 {
 407     CaptureVoiceOut *cap;
 408
 409     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
 410         if (audio_pcm_info_eq (&cap->hw.info, as)) {
 411             return cap;
 412         }
 413     }
 414     return NULL;
 415 }
 416
 417 static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
 418 {
 419     struct capture_callback *cb;
 420
 421 #ifdef DEBUG_CAPTURE
 422     dolog ("notification %d sent\n", cmd);
 423 #endif
 424     for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
 425         cb->ops.notify (cb->opaque, cmd);
 426     }
 427 }
 428
 429 static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
 430 {
 431     if (cap->hw.enabled != enabled) {
 432         audcnotification_e cmd;
 433         cap->hw.enabled = enabled;
 434         cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
 435         audio_notify_capture (cap, cmd);
 436     }
 437 }
 438
 439 static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
 440 {
 441     HWVoiceOut *hw = &cap->hw;
 442     SWVoiceOut *sw;
 443     int enabled = 0;
 444
 445     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
 446         if (sw->active) {
 447             enabled = 1;
 448             break;
 449         }
 450     }
 451     audio_capture_maybe_changed (cap, enabled);
 452 }
 453
 454 static void audio_detach_capture (HWVoiceOut *hw)
 455 {
 456     SWVoiceCap *sc = hw->cap_head.lh_first;
 457
 458     while (sc) {
 459         SWVoiceCap *sc1 = sc->entries.le_next;
 460         SWVoiceOut *sw = &sc->sw;
 461         CaptureVoiceOut *cap = sc->cap;
 462         int was_active = sw->active;
 463
 464         if (sw->rate) {
 465             st_rate_stop (sw->rate);
 466             sw->rate = NULL;
 467         }
 468
 469         QLIST_REMOVE (sw, entries);
 470         QLIST_REMOVE (sc, entries);
 471         g_free (sc);
 472         if (was_active) {
 473             /* We have removed soft voice from the capture:
 474                this might have changed the overall status of the capture
 475                since this might have been the only active voice */
 476             audio_recalc_and_notify_capture (cap);
 477         }
 478         sc = sc1;
 479     }
 480 }
 481
 482 static int audio_attach_capture (HWVoiceOut *hw)
 483 {
 484     AudioState *s = hw->s;
 485     CaptureVoiceOut *cap;
 486
 487     audio_detach_capture (hw);
 488     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
 489         SWVoiceCap *sc;
 490         SWVoiceOut *sw;
 491         HWVoiceOut *hw_cap = &cap->hw;
 492
 493         sc = g_malloc0(sizeof(*sc));
 494
 495         sc->cap = cap;
 496         sw = &sc->sw;
 497         sw->hw = hw_cap;
 498         sw->info = hw->info;
 499         sw->empty = 1;
 500         sw->active = hw->enabled;
 501         sw->conv = noop_conv;
 502         sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
 503         sw->vol = nominal_volume;
 504         sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
 505         if (!sw->rate) {
 506             dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
 507             g_free (sw);
 508             return -1;
 509         }
 510         QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
 511         QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
 512 #ifdef DEBUG_CAPTURE
 513         sw->name = g_strdup_printf ("for %p %d,%d,%d",
 514                                     hw, sw->info.freq, sw->info.bits,
 515                                     sw->info.nchannels);
 516         dolog ("Added %s active = %d\n", sw->name, sw->active);
 517 #endif
 518         if (sw->active) {
 519             audio_capture_maybe_changed (cap, 1);
 520         }
 521     }
 522     return 0;
 523 }
 524
 525 /*
 526  * Hard voice (capture)
 527  */
 528 static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
 529 {
 530     SWVoiceIn *sw;
 531     size_t m = hw->total_samples_captured;
 532
 533     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
 534         if (sw->active) {
 535             m = MIN (m, sw->total_hw_samples_acquired);
 536         }
 537     }
 538     return m;
 539 }
 540
 541 static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
 542 {
 543     size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
 544     if (audio_bug(__func__, live > hw->conv_buf->size)) {
 545         dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
 546         return 0;
 547     }
 548     return live;
 549 }
 550
 551 static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples)
 552 {
 553     size_t conv = 0;
 554     STSampleBuffer *conv_buf = hw->conv_buf;
 555
 556     while (samples) {
 557         uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame);
 558         size_t proc = MIN(samples, conv_buf->size - conv_buf->pos);
 559
 560         hw->conv(conv_buf->samples + conv_buf->pos, src, proc);
 561         conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
 562         samples -= proc;
 563         conv += proc;
 564     }
 565
 566     return conv;
 567 }
 568
 569 /*
 570  * Soft voice (capture)
 571  */
 572 static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
 573 {
 574     HWVoiceIn *hw = sw->hw;
 575     size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
 576     struct st_sample *src, *dst = sw->buf;
 577
 578     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
 579     if (!live) {
 580         return 0;
 581     }
 582     if (audio_bug(__func__, live > hw->conv_buf->size)) {
 583         dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
 584         return 0;
 585     }
 586
 587     rpos = audio_ring_posb(hw->conv_buf->pos, live, hw->conv_buf->size);
 588
 589     samples = size / sw->info.bytes_per_frame;
 590
 591     swlim = (live * sw->ratio) >> 32;
 592     swlim = MIN (swlim, samples);
 593
 594     while (swlim) {
 595         src = hw->conv_buf->samples + rpos;
 596         if (hw->conv_buf->pos > rpos) {
 597             isamp = hw->conv_buf->pos - rpos;
 598         } else {
 599             isamp = hw->conv_buf->size - rpos;
 600         }
 601
 602         if (!isamp) {
 603             break;
 604         }
 605         osamp = swlim;
 606
 607         st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
 608         swlim -= osamp;
 609         rpos = (rpos + isamp) % hw->conv_buf->size;
 610         dst += osamp;
 611         ret += osamp;
 612         total += isamp;
 613     }
 614
 615     if (!hw->pcm_ops->volume_in) {
 616         mixeng_volume (sw->buf, ret, &sw->vol);
 617     }
 618
 619     sw->clip (buf, sw->buf, ret);
 620     sw->total_hw_samples_acquired += total;
 621     return ret * sw->info.bytes_per_frame;
 622 }
 623
 624 /*
 625  * Hard voice (playback)
 626  */
 627 static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
 628 {
 629     SWVoiceOut *sw;
 630     size_t m = SIZE_MAX;
 631     int nb_live = 0;
 632
 633     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
 634         if (sw->active || !sw->empty) {
 635             m = MIN (m, sw->total_hw_samples_mixed);
 636             nb_live += 1;
 637         }
 638     }
 639
 640     *nb_livep = nb_live;
 641     return m;
 642 }
 643
 644 static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
 645 {
 646     size_t smin;
 647     int nb_live1;
 648
 649     smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
 650     if (nb_live) {
 651         *nb_live = nb_live1;
 652     }
 653
 654     if (nb_live1) {
 655         size_t live = smin;
 656
 657         if (audio_bug(__func__, live > hw->mix_buf->size)) {
 658             dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
 659             return 0;
 660         }
 661         return live;
 662     }
 663     return 0;
 664 }
 665
 666 static size_t audio_pcm_hw_get_free(HWVoiceOut *hw)
 667 {
 668     return (hw->pcm_ops->buffer_get_free ? hw->pcm_ops->buffer_get_free(hw) :
 669             INT_MAX) / hw->info.bytes_per_frame;
 670 }
 671
 672 static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
 673 {
 674     size_t clipped = 0;
 675     size_t pos = hw->mix_buf->pos;
 676
 677     while (len) {
 678         st_sample *src = hw->mix_buf->samples + pos;
 679         uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
 680         size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
 681         size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
 682
 683         hw->clip(dst, src, samples_to_clip);
 684
 685         pos = (pos + samples_to_clip) % hw->mix_buf->size;
 686         len -= samples_to_clip;
 687         clipped += samples_to_clip;
 688     }
 689 }
 690
 691 /*
 692  * Soft voice (playback)
 693  */
 694 static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
 695 {
 696     size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, blck;
 697     size_t hw_free;
 698     size_t ret = 0, pos = 0, total = 0;
 699
 700     if (!sw) {
 701         return size;
 702     }
 703
 704     hwsamples = sw->hw->mix_buf->size;
 705
 706     live = sw->total_hw_samples_mixed;
 707     if (audio_bug(__func__, live > hwsamples)) {
 708         dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
 709         return 0;
 710     }
 711
 712     if (live == hwsamples) {
 713 #ifdef DEBUG_OUT
 714         dolog ("%s is full %zu\n", sw->name, live);
 715 #endif
 716         return 0;
 717     }
 718
 719     wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
 720
 721     dead = hwsamples - live;
 722     hw_free = audio_pcm_hw_get_free(sw->hw);
 723     hw_free = hw_free > live ? hw_free - live : 0;
 724     samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio;
 725     samples = MIN(samples, size / sw->info.bytes_per_frame);
 726     if (samples) {
 727         sw->conv(sw->buf, buf, samples);
 728
 729         if (!sw->hw->pcm_ops->volume_out) {
 730             mixeng_volume(sw->buf, samples, &sw->vol);
 731         }
 732     }
 733
 734     while (samples) {
 735         dead = hwsamples - live;
 736         left = hwsamples - wpos;
 737         blck = MIN (dead, left);
 738         if (!blck) {
 739             break;
 740         }
 741         isamp = samples;
 742         osamp = blck;
 743         st_rate_flow_mix (
 744             sw->rate,
 745             sw->buf + pos,
 746             sw->hw->mix_buf->samples + wpos,
 747             &isamp,
 748             &osamp
 749             );
 750         ret += isamp;
 751         samples -= isamp;
 752         pos += isamp;
 753         live += osamp;
 754         wpos = (wpos + osamp) % hwsamples;
 755         total += osamp;
 756     }
 757
 758     sw->total_hw_samples_mixed += total;
 759     sw->empty = sw->total_hw_samples_mixed == 0;
 760
 761 #ifdef DEBUG_OUT
 762     dolog (
 763         "%s: write size %zu ret %zu total sw %zu\n",
 764         SW_NAME (sw),
 765         size / sw->info.bytes_per_frame,
 766         ret,
 767         sw->total_hw_samples_mixed
 768         );
 769 #endif
 770
 771     return ret * sw->info.bytes_per_frame;
 772 }
 773
 774 #ifdef DEBUG_AUDIO
 775 static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
 776 {
 777     dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",
 778           cap, info->bits, info->is_signed, info->is_float, info->freq,
 779           info->nchannels);
 780 }
 781 #endif
 782
 783 #define DAC
 784 #include "audio_template.h"
 785 #undef DAC
 786 #include "audio_template.h"
 787
 788 /*
 789  * Timer
 790  */
 791 static int audio_is_timer_needed(AudioState *s)
 792 {
 793     HWVoiceIn *hwi = NULL;
 794     HWVoiceOut *hwo = NULL;
 795
 796     while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
 797         if (!hwo->poll_mode) {
 798             return 1;
 799         }
 800     }
 801     while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
 802         if (!hwi->poll_mode) {
 803             return 1;
 804         }
 805     }
 806     return 0;
 807 }
 808
 809 static void audio_reset_timer (AudioState *s)
 810 {
 811     if (audio_is_timer_needed(s)) {
 812         timer_mod_anticipate_ns(s->ts,
 813             qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
 814         if (!s->timer_running) {
 815             s->timer_running = true;
 816             s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 817             trace_audio_timer_start(s->period_ticks / SCALE_MS);
 818         }
 819     } else {
 820         timer_del(s->ts);
 821         if (s->timer_running) {
 822             s->timer_running = false;
 823             trace_audio_timer_stop();
 824         }
 825     }
 826 }
 827
 828 static void audio_timer (void *opaque)
 829 {
 830     int64_t now, diff;
 831     AudioState *s = opaque;
 832
 833     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 834     diff = now - s->timer_last;
 835     if (diff > s->period_ticks * 3 / 2) {
 836         trace_audio_timer_delayed(diff / SCALE_MS);
 837     }
 838     s->timer_last = now;
 839
 840     audio_run(s, "timer");
 841     audio_reset_timer(s);
 842 }
 843
 844 /*
 845  * Public API
 846  */
 847 size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
 848 {
 849     HWVoiceOut *hw;
 850
 851     if (!sw) {
 852         /* XXX: Consider options */
 853         return size;
 854     }
 855     hw = sw->hw;
 856
 857     if (!hw->enabled) {
 858         dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
 859         return 0;
 860     }
 861
 862     if (audio_get_pdo_out(hw->s->dev)->mixing_engine) {
 863         return audio_pcm_sw_write(sw, buf, size);
 864     } else {
 865         return hw->pcm_ops->write(hw, buf, size);
 866     }
 867 }
 868
 869 size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
 870 {
 871     HWVoiceIn *hw;
 872
 873     if (!sw) {
 874         /* XXX: Consider options */
 875         return size;
 876     }
 877     hw = sw->hw;
 878
 879     if (!hw->enabled) {
 880         dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
 881         return 0;
 882     }
 883
 884     if (audio_get_pdo_in(hw->s->dev)->mixing_engine) {
 885         return audio_pcm_sw_read(sw, buf, size);
 886     } else {
 887         return hw->pcm_ops->read(hw, buf, size);
 888     }
 889 }
 890
 891 int AUD_get_buffer_size_out(SWVoiceOut *sw)
 892 {
 893     return sw->hw->samples * sw->hw->info.bytes_per_frame;
 894 }
 895
 896 void AUD_set_active_out (SWVoiceOut *sw, int on)
 897 {
 898     HWVoiceOut *hw;
 899
 900     if (!sw) {
 901         return;
 902     }
 903
 904     hw = sw->hw;
 905     if (sw->active != on) {
 906         AudioState *s = sw->s;
 907         SWVoiceOut *temp_sw;
 908         SWVoiceCap *sc;
 909
 910         if (on) {
 911             hw->pending_disable = 0;
 912             if (!hw->enabled) {
 913                 hw->enabled = 1;
 914                 if (s->vm_running) {
 915                     if (hw->pcm_ops->enable_out) {
 916                         hw->pcm_ops->enable_out(hw, true);
 917                     }
 918                     audio_reset_timer (s);
 919                 }
 920             }
 921         } else {
 922             if (hw->enabled) {
 923                 int nb_active = 0;
 924
 925                 for (temp_sw = hw->sw_head.lh_first; temp_sw;
 926                      temp_sw = temp_sw->entries.le_next) {
 927                     nb_active += temp_sw->active != 0;
 928                 }
 929
 930                 hw->pending_disable = nb_active == 1;
 931             }
 932         }
 933
 934         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
 935             sc->sw.active = hw->enabled;
 936             if (hw->enabled) {
 937                 audio_capture_maybe_changed (sc->cap, 1);
 938             }
 939         }
 940         sw->active = on;
 941     }
 942 }
 943
 944 void AUD_set_active_in (SWVoiceIn *sw, int on)
 945 {
 946     HWVoiceIn *hw;
 947
 948     if (!sw) {
 949         return;
 950     }
 951
 952     hw = sw->hw;
 953     if (sw->active != on) {
 954         AudioState *s = sw->s;
 955         SWVoiceIn *temp_sw;
 956
 957         if (on) {
 958             if (!hw->enabled) {
 959                 hw->enabled = 1;
 960                 if (s->vm_running) {
 961                     if (hw->pcm_ops->enable_in) {
 962                         hw->pcm_ops->enable_in(hw, true);
 963                     }
 964                     audio_reset_timer (s);
 965                 }
 966             }
 967             sw->total_hw_samples_acquired = hw->total_samples_captured;
 968         } else {
 969             if (hw->enabled) {
 970                 int nb_active = 0;
 971
 972                 for (temp_sw = hw->sw_head.lh_first; temp_sw;
 973                      temp_sw = temp_sw->entries.le_next) {
 974                     nb_active += temp_sw->active != 0;
 975                 }
 976
 977                 if (nb_active == 1) {
 978                     hw->enabled = 0;
 979                     if (hw->pcm_ops->enable_in) {
 980                         hw->pcm_ops->enable_in(hw, false);
 981                     }
 982                 }
 983             }
 984         }
 985         sw->active = on;
 986     }
 987 }
 988
 989 static size_t audio_get_avail (SWVoiceIn *sw)
 990 {
 991     size_t live;
 992
 993     if (!sw) {
 994         return 0;
 995     }
 996
 997     live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
 998     if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
 999         dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
1000               sw->hw->conv_buf->size);
1001         return 0;
1002     }
1003
1004     ldebug (
1005         "%s: get_avail live %zu ret %" PRId64 "\n",
1006         SW_NAME (sw),
1007         live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
1008         );
1009
1010     return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
1011 }
1012
1013 static size_t audio_sw_bytes_free(SWVoiceOut *sw, size_t free)
1014 {
1015     return (((int64_t)free << 32) / sw->ratio) * sw->info.bytes_per_frame;
1016 }
1017
1018 static size_t audio_get_free(SWVoiceOut *sw)
1019 {
1020     size_t live, dead;
1021
1022     if (!sw) {
1023         return 0;
1024     }
1025
1026     live = sw->total_hw_samples_mixed;
1027
1028     if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
1029         dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
1030               sw->hw->mix_buf->size);
1031         return 0;
1032     }
1033
1034     dead = sw->hw->mix_buf->size - live;
1035
1036 #ifdef DEBUG_OUT
1037     dolog("%s: get_free live %zu dead %zu sw_bytes %zu\n",
1038           SW_NAME(sw), live, dead, audio_sw_bytes_free(sw, dead));
1039 #endif
1040
1041     return dead;
1042 }
1043
1044 static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
1045                                         size_t samples)
1046 {
1047     size_t n;
1048
1049     if (hw->enabled) {
1050         SWVoiceCap *sc;
1051
1052         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1053             SWVoiceOut *sw = &sc->sw;
1054             int rpos2 = rpos;
1055
1056             n = samples;
1057             while (n) {
1058                 size_t till_end_of_hw = hw->mix_buf->size - rpos2;
1059                 size_t to_write = MIN(till_end_of_hw, n);
1060                 size_t bytes = to_write * hw->info.bytes_per_frame;
1061                 size_t written;
1062
1063                 sw->buf = hw->mix_buf->samples + rpos2;
1064                 written = audio_pcm_sw_write (sw, NULL, bytes);
1065                 if (written - bytes) {
1066                     dolog("Could not mix %zu bytes into a capture "
1067                           "buffer, mixed %zu\n",
1068                           bytes, written);
1069                     break;
1070                 }
1071                 n -= to_write;
1072                 rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
1073             }
1074         }
1075     }
1076
1077     n = MIN(samples, hw->mix_buf->size - rpos);
1078     mixeng_clear(hw->mix_buf->samples + rpos, n);
1079     mixeng_clear(hw->mix_buf->samples, samples - n);
1080 }
1081
1082 static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
1083 {
1084     size_t clipped = 0;
1085
1086     while (live) {
1087         size_t size = live * hw->info.bytes_per_frame;
1088         size_t decr, proc;
1089         void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
1090
1091         if (size == 0) {
1092             break;
1093         }
1094
1095         decr = MIN(size / hw->info.bytes_per_frame, live);
1096         if (buf) {
1097             audio_pcm_hw_clip_out(hw, buf, decr);
1098         }
1099         proc = hw->pcm_ops->put_buffer_out(hw, buf,
1100                                            decr * hw->info.bytes_per_frame) /
1101             hw->info.bytes_per_frame;
1102
1103         live -= proc;
1104         clipped += proc;
1105         hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
1106
1107         if (proc == 0 || proc < decr) {
1108             break;
1109         }
1110     }
1111
1112     if (hw->pcm_ops->run_buffer_out) {
1113         hw->pcm_ops->run_buffer_out(hw);
1114     }
1115
1116     return clipped;
1117 }
1118
1119 static void audio_run_out (AudioState *s)
1120 {
1121     HWVoiceOut *hw = NULL;
1122     SWVoiceOut *sw;
1123
1124     if (!audio_get_pdo_out(s->dev)->mixing_engine) {
1125         while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
1126             /* there is exactly 1 sw for each hw with no mixeng */
1127             sw = hw->sw_head.lh_first;
1128
1129             if (hw->pending_disable) {
1130                 hw->enabled = 0;
1131                 hw->pending_disable = 0;
1132                 if (hw->pcm_ops->enable_out) {
1133                     hw->pcm_ops->enable_out(hw, false);
1134                 }
1135             }
1136
1137             if (sw->active) {
1138                 sw->callback.fn(sw->callback.opaque, INT_MAX);
1139             }
1140         }
1141         return;
1142     }
1143
1144     while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
1145         size_t played, live, prev_rpos;
1146         size_t hw_free = audio_pcm_hw_get_free(hw);
1147         int nb_live;
1148
1149         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1150             if (sw->active) {
1151                 size_t sw_free = audio_get_free(sw);
1152                 size_t free;
1153
1154                 if (hw_free > sw->total_hw_samples_mixed) {
1155                     free = audio_sw_bytes_free(sw,
1156                         MIN(sw_free, hw_free - sw->total_hw_samples_mixed));
1157                 } else {
1158                     free = 0;
1159                 }
1160                 if (free > 0) {
1161                     sw->callback.fn(sw->callback.opaque, free);
1162                 }
1163             }
1164         }
1165
1166         live = audio_pcm_hw_get_live_out (hw, &nb_live);
1167         if (!nb_live) {
1168             live = 0;
1169         }
1170
1171         if (audio_bug(__func__, live > hw->mix_buf->size)) {
1172             dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
1173             continue;
1174         }
1175
1176         if (hw->pending_disable && !nb_live) {
1177             SWVoiceCap *sc;
1178 #ifdef DEBUG_OUT
1179             dolog ("Disabling voice\n");
1180 #endif
1181             hw->enabled = 0;
1182             hw->pending_disable = 0;
1183             if (hw->pcm_ops->enable_out) {
1184                 hw->pcm_ops->enable_out(hw, false);
1185             }
1186             for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1187                 sc->sw.active = 0;
1188                 audio_recalc_and_notify_capture (sc->cap);
1189             }
1190             continue;
1191         }
1192
1193         if (!live) {
1194             if (hw->pcm_ops->run_buffer_out) {
1195                 hw->pcm_ops->run_buffer_out(hw);
1196             }
1197             continue;
1198         }
1199
1200         prev_rpos = hw->mix_buf->pos;
1201         played = audio_pcm_hw_run_out(hw, live);
1202         replay_audio_out(&played);
1203         if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
1204             dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
1205                   hw->mix_buf->pos, hw->mix_buf->size, played);
1206             hw->mix_buf->pos = 0;
1207         }
1208
1209 #ifdef DEBUG_OUT
1210         dolog("played=%zu\n", played);
1211 #endif
1212
1213         if (played) {
1214             hw->ts_helper += played;
1215             audio_capture_mix_and_clear (hw, prev_rpos, played);
1216         }
1217
1218         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1219             if (!sw->active && sw->empty) {
1220                 continue;
1221             }
1222
1223             if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) {
1224                 dolog("played=%zu sw->total_hw_samples_mixed=%zu\n",
1225                       played, sw->total_hw_samples_mixed);
1226                 played = sw->total_hw_samples_mixed;
1227             }
1228
1229             sw->total_hw_samples_mixed -= played;
1230
1231             if (!sw->total_hw_samples_mixed) {
1232                 sw->empty = 1;
1233             }
1234         }
1235     }
1236 }
1237
1238 static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
1239 {
1240     size_t conv = 0;
1241
1242     if (hw->pcm_ops->run_buffer_in) {
1243         hw->pcm_ops->run_buffer_in(hw);
1244     }
1245
1246     while (samples) {
1247         size_t proc;
1248         size_t size = samples * hw->info.bytes_per_frame;
1249         void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
1250
1251         assert(size % hw->info.bytes_per_frame == 0);
1252         if (size == 0) {
1253             break;
1254         }
1255
1256         proc = audio_pcm_hw_conv_in(hw, buf, size / hw->info.bytes_per_frame);
1257
1258         samples -= proc;
1259         conv += proc;
1260         hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
1261     }
1262
1263     return conv;
1264 }
1265
1266 static void audio_run_in (AudioState *s)
1267 {
1268     HWVoiceIn *hw = NULL;
1269
1270     if (!audio_get_pdo_in(s->dev)->mixing_engine) {
1271         while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
1272             /* there is exactly 1 sw for each hw with no mixeng */
1273             SWVoiceIn *sw = hw->sw_head.lh_first;
1274             if (sw->active) {
1275                 sw->callback.fn(sw->callback.opaque, INT_MAX);
1276             }
1277         }
1278         return;
1279     }
1280
1281     while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
1282         SWVoiceIn *sw;
1283         size_t captured = 0, min;
1284
1285         if (replay_mode != REPLAY_MODE_PLAY) {
1286             captured = audio_pcm_hw_run_in(
1287                 hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
1288         }
1289         replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
1290                         hw->conv_buf->size);
1291
1292         min = audio_pcm_hw_find_min_in (hw);
1293         hw->total_samples_captured += captured - min;
1294         hw->ts_helper += captured;
1295
1296         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1297             sw->total_hw_samples_acquired -= min;
1298
1299             if (sw->active) {
1300                 size_t avail;
1301
1302                 avail = audio_get_avail (sw);
1303                 if (avail > 0) {
1304                     sw->callback.fn (sw->callback.opaque, avail);
1305                 }
1306             }
1307         }
1308     }
1309 }
1310
1311 static void audio_run_capture (AudioState *s)
1312 {
1313     CaptureVoiceOut *cap;
1314
1315     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
1316         size_t live, rpos, captured;
1317         HWVoiceOut *hw = &cap->hw;
1318         SWVoiceOut *sw;
1319
1320         captured = live = audio_pcm_hw_get_live_out (hw, NULL);
1321         rpos = hw->mix_buf->pos;
1322         while (live) {
1323             size_t left = hw->mix_buf->size - rpos;
1324             size_t to_capture = MIN(live, left);
1325             struct st_sample *src;
1326             struct capture_callback *cb;
1327
1328             src = hw->mix_buf->samples + rpos;
1329             hw->clip (cap->buf, src, to_capture);
1330             mixeng_clear (src, to_capture);
1331
1332             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1333                 cb->ops.capture (cb->opaque, cap->buf,
1334                                  to_capture * hw->info.bytes_per_frame);
1335             }
1336             rpos = (rpos + to_capture) % hw->mix_buf->size;
1337             live -= to_capture;
1338         }
1339         hw->mix_buf->pos = rpos;
1340
1341         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1342             if (!sw->active && sw->empty) {
1343                 continue;
1344             }
1345
1346             if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
1347                 dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
1348                       captured, sw->total_hw_samples_mixed);
1349                 captured = sw->total_hw_samples_mixed;
1350             }
1351
1352             sw->total_hw_samples_mixed -= captured;
1353             sw->empty = sw->total_hw_samples_mixed == 0;
1354         }
1355     }
1356 }
1357
1358 void audio_run(AudioState *s, const char *msg)
1359 {
1360     audio_run_out(s);
1361     audio_run_in(s);
1362     audio_run_capture(s);
1363
1364 #ifdef DEBUG_POLL
1365     {
1366         static double prevtime;
1367         double currtime;
1368         struct timeval tv;
1369
1370         if (gettimeofday (&tv, NULL)) {
1371             perror ("audio_run: gettimeofday");
1372             return;
1373         }
1374
1375         currtime = tv.tv_sec + tv.tv_usec * 1e-6;
1376         dolog ("Elapsed since last %s: %f\n", msg, currtime - prevtime);
1377         prevtime = currtime;
1378     }
1379 #endif
1380 }
1381
1382 void audio_generic_run_buffer_in(HWVoiceIn *hw)
1383 {
1384     if (unlikely(!hw->buf_emul)) {
1385         hw->size_emul = hw->samples * hw->info.bytes_per_frame;
1386         hw->buf_emul = g_malloc(hw->size_emul);
1387         hw->pos_emul = hw->pending_emul = 0;
1388     }
1389
1390     while (hw->pending_emul < hw->size_emul) {
1391         size_t read_len = MIN(hw->size_emul - hw->pos_emul,
1392                               hw->size_emul - hw->pending_emul);
1393         size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul,
1394                                         read_len);
1395         hw->pending_emul += read;
1396         hw->pos_emul = (hw->pos_emul + read) % hw->size_emul;
1397         if (read < read_len) {
1398             break;
1399         }
1400     }
1401 }
1402
1403 void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
1404 {
1405     size_t start;
1406
1407     start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
1408     assert(start < hw->size_emul);
1409
1410     *size = MIN(*size, hw->pending_emul);
1411     *size = MIN(*size, hw->size_emul - start);
1412     return hw->buf_emul + start;
1413 }
1414
1415 void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
1416 {
1417     assert(size <= hw->pending_emul);
1418     hw->pending_emul -= size;
1419 }
1420
1421 size_t audio_generic_buffer_get_free(HWVoiceOut *hw)
1422 {
1423     if (hw->buf_emul) {
1424         return hw->size_emul - hw->pending_emul;
1425     } else {
1426         return hw->samples * hw->info.bytes_per_frame;
1427     }
1428 }
1429
1430 void audio_generic_run_buffer_out(HWVoiceOut *hw)
1431 {
1432     while (hw->pending_emul) {
1433         size_t write_len, written, start;
1434
1435         start = audio_ring_posb(hw->pos_emul, hw->pending_emul, hw->size_emul);
1436         assert(start < hw->size_emul);
1437
1438         write_len = MIN(hw->pending_emul, hw->size_emul - start);
1439
1440         written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
1441         hw->pending_emul -= written;
1442
1443         if (written < write_len) {
1444             break;
1445         }
1446     }
1447 }
1448
1449 void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
1450 {
1451     if (unlikely(!hw->buf_emul)) {
1452         hw->size_emul = hw->samples * hw->info.bytes_per_frame;
1453         hw->buf_emul = g_malloc(hw->size_emul);
1454         hw->pos_emul = hw->pending_emul = 0;
1455     }
1456
1457     *size = MIN(hw->size_emul - hw->pending_emul,
1458                 hw->size_emul - hw->pos_emul);
1459     return hw->buf_emul + hw->pos_emul;
1460 }
1461
1462 size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
1463 {
1464     assert(buf == hw->buf_emul + hw->pos_emul &&
1465            size + hw->pending_emul <= hw->size_emul);
1466
1467     hw->pending_emul += size;
1468     hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
1469
1470     return size;
1471 }
1472
1473 size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
1474 {
1475     size_t total = 0;
1476
1477     if (hw->pcm_ops->buffer_get_free) {
1478         size_t free = hw->pcm_ops->buffer_get_free(hw);
1479
1480         size = MIN(size, free);
1481     }
1482
1483     while (total < size) {
1484         size_t dst_size = size - total;
1485         size_t copy_size, proc;
1486         void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
1487
1488         if (dst_size == 0) {
1489             break;
1490         }
1491
1492         copy_size = MIN(size - total, dst_size);
1493         if (dst) {
1494             memcpy(dst, (char *)buf + total, copy_size);
1495         }
1496         proc = hw->pcm_ops->put_buffer_out(hw, dst, copy_size);
1497         total += proc;
1498
1499         if (proc == 0 || proc < copy_size) {
1500             break;
1501         }
1502     }
1503
1504     if (hw->pcm_ops->run_buffer_out) {
1505         hw->pcm_ops->run_buffer_out(hw);
1506     }
1507
1508     return total;
1509 }
1510
1511 size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
1512 {
1513     size_t total = 0;
1514
1515     if (hw->pcm_ops->run_buffer_in) {
1516         hw->pcm_ops->run_buffer_in(hw);
1517     }
1518
1519     while (total < size) {
1520         size_t src_size = size - total;
1521         void *src = hw->pcm_ops->get_buffer_in(hw, &src_size);
1522
1523         if (src_size == 0) {
1524             break;
1525         }
1526
1527         memcpy((char *)buf + total, src, src_size);
1528         hw->pcm_ops->put_buffer_in(hw, src, src_size);
1529         total += src_size;
1530     }
1531
1532     return total;
1533 }
1534
1535 static int audio_driver_init(AudioState *s, struct audio_driver *drv,
1536                              bool msg, Audiodev *dev)
1537 {
1538     s->drv_opaque = drv->init(dev);
1539
1540     if (s->drv_opaque) {
1541         if (!drv->pcm_ops->get_buffer_in) {
1542             drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in;
1543             drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in;
1544         }
1545         if (!drv->pcm_ops->get_buffer_out) {
1546             drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out;
1547             drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out;
1548         }
1549
1550         audio_init_nb_voices_out(s, drv);
1551         audio_init_nb_voices_in(s, drv);
1552         s->drv = drv;
1553         return 0;
1554     } else {
1555         if (msg) {
1556             dolog("Could not init `%s' audio driver\n", drv->name);
1557         }
1558         return -1;
1559     }
1560 }
1561
1562 static void audio_vm_change_state_handler (void *opaque, bool running,
1563                                            RunState state)
1564 {
1565     AudioState *s = opaque;
1566     HWVoiceOut *hwo = NULL;
1567     HWVoiceIn *hwi = NULL;
1568
1569     s->vm_running = running;
1570     while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
1571         if (hwo->pcm_ops->enable_out) {
1572             hwo->pcm_ops->enable_out(hwo, running);
1573         }
1574     }
1575
1576     while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
1577         if (hwi->pcm_ops->enable_in) {
1578             hwi->pcm_ops->enable_in(hwi, running);
1579         }
1580     }
1581     audio_reset_timer (s);
1582 }
1583
1584 static void free_audio_state(AudioState *s)
1585 {
1586     HWVoiceOut *hwo, *hwon;
1587     HWVoiceIn *hwi, *hwin;
1588
1589     QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
1590         SWVoiceCap *sc;
1591
1592         if (hwo->enabled && hwo->pcm_ops->enable_out) {
1593             hwo->pcm_ops->enable_out(hwo, false);
1594         }
1595         hwo->pcm_ops->fini_out (hwo);
1596
1597         for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1598             CaptureVoiceOut *cap = sc->cap;
1599             struct capture_callback *cb;
1600
1601             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1602                 cb->ops.destroy (cb->opaque);
1603             }
1604         }
1605         QLIST_REMOVE(hwo, entries);
1606     }
1607
1608     QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
1609         if (hwi->enabled && hwi->pcm_ops->enable_in) {
1610             hwi->pcm_ops->enable_in(hwi, false);
1611         }
1612         hwi->pcm_ops->fini_in (hwi);
1613         QLIST_REMOVE(hwi, entries);
1614     }
1615
1616     if (s->drv) {
1617         s->drv->fini (s->drv_opaque);
1618         s->drv = NULL;
1619     }
1620
1621     if (s->dev) {
1622         qapi_free_Audiodev(s->dev);
1623         s->dev = NULL;
1624     }
1625
1626     if (s->ts) {
1627         timer_free(s->ts);
1628         s->ts = NULL;
1629     }
1630
1631     g_free(s);
1632 }
1633
1634 void audio_cleanup(void)
1635 {
1636     while (!QTAILQ_EMPTY(&audio_states)) {
1637         AudioState *s = QTAILQ_FIRST(&audio_states);
1638         QTAILQ_REMOVE(&audio_states, s, list);
1639         free_audio_state(s);
1640     }
1641 }
1642
1643 static bool vmstate_audio_needed(void *opaque)
1644 {
1645     /*
1646      * Never needed, this vmstate only exists in case
1647      * an old qemu sends it to us.
1648      */
1649     return false;
1650 }
1651
1652 static const VMStateDescription vmstate_audio = {
1653     .name = "audio",
1654     .version_id = 1,
1655     .minimum_version_id = 1,
1656     .needed = vmstate_audio_needed,
1657     .fields = (VMStateField[]) {
1658         VMSTATE_END_OF_LIST()
1659     }
1660 };
1661
1662 static void audio_validate_opts(Audiodev *dev, Error **errp);
1663
1664 static AudiodevListEntry *audiodev_find(
1665     AudiodevListHead *head, const char *drvname)
1666 {
1667     AudiodevListEntry *e;
1668     QSIMPLEQ_FOREACH(e, head, next) {
1669         if (strcmp(AudiodevDriver_str(e->dev->driver), drvname) == 0) {
1670             return e;
1671         }
1672     }
1673
1674     return NULL;
1675 }
1676
1677 /*
1678  * if we have dev, this function was called because of an -audiodev argument =>
1679  *   initialize a new state with it
1680  * if dev == NULL => legacy implicit initialization, return the already created
1681  *   state or create a new one
1682  */
1683 static AudioState *audio_init(Audiodev *dev, const char *name)
1684 {
1685     static bool atexit_registered;
1686     size_t i;
1687     int done = 0;
1688     const char *drvname = NULL;
1689     VMChangeStateEntry *e;
1690     AudioState *s;
1691     struct audio_driver *driver;
1692     /* silence gcc warning about uninitialized variable */
1693     AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head);
1694
1695     if (using_spice) {
1696         /*
1697          * When using spice allow the spice audio driver being picked
1698          * as default.
1699          *
1700          * Temporary hack.  Using audio devices without explicit
1701          * audiodev= property is already deprecated.  Same goes for
1702          * the -soundhw switch.  Once this support gets finally
1703          * removed we can also drop the concept of a default audio
1704          * backend and this can go away.
1705          */
1706         driver = audio_driver_lookup("spice");
1707         if (driver) {
1708             driver->can_be_default = 1;
1709         }
1710     }
1711
1712     if (dev) {
1713         /* -audiodev option */
1714         legacy_config = false;
1715         drvname = AudiodevDriver_str(dev->driver);
1716     } else if (!QTAILQ_EMPTY(&audio_states)) {
1717         if (!legacy_config) {
1718             dolog("Device %s: audiodev default parameter is deprecated, please "
1719                   "specify audiodev=%s\n", name,
1720                   QTAILQ_FIRST(&audio_states)->dev->id);
1721         }
1722         return QTAILQ_FIRST(&audio_states);
1723     } else {
1724         /* legacy implicit initialization */
1725         head = audio_handle_legacy_opts();
1726         /*
1727          * In case of legacy initialization, all Audiodevs in the list will have
1728          * the same configuration (except the driver), so it doesn't matter which
1729          * one we chose.  We need an Audiodev to set up AudioState before we can
1730          * init a driver.  Also note that dev at this point is still in the
1731          * list.
1732          */
1733         dev = QSIMPLEQ_FIRST(&head)->dev;
1734         audio_validate_opts(dev, &error_abort);
1735     }
1736
1737     s = g_malloc0(sizeof(AudioState));
1738     s->dev = dev;
1739
1740     QLIST_INIT (&s->hw_head_out);
1741     QLIST_INIT (&s->hw_head_in);
1742     QLIST_INIT (&s->cap_head);
1743     if (!atexit_registered) {
1744         atexit(audio_cleanup);
1745         atexit_registered = true;
1746     }
1747     QTAILQ_INSERT_TAIL(&audio_states, s, list);
1748
1749     s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
1750
1751     s->nb_hw_voices_out = audio_get_pdo_out(dev)->voices;
1752     s->nb_hw_voices_in = audio_get_pdo_in(dev)->voices;
1753
1754     if (s->nb_hw_voices_out <= 0) {
1755         dolog ("Bogus number of playback voices %d, setting to 1\n",
1756                s->nb_hw_voices_out);
1757         s->nb_hw_voices_out = 1;
1758     }
1759
1760     if (s->nb_hw_voices_in <= 0) {
1761         dolog ("Bogus number of capture voices %d, setting to 0\n",
1762                s->nb_hw_voices_in);
1763         s->nb_hw_voices_in = 0;
1764     }
1765
1766     if (drvname) {
1767         driver = audio_driver_lookup(drvname);
1768         if (driver) {
1769             done = !audio_driver_init(s, driver, true, dev);
1770         } else {
1771             dolog ("Unknown audio driver `%s'\n", drvname);
1772         }
1773     } else {
1774         for (i = 0; audio_prio_list[i]; i++) {
1775             AudiodevListEntry *e = audiodev_find(&head, audio_prio_list[i]);
1776             driver = audio_driver_lookup(audio_prio_list[i]);
1777
1778             if (e && driver) {
1779                 s->dev = dev = e->dev;
1780                 audio_validate_opts(dev, &error_abort);
1781                 done = !audio_driver_init(s, driver, false, dev);
1782                 if (done) {
1783                     e->dev = NULL;
1784                     break;
1785                 }
1786             }
1787         }
1788     }
1789     audio_free_audiodev_list(&head);
1790
1791     if (!done) {
1792         driver = audio_driver_lookup("none");
1793         done = !audio_driver_init(s, driver, false, dev);
1794         assert(done);
1795         dolog("warning: Using timer based audio emulation\n");
1796     }
1797
1798     if (dev->timer_period <= 0) {
1799         s->period_ticks = 1;
1800     } else {
1801         s->period_ticks = dev->timer_period * (int64_t)SCALE_US;
1802     }
1803
1804     e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
1805     if (!e) {
1806         dolog ("warning: Could not register change state handler\n"
1807                "(Audio can continue looping even after stopping the VM)\n");
1808     }
1809
1810     QLIST_INIT (&s->card_head);
1811     vmstate_register (NULL, 0, &vmstate_audio, s);
1812     return s;
1813 }
1814
1815 void audio_free_audiodev_list(AudiodevListHead *head)
1816 {
1817     AudiodevListEntry *e;
1818     while ((e = QSIMPLEQ_FIRST(head))) {
1819         QSIMPLEQ_REMOVE_HEAD(head, next);
1820         qapi_free_Audiodev(e->dev);
1821         g_free(e);
1822     }
1823 }
1824
1825 void AUD_register_card (const char *name, QEMUSoundCard *card)
1826 {
1827     if (!card->state) {
1828         card->state = audio_init(NULL, name);
1829     }
1830
1831     card->name = g_strdup (name);
1832     memset (&card->entries, 0, sizeof (card->entries));
1833     QLIST_INSERT_HEAD(&card->state->card_head, card, entries);
1834 }
1835
1836 void AUD_remove_card (QEMUSoundCard *card)
1837 {
1838     QLIST_REMOVE (card, entries);
1839     g_free (card->name);
1840 }
1841
1842 static struct audio_pcm_ops capture_pcm_ops;
1843
1844 CaptureVoiceOut *AUD_add_capture(
1845     AudioState *s,
1846     struct audsettings *as,
1847     struct audio_capture_ops *ops,
1848     void *cb_opaque
1849     )
1850 {
1851     CaptureVoiceOut *cap;
1852     struct capture_callback *cb;
1853
1854     if (!s) {
1855         if (!legacy_config) {
1856             dolog("Capturing without setting an audiodev is deprecated\n");
1857         }
1858         s = audio_init(NULL, NULL);
1859     }
1860
1861     if (!audio_get_pdo_out(s->dev)->mixing_engine) {
1862         dolog("Can't capture with mixeng disabled\n");
1863         return NULL;
1864     }
1865
1866     if (audio_validate_settings (as)) {
1867         dolog ("Invalid settings were passed when trying to add capture\n");
1868         audio_print_settings (as);
1869         return NULL;
1870     }
1871
1872     cb = g_malloc0(sizeof(*cb));
1873     cb->ops = *ops;
1874     cb->opaque = cb_opaque;
1875
1876     cap = audio_pcm_capture_find_specific(s, as);
1877     if (cap) {
1878         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1879         return cap;
1880     } else {
1881         HWVoiceOut *hw;
1882         CaptureVoiceOut *cap;
1883
1884         cap = g_malloc0(sizeof(*cap));
1885
1886         hw = &cap->hw;
1887         hw->s = s;
1888         hw->pcm_ops = &capture_pcm_ops;
1889         QLIST_INIT (&hw->sw_head);
1890         QLIST_INIT (&cap->cb_head);
1891
1892         /* XXX find a more elegant way */
1893         hw->samples = 4096 * 4;
1894         audio_pcm_hw_alloc_resources_out(hw);
1895
1896         audio_pcm_init_info (&hw->info, as);
1897
1898         cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
1899
1900         if (hw->info.is_float) {
1901             hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
1902         } else {
1903             hw->clip = mixeng_clip
1904                 [hw->info.nchannels == 2]
1905                 [hw->info.is_signed]
1906                 [hw->info.swap_endianness]
1907                 [audio_bits_to_index(hw->info.bits)];
1908         }
1909
1910         QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
1911         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1912
1913         QLIST_FOREACH(hw, &s->hw_head_out, entries) {
1914             audio_attach_capture (hw);
1915         }
1916         return cap;
1917     }
1918 }
1919
1920 void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
1921 {
1922     struct capture_callback *cb;
1923
1924     for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1925         if (cb->opaque == cb_opaque) {
1926             cb->ops.destroy (cb_opaque);
1927             QLIST_REMOVE (cb, entries);
1928             g_free (cb);
1929
1930             if (!cap->cb_head.lh_first) {
1931                 SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
1932
1933                 while (sw) {
1934                     SWVoiceCap *sc = (SWVoiceCap *) sw;
1935 #ifdef DEBUG_CAPTURE
1936                     dolog ("freeing %s\n", sw->name);
1937 #endif
1938
1939                     sw1 = sw->entries.le_next;
1940                     if (sw->rate) {
1941                         st_rate_stop (sw->rate);
1942                         sw->rate = NULL;
1943                     }
1944                     QLIST_REMOVE (sw, entries);
1945                     QLIST_REMOVE (sc, entries);
1946                     g_free (sc);
1947                     sw = sw1;
1948                 }
1949                 QLIST_REMOVE (cap, entries);
1950                 g_free (cap->hw.mix_buf);
1951                 g_free (cap->buf);
1952                 g_free (cap);
1953             }
1954             return;
1955         }
1956     }
1957 }
1958
1959 void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
1960 {
1961     Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
1962     audio_set_volume_out(sw, &vol);
1963 }
1964
1965 void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
1966 {
1967     if (sw) {
1968         HWVoiceOut *hw = sw->hw;
1969
1970         sw->vol.mute = vol->mute;
1971         sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
1972         sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] /
1973             255;
1974
1975         if (hw->pcm_ops->volume_out) {
1976             hw->pcm_ops->volume_out(hw, vol);
1977         }
1978     }
1979 }
1980
1981 void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
1982 {
1983     Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
1984     audio_set_volume_in(sw, &vol);
1985 }
1986
1987 void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
1988 {
1989     if (sw) {
1990         HWVoiceIn *hw = sw->hw;
1991
1992         sw->vol.mute = vol->mute;
1993         sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
1994         sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] /
1995             255;
1996
1997         if (hw->pcm_ops->volume_in) {
1998             hw->pcm_ops->volume_in(hw, vol);
1999         }
2000     }
2001 }
2002
2003 void audio_create_pdos(Audiodev *dev)
2004 {
2005     switch (dev->driver) {
2006 #define CASE(DRIVER, driver, pdo_name)                              \
2007     case AUDIODEV_DRIVER_##DRIVER:                                  \
2008         if (!dev->u.driver.has_in) {                                \
2009             dev->u.driver.in = g_malloc0(                           \
2010                 sizeof(Audiodev##pdo_name##PerDirectionOptions));   \
2011             dev->u.driver.has_in = true;                            \
2012         }                                                           \
2013         if (!dev->u.driver.has_out) {                               \
2014             dev->u.driver.out = g_malloc0(                          \
2015                 sizeof(Audiodev##pdo_name##PerDirectionOptions));   \
2016             dev->u.driver.has_out = true;                           \
2017         }                                                           \
2018         break
2019
2020         CASE(NONE, none, );
2021         CASE(ALSA, alsa, Alsa);
2022         CASE(COREAUDIO, coreaudio, Coreaudio);
2023         CASE(DBUS, dbus, );
2024         CASE(DSOUND, dsound, );
2025         CASE(JACK, jack, Jack);
2026         CASE(OSS, oss, Oss);
2027         CASE(PA, pa, Pa);
2028         CASE(SDL, sdl, Sdl);
2029         CASE(SPICE, spice, );
2030         CASE(WAV, wav, );
2031
2032     case AUDIODEV_DRIVER__MAX:
2033         abort();
2034     };
2035 }
2036
2037 static void audio_validate_per_direction_opts(
2038     AudiodevPerDirectionOptions *pdo, Error **errp)
2039 {
2040     if (!pdo->has_mixing_engine) {
2041         pdo->has_mixing_engine = true;
2042         pdo->mixing_engine = true;
2043     }
2044     if (!pdo->has_fixed_settings) {
2045         pdo->has_fixed_settings = true;
2046         pdo->fixed_settings = pdo->mixing_engine;
2047     }
2048     if (!pdo->fixed_settings &&
2049         (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
2050         error_setg(errp,
2051                    "You can't use frequency, channels or format with fixed-settings=off");
2052         return;
2053     }
2054     if (!pdo->mixing_engine && pdo->fixed_settings) {
2055         error_setg(errp, "You can't use fixed-settings without mixeng");
2056         return;
2057     }
2058
2059     if (!pdo->has_frequency) {
2060         pdo->has_frequency = true;
2061         pdo->frequency = 44100;
2062     }
2063     if (!pdo->has_channels) {
2064         pdo->has_channels = true;
2065         pdo->channels = 2;
2066     }
2067     if (!pdo->has_voices) {
2068         pdo->has_voices = true;
2069         pdo->voices = pdo->mixing_engine ? 1 : INT_MAX;
2070     }
2071     if (!pdo->has_format) {
2072         pdo->has_format = true;
2073         pdo->format = AUDIO_FORMAT_S16;
2074     }
2075 }
2076
2077 static void audio_validate_opts(Audiodev *dev, Error **errp)
2078 {
2079     Error *err = NULL;
2080
2081     audio_create_pdos(dev);
2082
2083     audio_validate_per_direction_opts(audio_get_pdo_in(dev), &err);
2084     if (err) {
2085         error_propagate(errp, err);
2086         return;
2087     }
2088
2089     audio_validate_per_direction_opts(audio_get_pdo_out(dev), &err);
2090     if (err) {
2091         error_propagate(errp, err);
2092         return;
2093     }
2094
2095     if (!dev->has_timer_period) {
2096         dev->has_timer_period = true;
2097         dev->timer_period = 10000; /* 100Hz -> 10ms */
2098     }
2099 }
2100
2101 void audio_parse_option(const char *opt)
2102 {
2103     AudiodevListEntry *e;
2104     Audiodev *dev = NULL;
2105
2106     Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal);
2107     visit_type_Audiodev(v, NULL, &dev, &error_fatal);
2108     visit_free(v);
2109
2110     audio_validate_opts(dev, &error_fatal);
2111
2112     e = g_malloc0(sizeof(AudiodevListEntry));
2113     e->dev = dev;
2114     QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next);
2115 }
2116
2117 void audio_init_audiodevs(void)
2118 {
2119     AudiodevListEntry *e;
2120
2121     QSIMPLEQ_FOREACH(e, &audiodevs, next) {
2122         audio_init(e->dev, NULL);
2123     }
2124 }
2125
2126 audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
2127 {
2128     return (audsettings) {
2129         .freq = pdo->frequency,
2130         .nchannels = pdo->channels,
2131         .fmt = pdo->format,
2132         .endianness = AUDIO_HOST_ENDIANNESS,
2133     };
2134 }
2135
2136 int audioformat_bytes_per_sample(AudioFormat fmt)
2137 {
2138     switch (fmt) {
2139     case AUDIO_FORMAT_U8:
2140     case AUDIO_FORMAT_S8:
2141         return 1;
2142
2143     case AUDIO_FORMAT_U16:
2144     case AUDIO_FORMAT_S16:
2145         return 2;
2146
2147     case AUDIO_FORMAT_U32:
2148     case AUDIO_FORMAT_S32:
2149     case AUDIO_FORMAT_F32:
2150         return 4;
2151
2152     case AUDIO_FORMAT__MAX:
2153         ;
2154     }
2155     abort();
2156 }
2157
2158
2159 /* frames = freq * usec / 1e6 */
2160 int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
2161                         audsettings *as, int def_usecs)
2162 {
2163     uint64_t usecs = pdo->has_buffer_length ? pdo->buffer_length : def_usecs;
2164     return (as->freq * usecs + 500000) / 1000000;
2165 }
2166
2167 /* samples = channels * frames = channels * freq * usec / 1e6 */
2168 int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
2169                          audsettings *as, int def_usecs)
2170 {
2171     return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
2172 }
2173
2174 /*
2175  * bytes = bytes_per_sample * samples =
2176  *     bytes_per_sample * channels * freq * usec / 1e6
2177  */
2178 int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
2179                        audsettings *as, int def_usecs)
2180 {
2181     return audio_buffer_samples(pdo, as, def_usecs) *
2182         audioformat_bytes_per_sample(as->fmt);
2183 }
2184
2185 AudioState *audio_state_by_name(const char *name)
2186 {
2187     AudioState *s;
2188     QTAILQ_FOREACH(s, &audio_states, list) {
2189         assert(s->dev);
2190         if (strcmp(name, s->dev->id) == 0) {
2191             return s;
2192         }
2193     }
2194     return NULL;
2195 }
2196
2197 const char *audio_get_id(QEMUSoundCard *card)
2198 {
2199     if (card->state) {
2200         assert(card->state->dev);
2201         return card->state->dev->id;
2202     } else {
2203         return "";
2204     }
2205 }
2206
2207 const char *audio_application_name(void)
2208 {
2209     const char *vm_name;
2210
2211     vm_name = qemu_get_vm_name();
2212     return vm_name ? vm_name : "qemu";
2213 }
2214
2215 void audio_rate_start(RateCtl *rate)
2216 {
2217     memset(rate, 0, sizeof(RateCtl));
2218     rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
2219 }
2220
2221 size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
2222                             size_t bytes_avail)
2223 {
2224     int64_t now;
2225     int64_t ticks;
2226     int64_t bytes;
2227     int64_t samples;
2228     size_t ret;
2229
2230     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
2231     ticks = now - rate->start_ticks;
2232     bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
2233     samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
2234     if (samples < 0 || samples > 65536) {
2235         AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
2236         audio_rate_start(rate);
2237         samples = 0;
2238     }
2239
2240     ret = MIN(samples * info->bytes_per_frame, bytes_avail);
2241     rate->bytes_sent += ret;
2242     return ret;
2243 }