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 "sysemu/replay.h"
  36 #include "sysemu/runstate.h"
  37 #include "trace.h"
  38
  39 #define AUDIO_CAP "audio"
  40 #include "audio_int.h"
  41
  42 /* #define DEBUG_LIVE */
  43 /* #define DEBUG_OUT */
  44 /* #define DEBUG_CAPTURE */
  45 /* #define DEBUG_POLL */
  46
  47 #define SW_NAME(sw) (sw)->name ? (sw)->name : "unknown"
  48
  49
  50 /* Order of CONFIG_AUDIO_DRIVERS is import.
  51    The 1st one is the one used by default, that is the reason
  52     that we generate the list.
  53 */
  54 const char *audio_prio_list[] = {
  55     "spice",
  56     CONFIG_AUDIO_DRIVERS
  57     "none",
  58     "wav",
  59     NULL
  60 };
  61
  62 static QLIST_HEAD(, audio_driver) audio_drivers;
  63 static AudiodevListHead audiodevs = QSIMPLEQ_HEAD_INITIALIZER(audiodevs);
  64
  65 void audio_driver_register(audio_driver *drv)
  66 {
  67     QLIST_INSERT_HEAD(&audio_drivers, drv, next);
  68 }
  69
  70 audio_driver *audio_driver_lookup(const char *name)
  71 {
  72     struct audio_driver *d;
  73
  74     QLIST_FOREACH(d, &audio_drivers, next) {
  75         if (strcmp(name, d->name) == 0) {
  76             return d;
  77         }
  78     }
  79
  80     audio_module_load_one(name);
  81     QLIST_FOREACH(d, &audio_drivers, next) {
  82         if (strcmp(name, d->name) == 0) {
  83             return d;
  84         }
  85     }
  86
  87     return NULL;
  88 }
  89
  90 static QTAILQ_HEAD(AudioStateHead, AudioState) audio_states =
  91     QTAILQ_HEAD_INITIALIZER(audio_states);
  92
  93 const struct mixeng_volume nominal_volume = {
  94     .mute = 0,
  95 #ifdef FLOAT_MIXENG
  96     .r = 1.0,
  97     .l = 1.0,
  98 #else
  99     .r = 1ULL << 32,
 100     .l = 1ULL << 32,
 101 #endif
 102 };
 103
 104 static bool legacy_config = true;
 105
 106 #ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
 107 #error No its not
 108 #else
 109 int audio_bug (const char *funcname, int cond)
 110 {
 111     if (cond) {
 112         static int shown;
 113
 114         AUD_log (NULL, "A bug was just triggered in %s\n", funcname);
 115         if (!shown) {
 116             shown = 1;
 117             AUD_log (NULL, "Save all your work and restart without audio\n");
 118             AUD_log (NULL, "I am sorry\n");
 119         }
 120         AUD_log (NULL, "Context:\n");
 121
 122 #if defined AUDIO_BREAKPOINT_ON_BUG
 123 #  if defined HOST_I386
 124 #    if defined __GNUC__
 125         __asm__ ("int3");
 126 #    elif defined _MSC_VER
 127         _asm _emit 0xcc;
 128 #    else
 129         abort ();
 130 #    endif
 131 #  else
 132         abort ();
 133 #  endif
 134 #endif
 135     }
 136
 137     return cond;
 138 }
 139 #endif
 140
 141 static inline int audio_bits_to_index (int bits)
 142 {
 143     switch (bits) {
 144     case 8:
 145         return 0;
 146
 147     case 16:
 148         return 1;
 149
 150     case 32:
 151         return 2;
 152
 153     default:
 154         audio_bug ("bits_to_index", 1);
 155         AUD_log (NULL, "invalid bits %d\n", bits);
 156         return 0;
 157     }
 158 }
 159
 160 void *audio_calloc (const char *funcname, int nmemb, size_t size)
 161 {
 162     int cond;
 163     size_t len;
 164
 165     len = nmemb * size;
 166     cond = !nmemb || !size;
 167     cond |= nmemb < 0;
 168     cond |= len < size;
 169
 170     if (audio_bug ("audio_calloc", cond)) {
 171         AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n",
 172                  funcname);
 173         AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len);
 174         return NULL;
 175     }
 176
 177     return g_malloc0 (len);
 178 }
 179
 180 void AUD_vlog (const char *cap, const char *fmt, va_list ap)
 181 {
 182     if (cap) {
 183         fprintf(stderr, "%s: ", cap);
 184     }
 185
 186     vfprintf(stderr, fmt, ap);
 187 }
 188
 189 void AUD_log (const char *cap, const char *fmt, ...)
 190 {
 191     va_list ap;
 192
 193     va_start (ap, fmt);
 194     AUD_vlog (cap, fmt, ap);
 195     va_end (ap);
 196 }
 197
 198 static void audio_print_settings (struct audsettings *as)
 199 {
 200     dolog ("frequency=%d nchannels=%d fmt=", as->freq, as->nchannels);
 201
 202     switch (as->fmt) {
 203     case AUDIO_FORMAT_S8:
 204         AUD_log (NULL, "S8");
 205         break;
 206     case AUDIO_FORMAT_U8:
 207         AUD_log (NULL, "U8");
 208         break;
 209     case AUDIO_FORMAT_S16:
 210         AUD_log (NULL, "S16");
 211         break;
 212     case AUDIO_FORMAT_U16:
 213         AUD_log (NULL, "U16");
 214         break;
 215     case AUDIO_FORMAT_S32:
 216         AUD_log (NULL, "S32");
 217         break;
 218     case AUDIO_FORMAT_U32:
 219         AUD_log (NULL, "U32");
 220         break;
 221     case AUDIO_FORMAT_F32:
 222         AUD_log (NULL, "F32");
 223         break;
 224     default:
 225         AUD_log (NULL, "invalid(%d)", as->fmt);
 226         break;
 227     }
 228
 229     AUD_log (NULL, " endianness=");
 230     switch (as->endianness) {
 231     case 0:
 232         AUD_log (NULL, "little");
 233         break;
 234     case 1:
 235         AUD_log (NULL, "big");
 236         break;
 237     default:
 238         AUD_log (NULL, "invalid");
 239         break;
 240     }
 241     AUD_log (NULL, "\n");
 242 }
 243
 244 static int audio_validate_settings (struct audsettings *as)
 245 {
 246     int invalid;
 247
 248     invalid = as->nchannels < 1;
 249     invalid |= as->endianness != 0 && as->endianness != 1;
 250
 251     switch (as->fmt) {
 252     case AUDIO_FORMAT_S8:
 253     case AUDIO_FORMAT_U8:
 254     case AUDIO_FORMAT_S16:
 255     case AUDIO_FORMAT_U16:
 256     case AUDIO_FORMAT_S32:
 257     case AUDIO_FORMAT_U32:
 258     case AUDIO_FORMAT_F32:
 259         break;
 260     default:
 261         invalid = 1;
 262         break;
 263     }
 264
 265     invalid |= as->freq <= 0;
 266     return invalid ? -1 : 0;
 267 }
 268
 269 static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)
 270 {
 271     int bits = 8;
 272     bool is_signed = false, is_float = false;
 273
 274     switch (as->fmt) {
 275     case AUDIO_FORMAT_S8:
 276         is_signed = true;
 277         /* fall through */
 278     case AUDIO_FORMAT_U8:
 279         break;
 280
 281     case AUDIO_FORMAT_S16:
 282         is_signed = true;
 283         /* fall through */
 284     case AUDIO_FORMAT_U16:
 285         bits = 16;
 286         break;
 287
 288     case AUDIO_FORMAT_F32:
 289         is_float = true;
 290         /* fall through */
 291     case AUDIO_FORMAT_S32:
 292         is_signed = true;
 293         /* fall through */
 294     case AUDIO_FORMAT_U32:
 295         bits = 32;
 296         break;
 297
 298     default:
 299         abort();
 300     }
 301     return info->freq == as->freq
 302         && info->nchannels == as->nchannels
 303         && info->is_signed == is_signed
 304         && info->is_float == is_float
 305         && info->bits == bits
 306         && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);
 307 }
 308
 309 void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)
 310 {
 311     int bits = 8, mul;
 312     bool is_signed = false, is_float = false;
 313
 314     switch (as->fmt) {
 315     case AUDIO_FORMAT_S8:
 316         is_signed = true;
 317         /* fall through */
 318     case AUDIO_FORMAT_U8:
 319         mul = 1;
 320         break;
 321
 322     case AUDIO_FORMAT_S16:
 323         is_signed = true;
 324         /* fall through */
 325     case AUDIO_FORMAT_U16:
 326         bits = 16;
 327         mul = 2;
 328         break;
 329
 330     case AUDIO_FORMAT_F32:
 331         is_float = true;
 332         /* fall through */
 333     case AUDIO_FORMAT_S32:
 334         is_signed = true;
 335         /* fall through */
 336     case AUDIO_FORMAT_U32:
 337         bits = 32;
 338         mul = 4;
 339         break;
 340
 341     default:
 342         abort();
 343     }
 344
 345     info->freq = as->freq;
 346     info->bits = bits;
 347     info->is_signed = is_signed;
 348     info->is_float = is_float;
 349     info->nchannels = as->nchannels;
 350     info->bytes_per_frame = as->nchannels * mul;
 351     info->bytes_per_second = info->freq * info->bytes_per_frame;
 352     info->swap_endianness = (as->endianness != AUDIO_HOST_ENDIANNESS);
 353 }
 354
 355 void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
 356 {
 357     if (!len) {
 358         return;
 359     }
 360
 361     if (info->is_signed || info->is_float) {
 362         memset(buf, 0x00, len * info->bytes_per_frame);
 363     }
 364     else {
 365         switch (info->bits) {
 366         case 8:
 367             memset(buf, 0x80, len * info->bytes_per_frame);
 368             break;
 369
 370         case 16:
 371             {
 372                 int i;
 373                 uint16_t *p = buf;
 374                 short s = INT16_MAX;
 375
 376                 if (info->swap_endianness) {
 377                     s = bswap16 (s);
 378                 }
 379
 380                 for (i = 0; i < len * info->nchannels; i++) {
 381                     p[i] = s;
 382                 }
 383             }
 384             break;
 385
 386         case 32:
 387             {
 388                 int i;
 389                 uint32_t *p = buf;
 390                 int32_t s = INT32_MAX;
 391
 392                 if (info->swap_endianness) {
 393                     s = bswap32 (s);
 394                 }
 395
 396                 for (i = 0; i < len * info->nchannels; i++) {
 397                     p[i] = s;
 398                 }
 399             }
 400             break;
 401
 402         default:
 403             AUD_log (NULL, "audio_pcm_info_clear_buf: invalid bits %d\n",
 404                      info->bits);
 405             break;
 406         }
 407     }
 408 }
 409
 410 /*
 411  * Capture
 412  */
 413 static void noop_conv (struct st_sample *dst, const void *src, int samples)
 414 {
 415     (void) src;
 416     (void) dst;
 417     (void) samples;
 418 }
 419
 420 static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s,
 421                                                         struct audsettings *as)
 422 {
 423     CaptureVoiceOut *cap;
 424
 425     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
 426         if (audio_pcm_info_eq (&cap->hw.info, as)) {
 427             return cap;
 428         }
 429     }
 430     return NULL;
 431 }
 432
 433 static void audio_notify_capture (CaptureVoiceOut *cap, audcnotification_e cmd)
 434 {
 435     struct capture_callback *cb;
 436
 437 #ifdef DEBUG_CAPTURE
 438     dolog ("notification %d sent\n", cmd);
 439 #endif
 440     for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
 441         cb->ops.notify (cb->opaque, cmd);
 442     }
 443 }
 444
 445 static void audio_capture_maybe_changed (CaptureVoiceOut *cap, int enabled)
 446 {
 447     if (cap->hw.enabled != enabled) {
 448         audcnotification_e cmd;
 449         cap->hw.enabled = enabled;
 450         cmd = enabled ? AUD_CNOTIFY_ENABLE : AUD_CNOTIFY_DISABLE;
 451         audio_notify_capture (cap, cmd);
 452     }
 453 }
 454
 455 static void audio_recalc_and_notify_capture (CaptureVoiceOut *cap)
 456 {
 457     HWVoiceOut *hw = &cap->hw;
 458     SWVoiceOut *sw;
 459     int enabled = 0;
 460
 461     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
 462         if (sw->active) {
 463             enabled = 1;
 464             break;
 465         }
 466     }
 467     audio_capture_maybe_changed (cap, enabled);
 468 }
 469
 470 static void audio_detach_capture (HWVoiceOut *hw)
 471 {
 472     SWVoiceCap *sc = hw->cap_head.lh_first;
 473
 474     while (sc) {
 475         SWVoiceCap *sc1 = sc->entries.le_next;
 476         SWVoiceOut *sw = &sc->sw;
 477         CaptureVoiceOut *cap = sc->cap;
 478         int was_active = sw->active;
 479
 480         if (sw->rate) {
 481             st_rate_stop (sw->rate);
 482             sw->rate = NULL;
 483         }
 484
 485         QLIST_REMOVE (sw, entries);
 486         QLIST_REMOVE (sc, entries);
 487         g_free (sc);
 488         if (was_active) {
 489             /* We have removed soft voice from the capture:
 490                this might have changed the overall status of the capture
 491                since this might have been the only active voice */
 492             audio_recalc_and_notify_capture (cap);
 493         }
 494         sc = sc1;
 495     }
 496 }
 497
 498 static int audio_attach_capture (HWVoiceOut *hw)
 499 {
 500     AudioState *s = hw->s;
 501     CaptureVoiceOut *cap;
 502
 503     audio_detach_capture (hw);
 504     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
 505         SWVoiceCap *sc;
 506         SWVoiceOut *sw;
 507         HWVoiceOut *hw_cap = &cap->hw;
 508
 509         sc = g_malloc0(sizeof(*sc));
 510
 511         sc->cap = cap;
 512         sw = &sc->sw;
 513         sw->hw = hw_cap;
 514         sw->info = hw->info;
 515         sw->empty = 1;
 516         sw->active = hw->enabled;
 517         sw->conv = noop_conv;
 518         sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq;
 519         sw->vol = nominal_volume;
 520         sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq);
 521         if (!sw->rate) {
 522             dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw));
 523             g_free (sw);
 524             return -1;
 525         }
 526         QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries);
 527         QLIST_INSERT_HEAD (&hw->cap_head, sc, entries);
 528 #ifdef DEBUG_CAPTURE
 529         sw->name = g_strdup_printf ("for %p %d,%d,%d",
 530                                     hw, sw->info.freq, sw->info.bits,
 531                                     sw->info.nchannels);
 532         dolog ("Added %s active = %d\n", sw->name, sw->active);
 533 #endif
 534         if (sw->active) {
 535             audio_capture_maybe_changed (cap, 1);
 536         }
 537     }
 538     return 0;
 539 }
 540
 541 /*
 542  * Hard voice (capture)
 543  */
 544 static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
 545 {
 546     SWVoiceIn *sw;
 547     size_t m = hw->total_samples_captured;
 548
 549     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
 550         if (sw->active) {
 551             m = MIN (m, sw->total_hw_samples_acquired);
 552         }
 553     }
 554     return m;
 555 }
 556
 557 static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw)
 558 {
 559     size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
 560     if (audio_bug(__func__, live > hw->conv_buf->size)) {
 561         dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
 562         return 0;
 563     }
 564     return live;
 565 }
 566
 567 static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len)
 568 {
 569     size_t clipped = 0;
 570     size_t pos = hw->mix_buf->pos;
 571
 572     while (len) {
 573         st_sample *src = hw->mix_buf->samples + pos;
 574         uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);
 575         size_t samples_till_end_of_buf = hw->mix_buf->size - pos;
 576         size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
 577
 578         hw->clip(dst, src, samples_to_clip);
 579
 580         pos = (pos + samples_to_clip) % hw->mix_buf->size;
 581         len -= samples_to_clip;
 582         clipped += samples_to_clip;
 583     }
 584 }
 585
 586 /*
 587  * Soft voice (capture)
 588  */
 589 static size_t audio_pcm_sw_get_rpos_in(SWVoiceIn *sw)
 590 {
 591     HWVoiceIn *hw = sw->hw;
 592     ssize_t live = hw->total_samples_captured - sw->total_hw_samples_acquired;
 593     ssize_t rpos;
 594
 595     if (audio_bug(__func__, live < 0 || live > hw->conv_buf->size)) {
 596         dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
 597         return 0;
 598     }
 599
 600     rpos = hw->conv_buf->pos - live;
 601     if (rpos >= 0) {
 602         return rpos;
 603     }
 604     else {
 605         return hw->conv_buf->size + rpos;
 606     }
 607 }
 608
 609 static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size)
 610 {
 611     HWVoiceIn *hw = sw->hw;
 612     size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
 613     struct st_sample *src, *dst = sw->buf;
 614
 615     rpos = audio_pcm_sw_get_rpos_in(sw) % hw->conv_buf->size;
 616
 617     live = hw->total_samples_captured - sw->total_hw_samples_acquired;
 618     if (audio_bug(__func__, live > hw->conv_buf->size)) {
 619         dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size);
 620         return 0;
 621     }
 622
 623     samples = size / sw->info.bytes_per_frame;
 624     if (!live) {
 625         return 0;
 626     }
 627
 628     swlim = (live * sw->ratio) >> 32;
 629     swlim = MIN (swlim, samples);
 630
 631     while (swlim) {
 632         src = hw->conv_buf->samples + rpos;
 633         if (hw->conv_buf->pos > rpos) {
 634             isamp = hw->conv_buf->pos - rpos;
 635         } else {
 636             isamp = hw->conv_buf->size - rpos;
 637         }
 638
 639         if (!isamp) {
 640             break;
 641         }
 642         osamp = swlim;
 643
 644         st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
 645         swlim -= osamp;
 646         rpos = (rpos + isamp) % hw->conv_buf->size;
 647         dst += osamp;
 648         ret += osamp;
 649         total += isamp;
 650     }
 651
 652     if (hw->pcm_ops && !hw->pcm_ops->volume_in) {
 653         mixeng_volume (sw->buf, ret, &sw->vol);
 654     }
 655
 656     sw->clip (buf, sw->buf, ret);
 657     sw->total_hw_samples_acquired += total;
 658     return ret * sw->info.bytes_per_frame;
 659 }
 660
 661 /*
 662  * Hard voice (playback)
 663  */
 664 static size_t audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
 665 {
 666     SWVoiceOut *sw;
 667     size_t m = SIZE_MAX;
 668     int nb_live = 0;
 669
 670     for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
 671         if (sw->active || !sw->empty) {
 672             m = MIN (m, sw->total_hw_samples_mixed);
 673             nb_live += 1;
 674         }
 675     }
 676
 677     *nb_livep = nb_live;
 678     return m;
 679 }
 680
 681 static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live)
 682 {
 683     size_t smin;
 684     int nb_live1;
 685
 686     smin = audio_pcm_hw_find_min_out (hw, &nb_live1);
 687     if (nb_live) {
 688         *nb_live = nb_live1;
 689     }
 690
 691     if (nb_live1) {
 692         size_t live = smin;
 693
 694         if (audio_bug(__func__, live > hw->mix_buf->size)) {
 695             dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
 696             return 0;
 697         }
 698         return live;
 699     }
 700     return 0;
 701 }
 702
 703 /*
 704  * Soft voice (playback)
 705  */
 706 static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size)
 707 {
 708     size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
 709     size_t ret = 0, pos = 0, total = 0;
 710
 711     if (!sw) {
 712         return size;
 713     }
 714
 715     hwsamples = sw->hw->mix_buf->size;
 716
 717     live = sw->total_hw_samples_mixed;
 718     if (audio_bug(__func__, live > hwsamples)) {
 719         dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples);
 720         return 0;
 721     }
 722
 723     if (live == hwsamples) {
 724 #ifdef DEBUG_OUT
 725         dolog ("%s is full %d\n", sw->name, live);
 726 #endif
 727         return 0;
 728     }
 729
 730     wpos = (sw->hw->mix_buf->pos + live) % hwsamples;
 731     samples = size / sw->info.bytes_per_frame;
 732
 733     dead = hwsamples - live;
 734     swlim = ((int64_t) dead << 32) / sw->ratio;
 735     swlim = MIN (swlim, samples);
 736     if (swlim) {
 737         sw->conv (sw->buf, buf, swlim);
 738
 739         if (sw->hw->pcm_ops && !sw->hw->pcm_ops->volume_out) {
 740             mixeng_volume (sw->buf, swlim, &sw->vol);
 741         }
 742     }
 743
 744     while (swlim) {
 745         dead = hwsamples - live;
 746         left = hwsamples - wpos;
 747         blck = MIN (dead, left);
 748         if (!blck) {
 749             break;
 750         }
 751         isamp = swlim;
 752         osamp = blck;
 753         st_rate_flow_mix (
 754             sw->rate,
 755             sw->buf + pos,
 756             sw->hw->mix_buf->samples + wpos,
 757             &isamp,
 758             &osamp
 759             );
 760         ret += isamp;
 761         swlim -= isamp;
 762         pos += isamp;
 763         live += osamp;
 764         wpos = (wpos + osamp) % hwsamples;
 765         total += osamp;
 766     }
 767
 768     sw->total_hw_samples_mixed += total;
 769     sw->empty = sw->total_hw_samples_mixed == 0;
 770
 771 #ifdef DEBUG_OUT
 772     dolog (
 773         "%s: write size %zu ret %zu total sw %zu\n",
 774         SW_NAME (sw),
 775         size / sw->info.bytes_per_frame,
 776         ret,
 777         sw->total_hw_samples_mixed
 778         );
 779 #endif
 780
 781     return ret * sw->info.bytes_per_frame;
 782 }
 783
 784 #ifdef DEBUG_AUDIO
 785 static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
 786 {
 787     dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",
 788           cap, info->bits, info->is_signed, info->is_float, info->freq,
 789           info->nchannels);
 790 }
 791 #endif
 792
 793 #define DAC
 794 #include "audio_template.h"
 795 #undef DAC
 796 #include "audio_template.h"
 797
 798 /*
 799  * Timer
 800  */
 801 static int audio_is_timer_needed(AudioState *s)
 802 {
 803     HWVoiceIn *hwi = NULL;
 804     HWVoiceOut *hwo = NULL;
 805
 806     while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
 807         if (!hwo->poll_mode) return 1;
 808     }
 809     while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
 810         if (!hwi->poll_mode) return 1;
 811     }
 812     return 0;
 813 }
 814
 815 static void audio_reset_timer (AudioState *s)
 816 {
 817     if (audio_is_timer_needed(s)) {
 818         timer_mod_anticipate_ns(s->ts,
 819             qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL) + s->period_ticks);
 820         if (!s->timer_running) {
 821             s->timer_running = true;
 822             s->timer_last = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 823             trace_audio_timer_start(s->period_ticks / SCALE_MS);
 824         }
 825     } else {
 826         timer_del(s->ts);
 827         if (s->timer_running) {
 828             s->timer_running = false;
 829             trace_audio_timer_stop();
 830         }
 831     }
 832 }
 833
 834 static void audio_timer (void *opaque)
 835 {
 836     int64_t now, diff;
 837     AudioState *s = opaque;
 838
 839     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 840     diff = now - s->timer_last;
 841     if (diff > s->period_ticks * 3 / 2) {
 842         trace_audio_timer_delayed(diff / SCALE_MS);
 843     }
 844     s->timer_last = now;
 845
 846     audio_run(s, "timer");
 847     audio_reset_timer(s);
 848 }
 849
 850 /*
 851  * Public API
 852  */
 853 size_t AUD_write(SWVoiceOut *sw, void *buf, size_t size)
 854 {
 855     HWVoiceOut *hw;
 856
 857     if (!sw) {
 858         /* XXX: Consider options */
 859         return size;
 860     }
 861     hw = sw->hw;
 862
 863     if (!hw->enabled) {
 864         dolog ("Writing to disabled voice %s\n", SW_NAME (sw));
 865         return 0;
 866     }
 867
 868     if (audio_get_pdo_out(hw->s->dev)->mixing_engine) {
 869         return audio_pcm_sw_write(sw, buf, size);
 870     } else {
 871         return hw->pcm_ops->write(hw, buf, size);
 872     }
 873 }
 874
 875 size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size)
 876 {
 877     HWVoiceIn *hw;
 878
 879     if (!sw) {
 880         /* XXX: Consider options */
 881         return size;
 882     }
 883     hw = sw->hw;
 884
 885     if (!hw->enabled) {
 886         dolog ("Reading from disabled voice %s\n", SW_NAME (sw));
 887         return 0;
 888     }
 889
 890     if (audio_get_pdo_in(hw->s->dev)->mixing_engine) {
 891         return audio_pcm_sw_read(sw, buf, size);
 892     } else {
 893         return hw->pcm_ops->read(hw, buf, size);
 894     }
 895 }
 896
 897 int AUD_get_buffer_size_out(SWVoiceOut *sw)
 898 {
 899     return sw->hw->samples * sw->hw->info.bytes_per_frame;
 900 }
 901
 902 void AUD_set_active_out (SWVoiceOut *sw, int on)
 903 {
 904     HWVoiceOut *hw;
 905
 906     if (!sw) {
 907         return;
 908     }
 909
 910     hw = sw->hw;
 911     if (sw->active != on) {
 912         AudioState *s = sw->s;
 913         SWVoiceOut *temp_sw;
 914         SWVoiceCap *sc;
 915
 916         if (on) {
 917             hw->pending_disable = 0;
 918             if (!hw->enabled) {
 919                 hw->enabled = 1;
 920                 if (s->vm_running) {
 921                     if (hw->pcm_ops->enable_out) {
 922                         hw->pcm_ops->enable_out(hw, true);
 923                     }
 924                     audio_reset_timer (s);
 925                 }
 926             }
 927         }
 928         else {
 929             if (hw->enabled) {
 930                 int nb_active = 0;
 931
 932                 for (temp_sw = hw->sw_head.lh_first; temp_sw;
 933                      temp_sw = temp_sw->entries.le_next) {
 934                     nb_active += temp_sw->active != 0;
 935                 }
 936
 937                 hw->pending_disable = nb_active == 1;
 938             }
 939         }
 940
 941         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
 942             sc->sw.active = hw->enabled;
 943             if (hw->enabled) {
 944                 audio_capture_maybe_changed (sc->cap, 1);
 945             }
 946         }
 947         sw->active = on;
 948     }
 949 }
 950
 951 void AUD_set_active_in (SWVoiceIn *sw, int on)
 952 {
 953     HWVoiceIn *hw;
 954
 955     if (!sw) {
 956         return;
 957     }
 958
 959     hw = sw->hw;
 960     if (sw->active != on) {
 961         AudioState *s = sw->s;
 962         SWVoiceIn *temp_sw;
 963
 964         if (on) {
 965             if (!hw->enabled) {
 966                 hw->enabled = 1;
 967                 if (s->vm_running) {
 968                     if (hw->pcm_ops->enable_in) {
 969                         hw->pcm_ops->enable_in(hw, true);
 970                     }
 971                     audio_reset_timer (s);
 972                 }
 973             }
 974             sw->total_hw_samples_acquired = hw->total_samples_captured;
 975         }
 976         else {
 977             if (hw->enabled) {
 978                 int nb_active = 0;
 979
 980                 for (temp_sw = hw->sw_head.lh_first; temp_sw;
 981                      temp_sw = temp_sw->entries.le_next) {
 982                     nb_active += temp_sw->active != 0;
 983                 }
 984
 985                 if (nb_active == 1) {
 986                     hw->enabled = 0;
 987                     if (hw->pcm_ops->enable_in) {
 988                         hw->pcm_ops->enable_in(hw, false);
 989                     }
 990                 }
 991             }
 992         }
 993         sw->active = on;
 994     }
 995 }
 996
 997 static size_t audio_get_avail (SWVoiceIn *sw)
 998 {
 999     size_t live;
1000
1001     if (!sw) {
1002         return 0;
1003     }
1004
1005     live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
1006     if (audio_bug(__func__, live > sw->hw->conv_buf->size)) {
1007         dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live,
1008               sw->hw->conv_buf->size);
1009         return 0;
1010     }
1011
1012     ldebug (
1013         "%s: get_avail live %d ret %" PRId64 "\n",
1014         SW_NAME (sw),
1015         live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame
1016         );
1017
1018     return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;
1019 }
1020
1021 static size_t audio_get_free(SWVoiceOut *sw)
1022 {
1023     size_t live, dead;
1024
1025     if (!sw) {
1026         return 0;
1027     }
1028
1029     live = sw->total_hw_samples_mixed;
1030
1031     if (audio_bug(__func__, live > sw->hw->mix_buf->size)) {
1032         dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live,
1033               sw->hw->mix_buf->size);
1034         return 0;
1035     }
1036
1037     dead = sw->hw->mix_buf->size - live;
1038
1039 #ifdef DEBUG_OUT
1040     dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",
1041            SW_NAME (sw),
1042            live, dead, (((int64_t) dead << 32) / sw->ratio) *
1043            sw->info.bytes_per_frame);
1044 #endif
1045
1046     return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;
1047 }
1048
1049 static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
1050                                         size_t samples)
1051 {
1052     size_t n;
1053
1054     if (hw->enabled) {
1055         SWVoiceCap *sc;
1056
1057         for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1058             SWVoiceOut *sw = &sc->sw;
1059             int rpos2 = rpos;
1060
1061             n = samples;
1062             while (n) {
1063                 size_t till_end_of_hw = hw->mix_buf->size - rpos2;
1064                 size_t to_write = MIN(till_end_of_hw, n);
1065                 size_t bytes = to_write * hw->info.bytes_per_frame;
1066                 size_t written;
1067
1068                 sw->buf = hw->mix_buf->samples + rpos2;
1069                 written = audio_pcm_sw_write (sw, NULL, bytes);
1070                 if (written - bytes) {
1071                     dolog("Could not mix %zu bytes into a capture "
1072                           "buffer, mixed %zu\n",
1073                           bytes, written);
1074                     break;
1075                 }
1076                 n -= to_write;
1077                 rpos2 = (rpos2 + to_write) % hw->mix_buf->size;
1078             }
1079         }
1080     }
1081
1082     n = MIN(samples, hw->mix_buf->size - rpos);
1083     mixeng_clear(hw->mix_buf->samples + rpos, n);
1084     mixeng_clear(hw->mix_buf->samples, samples - n);
1085 }
1086
1087 static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live)
1088 {
1089     size_t clipped = 0;
1090
1091     while (live) {
1092         size_t size, decr, proc;
1093         void *buf = hw->pcm_ops->get_buffer_out(hw, &size);
1094         if (!buf || size == 0) {
1095             break;
1096         }
1097
1098         decr = MIN(size / hw->info.bytes_per_frame, live);
1099         audio_pcm_hw_clip_out(hw, buf, decr);
1100         proc = hw->pcm_ops->put_buffer_out(hw, buf,
1101                                            decr * hw->info.bytes_per_frame) /
1102             hw->info.bytes_per_frame;
1103
1104         live -= proc;
1105         clipped += proc;
1106         hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size;
1107
1108         if (proc == 0 || proc < decr) {
1109             break;
1110         }
1111     }
1112
1113     if (hw->pcm_ops->run_buffer_out) {
1114         hw->pcm_ops->run_buffer_out(hw);
1115     }
1116
1117     return clipped;
1118 }
1119
1120 static void audio_run_out (AudioState *s)
1121 {
1122     HWVoiceOut *hw = NULL;
1123     SWVoiceOut *sw;
1124
1125     if (!audio_get_pdo_out(s->dev)->mixing_engine) {
1126         while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
1127             /* there is exactly 1 sw for each hw with no mixeng */
1128             sw = hw->sw_head.lh_first;
1129
1130             if (hw->pending_disable) {
1131                 hw->enabled = 0;
1132                 hw->pending_disable = 0;
1133                 if (hw->pcm_ops->enable_out) {
1134                     hw->pcm_ops->enable_out(hw, false);
1135                 }
1136             }
1137
1138             if (sw->active) {
1139                 sw->callback.fn(sw->callback.opaque, INT_MAX);
1140             }
1141         }
1142         return;
1143     }
1144
1145     while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) {
1146         size_t played, live, prev_rpos, free;
1147         int nb_live, cleanup_required;
1148
1149         live = audio_pcm_hw_get_live_out (hw, &nb_live);
1150         if (!nb_live) {
1151             live = 0;
1152         }
1153
1154         if (audio_bug(__func__, live > hw->mix_buf->size)) {
1155             dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size);
1156             continue;
1157         }
1158
1159         if (hw->pending_disable && !nb_live) {
1160             SWVoiceCap *sc;
1161 #ifdef DEBUG_OUT
1162             dolog ("Disabling voice\n");
1163 #endif
1164             hw->enabled = 0;
1165             hw->pending_disable = 0;
1166             if (hw->pcm_ops->enable_out) {
1167                 hw->pcm_ops->enable_out(hw, false);
1168             }
1169             for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1170                 sc->sw.active = 0;
1171                 audio_recalc_and_notify_capture (sc->cap);
1172             }
1173             continue;
1174         }
1175
1176         if (!live) {
1177             for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1178                 if (sw->active) {
1179                     free = audio_get_free (sw);
1180                     if (free > 0) {
1181                         sw->callback.fn (sw->callback.opaque, free);
1182                     }
1183                 }
1184             }
1185             continue;
1186         }
1187
1188         prev_rpos = hw->mix_buf->pos;
1189         played = audio_pcm_hw_run_out(hw, live);
1190         replay_audio_out(&played);
1191         if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) {
1192             dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n",
1193                   hw->mix_buf->pos, hw->mix_buf->size, played);
1194             hw->mix_buf->pos = 0;
1195         }
1196
1197 #ifdef DEBUG_OUT
1198         dolog("played=%zu\n", played);
1199 #endif
1200
1201         if (played) {
1202             hw->ts_helper += played;
1203             audio_capture_mix_and_clear (hw, prev_rpos, played);
1204         }
1205
1206         cleanup_required = 0;
1207         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1208             if (!sw->active && sw->empty) {
1209                 continue;
1210             }
1211
1212             if (audio_bug(__func__, played > sw->total_hw_samples_mixed)) {
1213                 dolog("played=%zu sw->total_hw_samples_mixed=%zu\n",
1214                       played, sw->total_hw_samples_mixed);
1215                 played = sw->total_hw_samples_mixed;
1216             }
1217
1218             sw->total_hw_samples_mixed -= played;
1219
1220             if (!sw->total_hw_samples_mixed) {
1221                 sw->empty = 1;
1222                 cleanup_required |= !sw->active && !sw->callback.fn;
1223             }
1224
1225             if (sw->active) {
1226                 free = audio_get_free (sw);
1227                 if (free > 0) {
1228                     sw->callback.fn (sw->callback.opaque, free);
1229                 }
1230             }
1231         }
1232
1233         if (cleanup_required) {
1234             SWVoiceOut *sw1;
1235
1236             sw = hw->sw_head.lh_first;
1237             while (sw) {
1238                 sw1 = sw->entries.le_next;
1239                 if (!sw->active && !sw->callback.fn) {
1240                     audio_close_out (sw);
1241                 }
1242                 sw = sw1;
1243             }
1244         }
1245     }
1246 }
1247
1248 static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples)
1249 {
1250     size_t conv = 0;
1251     STSampleBuffer *conv_buf = hw->conv_buf;
1252
1253     while (samples) {
1254         size_t proc;
1255         size_t size = samples * hw->info.bytes_per_frame;
1256         void *buf = hw->pcm_ops->get_buffer_in(hw, &size);
1257
1258         assert(size % hw->info.bytes_per_frame == 0);
1259         if (size == 0) {
1260             hw->pcm_ops->put_buffer_in(hw, buf, size);
1261             break;
1262         }
1263
1264         proc = MIN(size / hw->info.bytes_per_frame,
1265                    conv_buf->size - conv_buf->pos);
1266
1267         hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
1268         conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size;
1269
1270         samples -= proc;
1271         conv += proc;
1272         hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);
1273     }
1274
1275     return conv;
1276 }
1277
1278 static void audio_run_in (AudioState *s)
1279 {
1280     HWVoiceIn *hw = NULL;
1281
1282     if (!audio_get_pdo_in(s->dev)->mixing_engine) {
1283         while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
1284             /* there is exactly 1 sw for each hw with no mixeng */
1285             SWVoiceIn *sw = hw->sw_head.lh_first;
1286             if (sw->active) {
1287                 sw->callback.fn(sw->callback.opaque, INT_MAX);
1288             }
1289         }
1290         return;
1291     }
1292
1293     while ((hw = audio_pcm_hw_find_any_enabled_in(s, hw))) {
1294         SWVoiceIn *sw;
1295         size_t captured = 0, min;
1296
1297         if (replay_mode != REPLAY_MODE_PLAY) {
1298             captured = audio_pcm_hw_run_in(
1299                 hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw));
1300         }
1301         replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos,
1302                         hw->conv_buf->size);
1303
1304         min = audio_pcm_hw_find_min_in (hw);
1305         hw->total_samples_captured += captured - min;
1306         hw->ts_helper += captured;
1307
1308         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1309             sw->total_hw_samples_acquired -= min;
1310
1311             if (sw->active) {
1312                 size_t avail;
1313
1314                 avail = audio_get_avail (sw);
1315                 if (avail > 0) {
1316                     sw->callback.fn (sw->callback.opaque, avail);
1317                 }
1318             }
1319         }
1320     }
1321 }
1322
1323 static void audio_run_capture (AudioState *s)
1324 {
1325     CaptureVoiceOut *cap;
1326
1327     for (cap = s->cap_head.lh_first; cap; cap = cap->entries.le_next) {
1328         size_t live, rpos, captured;
1329         HWVoiceOut *hw = &cap->hw;
1330         SWVoiceOut *sw;
1331
1332         captured = live = audio_pcm_hw_get_live_out (hw, NULL);
1333         rpos = hw->mix_buf->pos;
1334         while (live) {
1335             size_t left = hw->mix_buf->size - rpos;
1336             size_t to_capture = MIN(live, left);
1337             struct st_sample *src;
1338             struct capture_callback *cb;
1339
1340             src = hw->mix_buf->samples + rpos;
1341             hw->clip (cap->buf, src, to_capture);
1342             mixeng_clear (src, to_capture);
1343
1344             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1345                 cb->ops.capture (cb->opaque, cap->buf,
1346                                  to_capture * hw->info.bytes_per_frame);
1347             }
1348             rpos = (rpos + to_capture) % hw->mix_buf->size;
1349             live -= to_capture;
1350         }
1351         hw->mix_buf->pos = rpos;
1352
1353         for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
1354             if (!sw->active && sw->empty) {
1355                 continue;
1356             }
1357
1358             if (audio_bug(__func__, captured > sw->total_hw_samples_mixed)) {
1359                 dolog("captured=%zu sw->total_hw_samples_mixed=%zu\n",
1360                       captured, sw->total_hw_samples_mixed);
1361                 captured = sw->total_hw_samples_mixed;
1362             }
1363
1364             sw->total_hw_samples_mixed -= captured;
1365             sw->empty = sw->total_hw_samples_mixed == 0;
1366         }
1367     }
1368 }
1369
1370 void audio_run(AudioState *s, const char *msg)
1371 {
1372     audio_run_out(s);
1373     audio_run_in(s);
1374     audio_run_capture(s);
1375
1376 #ifdef DEBUG_POLL
1377     {
1378         static double prevtime;
1379         double currtime;
1380         struct timeval tv;
1381
1382         if (gettimeofday (&tv, NULL)) {
1383             perror ("audio_run: gettimeofday");
1384             return;
1385         }
1386
1387         currtime = tv.tv_sec + tv.tv_usec * 1e-6;
1388         dolog ("Elapsed since last %s: %f\n", msg, currtime - prevtime);
1389         prevtime = currtime;
1390     }
1391 #endif
1392 }
1393
1394 void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size)
1395 {
1396     ssize_t start;
1397
1398     if (unlikely(!hw->buf_emul)) {
1399         size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame;
1400         hw->buf_emul = g_malloc(calc_size);
1401         hw->size_emul = calc_size;
1402         hw->pos_emul = hw->pending_emul = 0;
1403     }
1404
1405     while (hw->pending_emul < hw->size_emul) {
1406         size_t read_len = MIN(hw->size_emul - hw->pos_emul,
1407                               hw->size_emul - hw->pending_emul);
1408         size_t read = hw->pcm_ops->read(hw, hw->buf_emul + hw->pos_emul,
1409                                         read_len);
1410         hw->pending_emul += read;
1411         hw->pos_emul = (hw->pos_emul + read) % hw->size_emul;
1412         if (read < read_len) {
1413             break;
1414         }
1415     }
1416
1417     start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
1418     if (start < 0) {
1419         start += hw->size_emul;
1420     }
1421     assert(start >= 0 && start < hw->size_emul);
1422
1423     *size = MIN(*size, hw->pending_emul);
1424     *size = MIN(*size, hw->size_emul - start);
1425     return hw->buf_emul + start;
1426 }
1427
1428 void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size)
1429 {
1430     assert(size <= hw->pending_emul);
1431     hw->pending_emul -= size;
1432 }
1433
1434 void audio_generic_run_buffer_out(HWVoiceOut *hw)
1435 {
1436     while (hw->pending_emul) {
1437         size_t write_len, written;
1438         ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;
1439
1440         if (start < 0) {
1441             start += hw->size_emul;
1442         }
1443         assert(start >= 0 && start < hw->size_emul);
1444
1445         write_len = MIN(hw->pending_emul, hw->size_emul - start);
1446
1447         written = hw->pcm_ops->write(hw, hw->buf_emul + start, write_len);
1448         hw->pending_emul -= written;
1449
1450         if (written < write_len) {
1451             break;
1452         }
1453     }
1454 }
1455
1456 void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)
1457 {
1458     if (unlikely(!hw->buf_emul)) {
1459         size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame;
1460
1461         hw->buf_emul = g_malloc(calc_size);
1462         hw->size_emul = calc_size;
1463         hw->pos_emul = hw->pending_emul = 0;
1464     }
1465
1466     *size = MIN(hw->size_emul - hw->pending_emul,
1467                 hw->size_emul - hw->pos_emul);
1468     return hw->buf_emul + hw->pos_emul;
1469 }
1470
1471 size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size)
1472 {
1473     assert(buf == hw->buf_emul + hw->pos_emul &&
1474            size + hw->pending_emul <= hw->size_emul);
1475
1476     hw->pending_emul += size;
1477     hw->pos_emul = (hw->pos_emul + size) % hw->size_emul;
1478
1479     return size;
1480 }
1481
1482 size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size)
1483 {
1484     size_t dst_size, copy_size;
1485     void *dst = hw->pcm_ops->get_buffer_out(hw, &dst_size);
1486     copy_size = MIN(size, dst_size);
1487
1488     memcpy(dst, buf, copy_size);
1489     return hw->pcm_ops->put_buffer_out(hw, dst, copy_size);
1490 }
1491
1492 size_t audio_generic_read(HWVoiceIn *hw, void *buf, size_t size)
1493 {
1494     void *src = hw->pcm_ops->get_buffer_in(hw, &size);
1495
1496     memcpy(buf, src, size);
1497     hw->pcm_ops->put_buffer_in(hw, src, size);
1498
1499     return size;
1500 }
1501
1502 static int audio_driver_init(AudioState *s, struct audio_driver *drv,
1503                              bool msg, Audiodev *dev)
1504 {
1505     s->drv_opaque = drv->init(dev);
1506
1507     if (s->drv_opaque) {
1508         if (!drv->pcm_ops->get_buffer_in) {
1509             drv->pcm_ops->get_buffer_in = audio_generic_get_buffer_in;
1510             drv->pcm_ops->put_buffer_in = audio_generic_put_buffer_in;
1511         }
1512         if (!drv->pcm_ops->get_buffer_out) {
1513             drv->pcm_ops->get_buffer_out = audio_generic_get_buffer_out;
1514             drv->pcm_ops->put_buffer_out = audio_generic_put_buffer_out;
1515         }
1516
1517         audio_init_nb_voices_out(s, drv);
1518         audio_init_nb_voices_in(s, drv);
1519         s->drv = drv;
1520         return 0;
1521     }
1522     else {
1523         if (msg) {
1524             dolog("Could not init `%s' audio driver\n", drv->name);
1525         }
1526         return -1;
1527     }
1528 }
1529
1530 static void audio_vm_change_state_handler (void *opaque, int running,
1531                                            RunState state)
1532 {
1533     AudioState *s = opaque;
1534     HWVoiceOut *hwo = NULL;
1535     HWVoiceIn *hwi = NULL;
1536
1537     s->vm_running = running;
1538     while ((hwo = audio_pcm_hw_find_any_enabled_out(s, hwo))) {
1539         if (hwo->pcm_ops->enable_out) {
1540             hwo->pcm_ops->enable_out(hwo, running);
1541         }
1542     }
1543
1544     while ((hwi = audio_pcm_hw_find_any_enabled_in(s, hwi))) {
1545         if (hwi->pcm_ops->enable_in) {
1546             hwi->pcm_ops->enable_in(hwi, running);
1547         }
1548     }
1549     audio_reset_timer (s);
1550 }
1551
1552 static bool is_cleaning_up;
1553
1554 bool audio_is_cleaning_up(void)
1555 {
1556     return is_cleaning_up;
1557 }
1558
1559 static void free_audio_state(AudioState *s)
1560 {
1561     HWVoiceOut *hwo, *hwon;
1562     HWVoiceIn *hwi, *hwin;
1563
1564     QLIST_FOREACH_SAFE(hwo, &s->hw_head_out, entries, hwon) {
1565         SWVoiceCap *sc;
1566
1567         if (hwo->enabled && hwo->pcm_ops->enable_out) {
1568             hwo->pcm_ops->enable_out(hwo, false);
1569         }
1570         hwo->pcm_ops->fini_out (hwo);
1571
1572         for (sc = hwo->cap_head.lh_first; sc; sc = sc->entries.le_next) {
1573             CaptureVoiceOut *cap = sc->cap;
1574             struct capture_callback *cb;
1575
1576             for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1577                 cb->ops.destroy (cb->opaque);
1578             }
1579         }
1580         QLIST_REMOVE(hwo, entries);
1581     }
1582
1583     QLIST_FOREACH_SAFE(hwi, &s->hw_head_in, entries, hwin) {
1584         if (hwi->enabled && hwi->pcm_ops->enable_in) {
1585             hwi->pcm_ops->enable_in(hwi, false);
1586         }
1587         hwi->pcm_ops->fini_in (hwi);
1588         QLIST_REMOVE(hwi, entries);
1589     }
1590
1591     if (s->drv) {
1592         s->drv->fini (s->drv_opaque);
1593         s->drv = NULL;
1594     }
1595
1596     if (s->dev) {
1597         qapi_free_Audiodev(s->dev);
1598         s->dev = NULL;
1599     }
1600
1601     if (s->ts) {
1602         timer_free(s->ts);
1603         s->ts = NULL;
1604     }
1605
1606     g_free(s);
1607 }
1608
1609 void audio_cleanup(void)
1610 {
1611     is_cleaning_up = true;
1612     while (!QTAILQ_EMPTY(&audio_states)) {
1613         AudioState *s = QTAILQ_FIRST(&audio_states);
1614         QTAILQ_REMOVE(&audio_states, s, list);
1615         free_audio_state(s);
1616     }
1617 }
1618
1619 static const VMStateDescription vmstate_audio = {
1620     .name = "audio",
1621     .version_id = 1,
1622     .minimum_version_id = 1,
1623     .fields = (VMStateField[]) {
1624         VMSTATE_END_OF_LIST()
1625     }
1626 };
1627
1628 static void audio_validate_opts(Audiodev *dev, Error **errp);
1629
1630 static AudiodevListEntry *audiodev_find(
1631     AudiodevListHead *head, const char *drvname)
1632 {
1633     AudiodevListEntry *e;
1634     QSIMPLEQ_FOREACH(e, head, next) {
1635         if (strcmp(AudiodevDriver_str(e->dev->driver), drvname) == 0) {
1636             return e;
1637         }
1638     }
1639
1640     return NULL;
1641 }
1642
1643 /*
1644  * if we have dev, this function was called because of an -audiodev argument =>
1645  *   initialize a new state with it
1646  * if dev == NULL => legacy implicit initialization, return the already created
1647  *   state or create a new one
1648  */
1649 static AudioState *audio_init(Audiodev *dev, const char *name)
1650 {
1651     static bool atexit_registered;
1652     size_t i;
1653     int done = 0;
1654     const char *drvname = NULL;
1655     VMChangeStateEntry *e;
1656     AudioState *s;
1657     struct audio_driver *driver;
1658     /* silence gcc warning about uninitialized variable */
1659     AudiodevListHead head = QSIMPLEQ_HEAD_INITIALIZER(head);
1660
1661     if (dev) {
1662         /* -audiodev option */
1663         legacy_config = false;
1664         drvname = AudiodevDriver_str(dev->driver);
1665     } else if (!QTAILQ_EMPTY(&audio_states)) {
1666         if (!legacy_config) {
1667             dolog("Device %s: audiodev default parameter is deprecated, please "
1668                   "specify audiodev=%s\n", name,
1669                   QTAILQ_FIRST(&audio_states)->dev->id);
1670         }
1671         return QTAILQ_FIRST(&audio_states);
1672     } else {
1673         /* legacy implicit initialization */
1674         head = audio_handle_legacy_opts();
1675         /*
1676          * In case of legacy initialization, all Audiodevs in the list will have
1677          * the same configuration (except the driver), so it does't matter which
1678          * one we chose.  We need an Audiodev to set up AudioState before we can
1679          * init a driver.  Also note that dev at this point is still in the
1680          * list.
1681          */
1682         dev = QSIMPLEQ_FIRST(&head)->dev;
1683         audio_validate_opts(dev, &error_abort);
1684     }
1685
1686     s = g_malloc0(sizeof(AudioState));
1687     s->dev = dev;
1688
1689     QLIST_INIT (&s->hw_head_out);
1690     QLIST_INIT (&s->hw_head_in);
1691     QLIST_INIT (&s->cap_head);
1692     if (!atexit_registered) {
1693         atexit(audio_cleanup);
1694         atexit_registered = true;
1695     }
1696     QTAILQ_INSERT_TAIL(&audio_states, s, list);
1697
1698     s->ts = timer_new_ns(QEMU_CLOCK_VIRTUAL, audio_timer, s);
1699
1700     s->nb_hw_voices_out = audio_get_pdo_out(dev)->voices;
1701     s->nb_hw_voices_in = audio_get_pdo_in(dev)->voices;
1702
1703     if (s->nb_hw_voices_out <= 0) {
1704         dolog ("Bogus number of playback voices %d, setting to 1\n",
1705                s->nb_hw_voices_out);
1706         s->nb_hw_voices_out = 1;
1707     }
1708
1709     if (s->nb_hw_voices_in <= 0) {
1710         dolog ("Bogus number of capture voices %d, setting to 0\n",
1711                s->nb_hw_voices_in);
1712         s->nb_hw_voices_in = 0;
1713     }
1714
1715     if (drvname) {
1716         driver = audio_driver_lookup(drvname);
1717         if (driver) {
1718             done = !audio_driver_init(s, driver, true, dev);
1719         } else {
1720             dolog ("Unknown audio driver `%s'\n", drvname);
1721         }
1722     } else {
1723         for (i = 0; audio_prio_list[i]; i++) {
1724             AudiodevListEntry *e = audiodev_find(&head, audio_prio_list[i]);
1725             driver = audio_driver_lookup(audio_prio_list[i]);
1726
1727             if (e && driver) {
1728                 s->dev = dev = e->dev;
1729                 audio_validate_opts(dev, &error_abort);
1730                 done = !audio_driver_init(s, driver, false, dev);
1731                 if (done) {
1732                     e->dev = NULL;
1733                     break;
1734                 }
1735             }
1736         }
1737     }
1738     audio_free_audiodev_list(&head);
1739
1740     if (!done) {
1741         driver = audio_driver_lookup("none");
1742         done = !audio_driver_init(s, driver, false, dev);
1743         assert(done);
1744         dolog("warning: Using timer based audio emulation\n");
1745     }
1746
1747     if (dev->timer_period <= 0) {
1748         s->period_ticks = 1;
1749     } else {
1750         s->period_ticks = dev->timer_period * (int64_t)SCALE_US;
1751     }
1752
1753     e = qemu_add_vm_change_state_handler (audio_vm_change_state_handler, s);
1754     if (!e) {
1755         dolog ("warning: Could not register change state handler\n"
1756                "(Audio can continue looping even after stopping the VM)\n");
1757     }
1758
1759     QLIST_INIT (&s->card_head);
1760     vmstate_register (NULL, 0, &vmstate_audio, s);
1761     return s;
1762 }
1763
1764 void audio_free_audiodev_list(AudiodevListHead *head)
1765 {
1766     AudiodevListEntry *e;
1767     while ((e = QSIMPLEQ_FIRST(head))) {
1768         QSIMPLEQ_REMOVE_HEAD(head, next);
1769         qapi_free_Audiodev(e->dev);
1770         g_free(e);
1771     }
1772 }
1773
1774 void AUD_register_card (const char *name, QEMUSoundCard *card)
1775 {
1776     if (!card->state) {
1777         card->state = audio_init(NULL, name);
1778     }
1779
1780     card->name = g_strdup (name);
1781     memset (&card->entries, 0, sizeof (card->entries));
1782     QLIST_INSERT_HEAD(&card->state->card_head, card, entries);
1783 }
1784
1785 void AUD_remove_card (QEMUSoundCard *card)
1786 {
1787     QLIST_REMOVE (card, entries);
1788     g_free (card->name);
1789 }
1790
1791
1792 CaptureVoiceOut *AUD_add_capture(
1793     AudioState *s,
1794     struct audsettings *as,
1795     struct audio_capture_ops *ops,
1796     void *cb_opaque
1797     )
1798 {
1799     CaptureVoiceOut *cap;
1800     struct capture_callback *cb;
1801
1802     if (!s) {
1803         if (!legacy_config) {
1804             dolog("Capturing without setting an audiodev is deprecated\n");
1805         }
1806         s = audio_init(NULL, NULL);
1807     }
1808
1809     if (!audio_get_pdo_out(s->dev)->mixing_engine) {
1810         dolog("Can't capture with mixeng disabled\n");
1811         return NULL;
1812     }
1813
1814     if (audio_validate_settings (as)) {
1815         dolog ("Invalid settings were passed when trying to add capture\n");
1816         audio_print_settings (as);
1817         return NULL;
1818     }
1819
1820     cb = g_malloc0(sizeof(*cb));
1821     cb->ops = *ops;
1822     cb->opaque = cb_opaque;
1823
1824     cap = audio_pcm_capture_find_specific(s, as);
1825     if (cap) {
1826         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1827         return cap;
1828     }
1829     else {
1830         HWVoiceOut *hw;
1831         CaptureVoiceOut *cap;
1832
1833         cap = g_malloc0(sizeof(*cap));
1834
1835         hw = &cap->hw;
1836         hw->s = s;
1837         QLIST_INIT (&hw->sw_head);
1838         QLIST_INIT (&cap->cb_head);
1839
1840         /* XXX find a more elegant way */
1841         hw->samples = 4096 * 4;
1842         audio_pcm_hw_alloc_resources_out(hw);
1843
1844         audio_pcm_init_info (&hw->info, as);
1845
1846         cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);
1847
1848         if (hw->info.is_float) {
1849             hw->clip = mixeng_clip_float[hw->info.nchannels == 2];
1850         } else {
1851             hw->clip = mixeng_clip
1852                 [hw->info.nchannels == 2]
1853                 [hw->info.is_signed]
1854                 [hw->info.swap_endianness]
1855                 [audio_bits_to_index(hw->info.bits)];
1856         }
1857
1858         QLIST_INSERT_HEAD (&s->cap_head, cap, entries);
1859         QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
1860
1861         QLIST_FOREACH(hw, &s->hw_head_out, entries) {
1862             audio_attach_capture (hw);
1863         }
1864         return cap;
1865     }
1866 }
1867
1868 void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque)
1869 {
1870     struct capture_callback *cb;
1871
1872     for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {
1873         if (cb->opaque == cb_opaque) {
1874             cb->ops.destroy (cb_opaque);
1875             QLIST_REMOVE (cb, entries);
1876             g_free (cb);
1877
1878             if (!cap->cb_head.lh_first) {
1879                 SWVoiceOut *sw = cap->hw.sw_head.lh_first, *sw1;
1880
1881                 while (sw) {
1882                     SWVoiceCap *sc = (SWVoiceCap *) sw;
1883 #ifdef DEBUG_CAPTURE
1884                     dolog ("freeing %s\n", sw->name);
1885 #endif
1886
1887                     sw1 = sw->entries.le_next;
1888                     if (sw->rate) {
1889                         st_rate_stop (sw->rate);
1890                         sw->rate = NULL;
1891                     }
1892                     QLIST_REMOVE (sw, entries);
1893                     QLIST_REMOVE (sc, entries);
1894                     g_free (sc);
1895                     sw = sw1;
1896                 }
1897                 QLIST_REMOVE (cap, entries);
1898                 g_free (cap->hw.mix_buf);
1899                 g_free (cap->buf);
1900                 g_free (cap);
1901             }
1902             return;
1903         }
1904     }
1905 }
1906
1907 void AUD_set_volume_out (SWVoiceOut *sw, int mute, uint8_t lvol, uint8_t rvol)
1908 {
1909     Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
1910     audio_set_volume_out(sw, &vol);
1911 }
1912
1913 void audio_set_volume_out(SWVoiceOut *sw, Volume *vol)
1914 {
1915     if (sw) {
1916         HWVoiceOut *hw = sw->hw;
1917
1918         sw->vol.mute = vol->mute;
1919         sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
1920         sw->vol.r = nominal_volume.l * vol->vol[vol->channels > 1 ? 1 : 0] /
1921             255;
1922
1923         if (hw->pcm_ops->volume_out) {
1924             hw->pcm_ops->volume_out(hw, vol);
1925         }
1926     }
1927 }
1928
1929 void AUD_set_volume_in (SWVoiceIn *sw, int mute, uint8_t lvol, uint8_t rvol)
1930 {
1931     Volume vol = { .mute = mute, .channels = 2, .vol = { lvol, rvol } };
1932     audio_set_volume_in(sw, &vol);
1933 }
1934
1935 void audio_set_volume_in(SWVoiceIn *sw, Volume *vol)
1936 {
1937     if (sw) {
1938         HWVoiceIn *hw = sw->hw;
1939
1940         sw->vol.mute = vol->mute;
1941         sw->vol.l = nominal_volume.l * vol->vol[0] / 255;
1942         sw->vol.r = nominal_volume.r * vol->vol[vol->channels > 1 ? 1 : 0] /
1943             255;
1944
1945         if (hw->pcm_ops->volume_in) {
1946             hw->pcm_ops->volume_in(hw, vol);
1947         }
1948     }
1949 }
1950
1951 void audio_create_pdos(Audiodev *dev)
1952 {
1953     switch (dev->driver) {
1954 #define CASE(DRIVER, driver, pdo_name)                              \
1955     case AUDIODEV_DRIVER_##DRIVER:                                  \
1956         if (!dev->u.driver.has_in) {                                \
1957             dev->u.driver.in = g_malloc0(                           \
1958                 sizeof(Audiodev##pdo_name##PerDirectionOptions));   \
1959             dev->u.driver.has_in = true;                            \
1960         }                                                           \
1961         if (!dev->u.driver.has_out) {                               \
1962             dev->u.driver.out = g_malloc0(                          \
1963                 sizeof(Audiodev##pdo_name##PerDirectionOptions));   \
1964             dev->u.driver.has_out = true;                           \
1965         }                                                           \
1966         break
1967
1968         CASE(NONE, none, );
1969         CASE(ALSA, alsa, Alsa);
1970         CASE(COREAUDIO, coreaudio, Coreaudio);
1971         CASE(DSOUND, dsound, );
1972         CASE(JACK, jack, Jack);
1973         CASE(OSS, oss, Oss);
1974         CASE(PA, pa, Pa);
1975         CASE(SDL, sdl, );
1976         CASE(SPICE, spice, );
1977         CASE(WAV, wav, );
1978
1979     case AUDIODEV_DRIVER__MAX:
1980         abort();
1981     };
1982 }
1983
1984 static void audio_validate_per_direction_opts(
1985     AudiodevPerDirectionOptions *pdo, Error **errp)
1986 {
1987     if (!pdo->has_mixing_engine) {
1988         pdo->has_mixing_engine = true;
1989         pdo->mixing_engine = true;
1990     }
1991     if (!pdo->has_fixed_settings) {
1992         pdo->has_fixed_settings = true;
1993         pdo->fixed_settings = pdo->mixing_engine;
1994     }
1995     if (!pdo->fixed_settings &&
1996         (pdo->has_frequency || pdo->has_channels || pdo->has_format)) {
1997         error_setg(errp,
1998                    "You can't use frequency, channels or format with fixed-settings=off");
1999         return;
2000     }
2001     if (!pdo->mixing_engine && pdo->fixed_settings) {
2002         error_setg(errp, "You can't use fixed-settings without mixeng");
2003         return;
2004     }
2005
2006     if (!pdo->has_frequency) {
2007         pdo->has_frequency = true;
2008         pdo->frequency = 44100;
2009     }
2010     if (!pdo->has_channels) {
2011         pdo->has_channels = true;
2012         pdo->channels = 2;
2013     }
2014     if (!pdo->has_voices) {
2015         pdo->has_voices = true;
2016         pdo->voices = pdo->mixing_engine ? 1 : INT_MAX;
2017     }
2018     if (!pdo->has_format) {
2019         pdo->has_format = true;
2020         pdo->format = AUDIO_FORMAT_S16;
2021     }
2022 }
2023
2024 static void audio_validate_opts(Audiodev *dev, Error **errp)
2025 {
2026     Error *err = NULL;
2027
2028     audio_create_pdos(dev);
2029
2030     audio_validate_per_direction_opts(audio_get_pdo_in(dev), &err);
2031     if (err) {
2032         error_propagate(errp, err);
2033         return;
2034     }
2035
2036     audio_validate_per_direction_opts(audio_get_pdo_out(dev), &err);
2037     if (err) {
2038         error_propagate(errp, err);
2039         return;
2040     }
2041
2042     if (!dev->has_timer_period) {
2043         dev->has_timer_period = true;
2044         dev->timer_period = 10000; /* 100Hz -> 10ms */
2045     }
2046 }
2047
2048 void audio_parse_option(const char *opt)
2049 {
2050     AudiodevListEntry *e;
2051     Audiodev *dev = NULL;
2052
2053     Visitor *v = qobject_input_visitor_new_str(opt, "driver", &error_fatal);
2054     visit_type_Audiodev(v, NULL, &dev, &error_fatal);
2055     visit_free(v);
2056
2057     audio_validate_opts(dev, &error_fatal);
2058
2059     e = g_malloc0(sizeof(AudiodevListEntry));
2060     e->dev = dev;
2061     QSIMPLEQ_INSERT_TAIL(&audiodevs, e, next);
2062 }
2063
2064 void audio_init_audiodevs(void)
2065 {
2066     AudiodevListEntry *e;
2067
2068     QSIMPLEQ_FOREACH(e, &audiodevs, next) {
2069         audio_init(e->dev, NULL);
2070     }
2071 }
2072
2073 audsettings audiodev_to_audsettings(AudiodevPerDirectionOptions *pdo)
2074 {
2075     return (audsettings) {
2076         .freq = pdo->frequency,
2077         .nchannels = pdo->channels,
2078         .fmt = pdo->format,
2079         .endianness = AUDIO_HOST_ENDIANNESS,
2080     };
2081 }
2082
2083 int audioformat_bytes_per_sample(AudioFormat fmt)
2084 {
2085     switch (fmt) {
2086     case AUDIO_FORMAT_U8:
2087     case AUDIO_FORMAT_S8:
2088         return 1;
2089
2090     case AUDIO_FORMAT_U16:
2091     case AUDIO_FORMAT_S16:
2092         return 2;
2093
2094     case AUDIO_FORMAT_U32:
2095     case AUDIO_FORMAT_S32:
2096     case AUDIO_FORMAT_F32:
2097         return 4;
2098
2099     case AUDIO_FORMAT__MAX:
2100         ;
2101     }
2102     abort();
2103 }
2104
2105
2106 /* frames = freq * usec / 1e6 */
2107 int audio_buffer_frames(AudiodevPerDirectionOptions *pdo,
2108                         audsettings *as, int def_usecs)
2109 {
2110     uint64_t usecs = pdo->has_buffer_length ? pdo->buffer_length : def_usecs;
2111     return (as->freq * usecs + 500000) / 1000000;
2112 }
2113
2114 /* samples = channels * frames = channels * freq * usec / 1e6 */
2115 int audio_buffer_samples(AudiodevPerDirectionOptions *pdo,
2116                          audsettings *as, int def_usecs)
2117 {
2118     return as->nchannels * audio_buffer_frames(pdo, as, def_usecs);
2119 }
2120
2121 /*
2122  * bytes = bytes_per_sample * samples =
2123  *     bytes_per_sample * channels * freq * usec / 1e6
2124  */
2125 int audio_buffer_bytes(AudiodevPerDirectionOptions *pdo,
2126                        audsettings *as, int def_usecs)
2127 {
2128     return audio_buffer_samples(pdo, as, def_usecs) *
2129         audioformat_bytes_per_sample(as->fmt);
2130 }
2131
2132 AudioState *audio_state_by_name(const char *name)
2133 {
2134     AudioState *s;
2135     QTAILQ_FOREACH(s, &audio_states, list) {
2136         assert(s->dev);
2137         if (strcmp(name, s->dev->id) == 0) {
2138             return s;
2139         }
2140     }
2141     return NULL;
2142 }
2143
2144 const char *audio_get_id(QEMUSoundCard *card)
2145 {
2146     if (card->state) {
2147         assert(card->state->dev);
2148         return card->state->dev->id;
2149     } else {
2150         return "";
2151     }
2152 }
2153
2154 void audio_rate_start(RateCtl *rate)
2155 {
2156     memset(rate, 0, sizeof(RateCtl));
2157     rate->start_ticks = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
2158 }
2159
2160 size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate,
2161                             size_t bytes_avail)
2162 {
2163     int64_t now;
2164     int64_t ticks;
2165     int64_t bytes;
2166     int64_t samples;
2167     size_t ret;
2168
2169     now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
2170     ticks = now - rate->start_ticks;
2171     bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);
2172     samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;
2173     if (samples < 0 || samples > 65536) {
2174         AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);
2175         audio_rate_start(rate);
2176         samples = 0;
2177     }
2178
2179     ret = MIN(samples * info->bytes_per_frame, bytes_avail);
2180     rate->bytes_sent += ret;
2181     return ret;
2182 }