hw/audio/hda-codec.c

   1 /*
   2  * Copyright (C) 2010 Red Hat, Inc.
   3  *
   4  * written by Gerd Hoffmann <kraxel@redhat.com>
   5  *
   6  * This program is free software; you can redistribute it and/or
   7  * modify it under the terms of the GNU General Public License as
   8  * published by the Free Software Foundation; either version 2 or
   9  * (at your option) version 3 of the License.
  10  *
  11  * This program is distributed in the hope that it will be useful,
  12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  14  * GNU General Public License for more details.
  15  *
  16  * You should have received a copy of the GNU General Public License
  17  * along with this program; if not, see <http://www.gnu.org/licenses/>.
  18  */
  19
  20 #include "qemu/osdep.h"
  21 #include "hw/hw.h"
  22 #include "hw/pci/pci.h"
  23 #include "intel-hda.h"
  24 #include "qemu/module.h"
  25 #include "intel-hda-defs.h"
  26 #include "audio/audio.h"
  27 #include "trace.h"
  28
  29 /* -------------------------------------------------------------------------- */
  30
  31 typedef struct desc_param {
  32     uint32_t id;
  33     uint32_t val;
  34 } desc_param;
  35
  36 typedef struct desc_node {
  37     uint32_t nid;
  38     const char *name;
  39     const desc_param *params;
  40     uint32_t nparams;
  41     uint32_t config;
  42     uint32_t pinctl;
  43     uint32_t *conn;
  44     uint32_t stindex;
  45 } desc_node;
  46
  47 typedef struct desc_codec {
  48     const char *name;
  49     uint32_t iid;
  50     const desc_node *nodes;
  51     uint32_t nnodes;
  52 } desc_codec;
  53
  54 static const desc_param* hda_codec_find_param(const desc_node *node, uint32_t id)
  55 {
  56     int i;
  57
  58     for (i = 0; i < node->nparams; i++) {
  59         if (node->params[i].id == id) {
  60             return &node->params[i];
  61         }
  62     }
  63     return NULL;
  64 }
  65
  66 static const desc_node* hda_codec_find_node(const desc_codec *codec, uint32_t nid)
  67 {
  68     int i;
  69
  70     for (i = 0; i < codec->nnodes; i++) {
  71         if (codec->nodes[i].nid == nid) {
  72             return &codec->nodes[i];
  73         }
  74     }
  75     return NULL;
  76 }
  77
  78 static void hda_codec_parse_fmt(uint32_t format, struct audsettings *as)
  79 {
  80     if (format & AC_FMT_TYPE_NON_PCM) {
  81         return;
  82     }
  83
  84     as->freq = (format & AC_FMT_BASE_44K) ? 44100 : 48000;
  85
  86     switch ((format & AC_FMT_MULT_MASK) >> AC_FMT_MULT_SHIFT) {
  87     case 1: as->freq *= 2; break;
  88     case 2: as->freq *= 3; break;
  89     case 3: as->freq *= 4; break;
  90     }
  91
  92     switch ((format & AC_FMT_DIV_MASK) >> AC_FMT_DIV_SHIFT) {
  93     case 1: as->freq /= 2; break;
  94     case 2: as->freq /= 3; break;
  95     case 3: as->freq /= 4; break;
  96     case 4: as->freq /= 5; break;
  97     case 5: as->freq /= 6; break;
  98     case 6: as->freq /= 7; break;
  99     case 7: as->freq /= 8; break;
 100     }
 101
 102     switch (format & AC_FMT_BITS_MASK) {
 103     case AC_FMT_BITS_8:  as->fmt = AUDIO_FORMAT_S8;  break;
 104     case AC_FMT_BITS_16: as->fmt = AUDIO_FORMAT_S16; break;
 105     case AC_FMT_BITS_32: as->fmt = AUDIO_FORMAT_S32; break;
 106     }
 107
 108     as->nchannels = ((format & AC_FMT_CHAN_MASK) >> AC_FMT_CHAN_SHIFT) + 1;
 109 }
 110
 111 /* -------------------------------------------------------------------------- */
 112 /*
 113  * HDA codec descriptions
 114  */
 115
 116 /* some defines */
 117
 118 #define QEMU_HDA_ID_VENDOR  0x1af4
 119 #define QEMU_HDA_PCM_FORMATS (AC_SUPPCM_BITS_16 |       \
 120                               0x1fc /* 16 -> 96 kHz */)
 121 #define QEMU_HDA_AMP_NONE    (0)
 122 #define QEMU_HDA_AMP_STEPS   0x4a
 123
 124 #define   PARAM mixemu
 125 #define   HDA_MIXER
 126 #include "hda-codec-common.h"
 127
 128 #define   PARAM nomixemu
 129 #include  "hda-codec-common.h"
 130
 131 #define HDA_TIMER_TICKS (SCALE_MS)
 132 #define B_SIZE sizeof(st->buf)
 133 #define B_MASK (sizeof(st->buf) - 1)
 134
 135 /* -------------------------------------------------------------------------- */
 136
 137 static const char *fmt2name[] = {
 138     [ AUDIO_FORMAT_U8  ] = "PCM-U8",
 139     [ AUDIO_FORMAT_S8  ] = "PCM-S8",
 140     [ AUDIO_FORMAT_U16 ] = "PCM-U16",
 141     [ AUDIO_FORMAT_S16 ] = "PCM-S16",
 142     [ AUDIO_FORMAT_U32 ] = "PCM-U32",
 143     [ AUDIO_FORMAT_S32 ] = "PCM-S32",
 144 };
 145
 146 typedef struct HDAAudioState HDAAudioState;
 147 typedef struct HDAAudioStream HDAAudioStream;
 148
 149 struct HDAAudioStream {
 150     HDAAudioState *state;
 151     const desc_node *node;
 152     bool output, running;
 153     uint32_t stream;
 154     uint32_t channel;
 155     uint32_t format;
 156     uint32_t gain_left, gain_right;
 157     bool mute_left, mute_right;
 158     struct audsettings as;
 159     union {
 160         SWVoiceIn *in;
 161         SWVoiceOut *out;
 162     } voice;
 163     uint8_t compat_buf[HDA_BUFFER_SIZE];
 164     uint32_t compat_bpos;
 165     uint8_t buf[8192]; /* size must be power of two */
 166     int64_t rpos;
 167     int64_t wpos;
 168     QEMUTimer *buft;
 169     int64_t buft_start;
 170 };
 171
 172 #define TYPE_HDA_AUDIO "hda-audio"
 173 #define HDA_AUDIO(obj) OBJECT_CHECK(HDAAudioState, (obj), TYPE_HDA_AUDIO)
 174
 175 struct HDAAudioState {
 176     HDACodecDevice hda;
 177     const char *name;
 178
 179     QEMUSoundCard card;
 180     const desc_codec *desc;
 181     HDAAudioStream st[4];
 182     bool running_compat[16];
 183     bool running_real[2 * 16];
 184
 185     /* properties */
 186     uint32_t debug;
 187     bool     mixer;
 188     bool     use_timer;
 189 };
 190
 191 static inline int64_t hda_bytes_per_second(HDAAudioStream *st)
 192 {
 193     return 2LL * st->as.nchannels * st->as.freq;
 194 }
 195
 196 static inline void hda_timer_sync_adjust(HDAAudioStream *st, int64_t target_pos)
 197 {
 198     int64_t limit = B_SIZE / 8;
 199     int64_t corr = 0;
 200
 201     if (target_pos > limit) {
 202         corr = HDA_TIMER_TICKS;
 203     }
 204     if (target_pos < -limit) {
 205         corr = -HDA_TIMER_TICKS;
 206     }
 207     if (target_pos < -(2 * limit)) {
 208         corr = -(4 * HDA_TIMER_TICKS);
 209     }
 210     if (corr == 0) {
 211         return;
 212     }
 213
 214     trace_hda_audio_adjust(st->node->name, target_pos);
 215     st->buft_start += corr;
 216 }
 217
 218 static void hda_audio_input_timer(void *opaque)
 219 {
 220     HDAAudioStream *st = opaque;
 221
 222     int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 223
 224     int64_t buft_start = st->buft_start;
 225     int64_t wpos = st->wpos;
 226     int64_t rpos = st->rpos;
 227
 228     int64_t wanted_rpos = hda_bytes_per_second(st) * (now - buft_start)
 229                           / NANOSECONDS_PER_SECOND;
 230     wanted_rpos &= -4; /* IMPORTANT! clip to frames */
 231
 232     if (wanted_rpos <= rpos) {
 233         /* we already transmitted the data */
 234         goto out_timer;
 235     }
 236
 237     int64_t to_transfer = audio_MIN(wpos - rpos, wanted_rpos - rpos);
 238     while (to_transfer) {
 239         uint32_t start = (rpos & B_MASK);
 240         uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
 241         int rc = hda_codec_xfer(
 242                 &st->state->hda, st->stream, false, st->buf + start, chunk);
 243         if (!rc) {
 244             break;
 245         }
 246         rpos += chunk;
 247         to_transfer -= chunk;
 248         st->rpos += chunk;
 249     }
 250
 251 out_timer:
 252
 253     if (st->running) {
 254         timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
 255     }
 256 }
 257
 258 static void hda_audio_input_cb(void *opaque, int avail)
 259 {
 260     HDAAudioStream *st = opaque;
 261
 262     int64_t wpos = st->wpos;
 263     int64_t rpos = st->rpos;
 264
 265     int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), avail);
 266
 267     hda_timer_sync_adjust(st, -((wpos - rpos) + to_transfer - (B_SIZE >> 1)));
 268
 269     while (to_transfer) {
 270         uint32_t start = (uint32_t) (wpos & B_MASK);
 271         uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
 272         uint32_t read = AUD_read(st->voice.in, st->buf + start, chunk);
 273         wpos += read;
 274         to_transfer -= read;
 275         st->wpos += read;
 276         if (chunk != read) {
 277             break;
 278         }
 279     }
 280 }
 281
 282 static void hda_audio_output_timer(void *opaque)
 283 {
 284     HDAAudioStream *st = opaque;
 285
 286     int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 287
 288     int64_t buft_start = st->buft_start;
 289     int64_t wpos = st->wpos;
 290     int64_t rpos = st->rpos;
 291
 292     int64_t wanted_wpos = hda_bytes_per_second(st) * (now - buft_start)
 293                           / NANOSECONDS_PER_SECOND;
 294     wanted_wpos &= -4; /* IMPORTANT! clip to frames */
 295
 296     if (wanted_wpos <= wpos) {
 297         /* we already received the data */
 298         goto out_timer;
 299     }
 300
 301     int64_t to_transfer = audio_MIN(B_SIZE - (wpos - rpos), wanted_wpos - wpos);
 302     while (to_transfer) {
 303         uint32_t start = (wpos & B_MASK);
 304         uint32_t chunk = audio_MIN(B_SIZE - start, to_transfer);
 305         int rc = hda_codec_xfer(
 306                 &st->state->hda, st->stream, true, st->buf + start, chunk);
 307         if (!rc) {
 308             break;
 309         }
 310         wpos += chunk;
 311         to_transfer -= chunk;
 312         st->wpos += chunk;
 313     }
 314
 315 out_timer:
 316
 317     if (st->running) {
 318         timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
 319     }
 320 }
 321
 322 static void hda_audio_output_cb(void *opaque, int avail)
 323 {
 324     HDAAudioStream *st = opaque;
 325
 326     int64_t wpos = st->wpos;
 327     int64_t rpos = st->rpos;
 328
 329     int64_t to_transfer = audio_MIN(wpos - rpos, avail);
 330
 331     if (wpos - rpos == B_SIZE) {
 332         /* drop buffer, reset timer adjust */
 333         st->rpos = 0;
 334         st->wpos = 0;
 335         st->buft_start = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 336         trace_hda_audio_overrun(st->node->name);
 337         return;
 338     }
 339
 340     hda_timer_sync_adjust(st, (wpos - rpos) - to_transfer - (B_SIZE >> 1));
 341
 342     while (to_transfer) {
 343         uint32_t start = (uint32_t) (rpos & B_MASK);
 344         uint32_t chunk = (uint32_t) audio_MIN(B_SIZE - start, to_transfer);
 345         uint32_t written = AUD_write(st->voice.out, st->buf + start, chunk);
 346         rpos += written;
 347         to_transfer -= written;
 348         st->rpos += written;
 349         if (chunk != written) {
 350             break;
 351         }
 352     }
 353 }
 354
 355 static void hda_audio_compat_input_cb(void *opaque, int avail)
 356 {
 357     HDAAudioStream *st = opaque;
 358     int recv = 0;
 359     int len;
 360     bool rc;
 361
 362     while (avail - recv >= sizeof(st->compat_buf)) {
 363         if (st->compat_bpos != sizeof(st->compat_buf)) {
 364             len = AUD_read(st->voice.in, st->compat_buf + st->compat_bpos,
 365                            sizeof(st->compat_buf) - st->compat_bpos);
 366             st->compat_bpos += len;
 367             recv += len;
 368             if (st->compat_bpos != sizeof(st->compat_buf)) {
 369                 break;
 370             }
 371         }
 372         rc = hda_codec_xfer(&st->state->hda, st->stream, false,
 373                             st->compat_buf, sizeof(st->compat_buf));
 374         if (!rc) {
 375             break;
 376         }
 377         st->compat_bpos = 0;
 378     }
 379 }
 380
 381 static void hda_audio_compat_output_cb(void *opaque, int avail)
 382 {
 383     HDAAudioStream *st = opaque;
 384     int sent = 0;
 385     int len;
 386     bool rc;
 387
 388     while (avail - sent >= sizeof(st->compat_buf)) {
 389         if (st->compat_bpos == sizeof(st->compat_buf)) {
 390             rc = hda_codec_xfer(&st->state->hda, st->stream, true,
 391                                 st->compat_buf, sizeof(st->compat_buf));
 392             if (!rc) {
 393                 break;
 394             }
 395             st->compat_bpos = 0;
 396         }
 397         len = AUD_write(st->voice.out, st->compat_buf + st->compat_bpos,
 398                         sizeof(st->compat_buf) - st->compat_bpos);
 399         st->compat_bpos += len;
 400         sent += len;
 401         if (st->compat_bpos != sizeof(st->compat_buf)) {
 402             break;
 403         }
 404     }
 405 }
 406
 407 static void hda_audio_set_running(HDAAudioStream *st, bool running)
 408 {
 409     if (st->node == NULL) {
 410         return;
 411     }
 412     if (st->running == running) {
 413         return;
 414     }
 415     st->running = running;
 416     trace_hda_audio_running(st->node->name, st->stream, st->running);
 417     if (st->state->use_timer) {
 418         if (running) {
 419             int64_t now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);
 420             st->rpos = 0;
 421             st->wpos = 0;
 422             st->buft_start = now;
 423             timer_mod_anticipate_ns(st->buft, now + HDA_TIMER_TICKS);
 424         } else {
 425             timer_del(st->buft);
 426         }
 427     }
 428     if (st->output) {
 429         AUD_set_active_out(st->voice.out, st->running);
 430     } else {
 431         AUD_set_active_in(st->voice.in, st->running);
 432     }
 433 }
 434
 435 static void hda_audio_set_amp(HDAAudioStream *st)
 436 {
 437     bool muted;
 438     uint32_t left, right;
 439
 440     if (st->node == NULL) {
 441         return;
 442     }
 443
 444     muted = st->mute_left && st->mute_right;
 445     left  = st->mute_left  ? 0 : st->gain_left;
 446     right = st->mute_right ? 0 : st->gain_right;
 447
 448     left = left * 255 / QEMU_HDA_AMP_STEPS;
 449     right = right * 255 / QEMU_HDA_AMP_STEPS;
 450
 451     if (!st->state->mixer) {
 452         return;
 453     }
 454     if (st->output) {
 455         AUD_set_volume_out(st->voice.out, muted, left, right);
 456     } else {
 457         AUD_set_volume_in(st->voice.in, muted, left, right);
 458     }
 459 }
 460
 461 static void hda_audio_setup(HDAAudioStream *st)
 462 {
 463     bool use_timer = st->state->use_timer;
 464     audio_callback_fn cb;
 465
 466     if (st->node == NULL) {
 467         return;
 468     }
 469
 470     trace_hda_audio_format(st->node->name, st->as.nchannels,
 471                            fmt2name[st->as.fmt], st->as.freq);
 472
 473     if (st->output) {
 474         if (use_timer) {
 475             cb = hda_audio_output_cb;
 476             st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
 477                                     hda_audio_output_timer, st);
 478         } else {
 479             cb = hda_audio_compat_output_cb;
 480         }
 481         st->voice.out = AUD_open_out(&st->state->card, st->voice.out,
 482                                      st->node->name, st, cb, &st->as);
 483     } else {
 484         if (use_timer) {
 485             cb = hda_audio_input_cb;
 486             st->buft = timer_new_ns(QEMU_CLOCK_VIRTUAL,
 487                                     hda_audio_input_timer, st);
 488         } else {
 489             cb = hda_audio_compat_input_cb;
 490         }
 491         st->voice.in = AUD_open_in(&st->state->card, st->voice.in,
 492                                    st->node->name, st, cb, &st->as);
 493     }
 494 }
 495
 496 static void hda_audio_command(HDACodecDevice *hda, uint32_t nid, uint32_t data)
 497 {
 498     HDAAudioState *a = HDA_AUDIO(hda);
 499     HDAAudioStream *st;
 500     const desc_node *node = NULL;
 501     const desc_param *param;
 502     uint32_t verb, payload, response, count, shift;
 503
 504     if ((data & 0x70000) == 0x70000) {
 505         /* 12/8 id/payload */
 506         verb = (data >> 8) & 0xfff;
 507         payload = data & 0x00ff;
 508     } else {
 509         /* 4/16 id/payload */
 510         verb = (data >> 8) & 0xf00;
 511         payload = data & 0xffff;
 512     }
 513
 514     node = hda_codec_find_node(a->desc, nid);
 515     if (node == NULL) {
 516         goto fail;
 517     }
 518     dprint(a, 2, "%s: nid %d (%s), verb 0x%x, payload 0x%x\n",
 519            __func__, nid, node->name, verb, payload);
 520
 521     switch (verb) {
 522     /* all nodes */
 523     case AC_VERB_PARAMETERS:
 524         param = hda_codec_find_param(node, payload);
 525         if (param == NULL) {
 526             goto fail;
 527         }
 528         hda_codec_response(hda, true, param->val);
 529         break;
 530     case AC_VERB_GET_SUBSYSTEM_ID:
 531         hda_codec_response(hda, true, a->desc->iid);
 532         break;
 533
 534     /* all functions */
 535     case AC_VERB_GET_CONNECT_LIST:
 536         param = hda_codec_find_param(node, AC_PAR_CONNLIST_LEN);
 537         count = param ? param->val : 0;
 538         response = 0;
 539         shift = 0;
 540         while (payload < count && shift < 32) {
 541             response |= node->conn[payload] << shift;
 542             payload++;
 543             shift += 8;
 544         }
 545         hda_codec_response(hda, true, response);
 546         break;
 547
 548     /* pin widget */
 549     case AC_VERB_GET_CONFIG_DEFAULT:
 550         hda_codec_response(hda, true, node->config);
 551         break;
 552     case AC_VERB_GET_PIN_WIDGET_CONTROL:
 553         hda_codec_response(hda, true, node->pinctl);
 554         break;
 555     case AC_VERB_SET_PIN_WIDGET_CONTROL:
 556         if (node->pinctl != payload) {
 557             dprint(a, 1, "unhandled pin control bit\n");
 558         }
 559         hda_codec_response(hda, true, 0);
 560         break;
 561
 562     /* audio in/out widget */
 563     case AC_VERB_SET_CHANNEL_STREAMID:
 564         st = a->st + node->stindex;
 565         if (st->node == NULL) {
 566             goto fail;
 567         }
 568         hda_audio_set_running(st, false);
 569         st->stream = (payload >> 4) & 0x0f;
 570         st->channel = payload & 0x0f;
 571         dprint(a, 2, "%s: stream %d, channel %d\n",
 572                st->node->name, st->stream, st->channel);
 573         hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
 574         hda_codec_response(hda, true, 0);
 575         break;
 576     case AC_VERB_GET_CONV:
 577         st = a->st + node->stindex;
 578         if (st->node == NULL) {
 579             goto fail;
 580         }
 581         response = st->stream << 4 | st->channel;
 582         hda_codec_response(hda, true, response);
 583         break;
 584     case AC_VERB_SET_STREAM_FORMAT:
 585         st = a->st + node->stindex;
 586         if (st->node == NULL) {
 587             goto fail;
 588         }
 589         st->format = payload;
 590         hda_codec_parse_fmt(st->format, &st->as);
 591         hda_audio_setup(st);
 592         hda_codec_response(hda, true, 0);
 593         break;
 594     case AC_VERB_GET_STREAM_FORMAT:
 595         st = a->st + node->stindex;
 596         if (st->node == NULL) {
 597             goto fail;
 598         }
 599         hda_codec_response(hda, true, st->format);
 600         break;
 601     case AC_VERB_GET_AMP_GAIN_MUTE:
 602         st = a->st + node->stindex;
 603         if (st->node == NULL) {
 604             goto fail;
 605         }
 606         if (payload & AC_AMP_GET_LEFT) {
 607             response = st->gain_left | (st->mute_left ? AC_AMP_MUTE : 0);
 608         } else {
 609             response = st->gain_right | (st->mute_right ? AC_AMP_MUTE : 0);
 610         }
 611         hda_codec_response(hda, true, response);
 612         break;
 613     case AC_VERB_SET_AMP_GAIN_MUTE:
 614         st = a->st + node->stindex;
 615         if (st->node == NULL) {
 616             goto fail;
 617         }
 618         dprint(a, 1, "amp (%s): %s%s%s%s index %d  gain %3d %s\n",
 619                st->node->name,
 620                (payload & AC_AMP_SET_OUTPUT) ? "o" : "-",
 621                (payload & AC_AMP_SET_INPUT)  ? "i" : "-",
 622                (payload & AC_AMP_SET_LEFT)   ? "l" : "-",
 623                (payload & AC_AMP_SET_RIGHT)  ? "r" : "-",
 624                (payload & AC_AMP_SET_INDEX) >> AC_AMP_SET_INDEX_SHIFT,
 625                (payload & AC_AMP_GAIN),
 626                (payload & AC_AMP_MUTE) ? "muted" : "");
 627         if (payload & AC_AMP_SET_LEFT) {
 628             st->gain_left = payload & AC_AMP_GAIN;
 629             st->mute_left = payload & AC_AMP_MUTE;
 630         }
 631         if (payload & AC_AMP_SET_RIGHT) {
 632             st->gain_right = payload & AC_AMP_GAIN;
 633             st->mute_right = payload & AC_AMP_MUTE;
 634         }
 635         hda_audio_set_amp(st);
 636         hda_codec_response(hda, true, 0);
 637         break;
 638
 639     /* not supported */
 640     case AC_VERB_SET_POWER_STATE:
 641     case AC_VERB_GET_POWER_STATE:
 642     case AC_VERB_GET_SDI_SELECT:
 643         hda_codec_response(hda, true, 0);
 644         break;
 645     default:
 646         goto fail;
 647     }
 648     return;
 649
 650 fail:
 651     dprint(a, 1, "%s: not handled: nid %d (%s), verb 0x%x, payload 0x%x\n",
 652            __func__, nid, node ? node->name : "?", verb, payload);
 653     hda_codec_response(hda, true, 0);
 654 }
 655
 656 static void hda_audio_stream(HDACodecDevice *hda, uint32_t stnr, bool running, bool output)
 657 {
 658     HDAAudioState *a = HDA_AUDIO(hda);
 659     int s;
 660
 661     a->running_compat[stnr] = running;
 662     a->running_real[output * 16 + stnr] = running;
 663     for (s = 0; s < ARRAY_SIZE(a->st); s++) {
 664         if (a->st[s].node == NULL) {
 665             continue;
 666         }
 667         if (a->st[s].output != output) {
 668             continue;
 669         }
 670         if (a->st[s].stream != stnr) {
 671             continue;
 672         }
 673         hda_audio_set_running(&a->st[s], running);
 674     }
 675 }
 676
 677 static int hda_audio_init(HDACodecDevice *hda, const struct desc_codec *desc)
 678 {
 679     HDAAudioState *a = HDA_AUDIO(hda);
 680     HDAAudioStream *st;
 681     const desc_node *node;
 682     const desc_param *param;
 683     uint32_t i, type;
 684
 685     a->desc = desc;
 686     a->name = object_get_typename(OBJECT(a));
 687     dprint(a, 1, "%s: cad %d\n", __func__, a->hda.cad);
 688
 689     AUD_register_card("hda", &a->card);
 690     for (i = 0; i < a->desc->nnodes; i++) {
 691         node = a->desc->nodes + i;
 692         param = hda_codec_find_param(node, AC_PAR_AUDIO_WIDGET_CAP);
 693         if (param == NULL) {
 694             continue;
 695         }
 696         type = (param->val & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
 697         switch (type) {
 698         case AC_WID_AUD_OUT:
 699         case AC_WID_AUD_IN:
 700             assert(node->stindex < ARRAY_SIZE(a->st));
 701             st = a->st + node->stindex;
 702             st->state = a;
 703             st->node = node;
 704             if (type == AC_WID_AUD_OUT) {
 705                 /* unmute output by default */
 706                 st->gain_left = QEMU_HDA_AMP_STEPS;
 707                 st->gain_right = QEMU_HDA_AMP_STEPS;
 708                 st->compat_bpos = sizeof(st->compat_buf);
 709                 st->output = true;
 710             } else {
 711                 st->output = false;
 712             }
 713             st->format = AC_FMT_TYPE_PCM | AC_FMT_BITS_16 |
 714                 (1 << AC_FMT_CHAN_SHIFT);
 715             hda_codec_parse_fmt(st->format, &st->as);
 716             hda_audio_setup(st);
 717             break;
 718         }
 719     }
 720     return 0;
 721 }
 722
 723 static void hda_audio_exit(HDACodecDevice *hda)
 724 {
 725     HDAAudioState *a = HDA_AUDIO(hda);
 726     HDAAudioStream *st;
 727     int i;
 728
 729     dprint(a, 1, "%s\n", __func__);
 730     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
 731         st = a->st + i;
 732         if (st->node == NULL) {
 733             continue;
 734         }
 735         if (a->use_timer) {
 736             timer_del(st->buft);
 737         }
 738         if (st->output) {
 739             AUD_close_out(&a->card, st->voice.out);
 740         } else {
 741             AUD_close_in(&a->card, st->voice.in);
 742         }
 743     }
 744     AUD_remove_card(&a->card);
 745 }
 746
 747 static int hda_audio_post_load(void *opaque, int version)
 748 {
 749     HDAAudioState *a = opaque;
 750     HDAAudioStream *st;
 751     int i;
 752
 753     dprint(a, 1, "%s\n", __func__);
 754     if (version == 1) {
 755         /* assume running_compat[] is for output streams */
 756         for (i = 0; i < ARRAY_SIZE(a->running_compat); i++)
 757             a->running_real[16 + i] = a->running_compat[i];
 758     }
 759
 760     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
 761         st = a->st + i;
 762         if (st->node == NULL)
 763             continue;
 764         hda_codec_parse_fmt(st->format, &st->as);
 765         hda_audio_setup(st);
 766         hda_audio_set_amp(st);
 767         hda_audio_set_running(st, a->running_real[st->output * 16 + st->stream]);
 768     }
 769     return 0;
 770 }
 771
 772 static void hda_audio_reset(DeviceState *dev)
 773 {
 774     HDAAudioState *a = HDA_AUDIO(dev);
 775     HDAAudioStream *st;
 776     int i;
 777
 778     dprint(a, 1, "%s\n", __func__);
 779     for (i = 0; i < ARRAY_SIZE(a->st); i++) {
 780         st = a->st + i;
 781         if (st->node != NULL) {
 782             hda_audio_set_running(st, false);
 783         }
 784     }
 785 }
 786
 787 static bool vmstate_hda_audio_stream_buf_needed(void *opaque)
 788 {
 789     HDAAudioStream *st = opaque;
 790     return st->state && st->state->use_timer;
 791 }
 792
 793 static const VMStateDescription vmstate_hda_audio_stream_buf = {
 794     .name = "hda-audio-stream/buffer",
 795     .version_id = 1,
 796     .needed = vmstate_hda_audio_stream_buf_needed,
 797     .fields = (VMStateField[]) {
 798         VMSTATE_BUFFER(buf, HDAAudioStream),
 799         VMSTATE_INT64(rpos, HDAAudioStream),
 800         VMSTATE_INT64(wpos, HDAAudioStream),
 801         VMSTATE_TIMER_PTR(buft, HDAAudioStream),
 802         VMSTATE_INT64(buft_start, HDAAudioStream),
 803         VMSTATE_END_OF_LIST()
 804     }
 805 };
 806
 807 static const VMStateDescription vmstate_hda_audio_stream = {
 808     .name = "hda-audio-stream",
 809     .version_id = 1,
 810     .fields = (VMStateField[]) {
 811         VMSTATE_UINT32(stream, HDAAudioStream),
 812         VMSTATE_UINT32(channel, HDAAudioStream),
 813         VMSTATE_UINT32(format, HDAAudioStream),
 814         VMSTATE_UINT32(gain_left, HDAAudioStream),
 815         VMSTATE_UINT32(gain_right, HDAAudioStream),
 816         VMSTATE_BOOL(mute_left, HDAAudioStream),
 817         VMSTATE_BOOL(mute_right, HDAAudioStream),
 818         VMSTATE_UINT32(compat_bpos, HDAAudioStream),
 819         VMSTATE_BUFFER(compat_buf, HDAAudioStream),
 820         VMSTATE_END_OF_LIST()
 821     },
 822     .subsections = (const VMStateDescription * []) {
 823         &vmstate_hda_audio_stream_buf,
 824         NULL
 825     }
 826 };
 827
 828 static const VMStateDescription vmstate_hda_audio = {
 829     .name = "hda-audio",
 830     .version_id = 2,
 831     .post_load = hda_audio_post_load,
 832     .fields = (VMStateField[]) {
 833         VMSTATE_STRUCT_ARRAY(st, HDAAudioState, 4, 0,
 834                              vmstate_hda_audio_stream,
 835                              HDAAudioStream),
 836         VMSTATE_BOOL_ARRAY(running_compat, HDAAudioState, 16),
 837         VMSTATE_BOOL_ARRAY_V(running_real, HDAAudioState, 2 * 16, 2),
 838         VMSTATE_END_OF_LIST()
 839     }
 840 };
 841
 842 static Property hda_audio_properties[] = {
 843     DEFINE_PROP_UINT32("debug", HDAAudioState, debug,   0),
 844     DEFINE_PROP_BOOL("mixer", HDAAudioState, mixer,  true),
 845     DEFINE_PROP_BOOL("use-timer", HDAAudioState, use_timer,  true),
 846     DEFINE_PROP_END_OF_LIST(),
 847 };
 848
 849 static int hda_audio_init_output(HDACodecDevice *hda)
 850 {
 851     HDAAudioState *a = HDA_AUDIO(hda);
 852
 853     if (!a->mixer) {
 854         return hda_audio_init(hda, &output_nomixemu);
 855     } else {
 856         return hda_audio_init(hda, &output_mixemu);
 857     }
 858 }
 859
 860 static int hda_audio_init_duplex(HDACodecDevice *hda)
 861 {
 862     HDAAudioState *a = HDA_AUDIO(hda);
 863
 864     if (!a->mixer) {
 865         return hda_audio_init(hda, &duplex_nomixemu);
 866     } else {
 867         return hda_audio_init(hda, &duplex_mixemu);
 868     }
 869 }
 870
 871 static int hda_audio_init_micro(HDACodecDevice *hda)
 872 {
 873     HDAAudioState *a = HDA_AUDIO(hda);
 874
 875     if (!a->mixer) {
 876         return hda_audio_init(hda, &micro_nomixemu);
 877     } else {
 878         return hda_audio_init(hda, &micro_mixemu);
 879     }
 880 }
 881
 882 static void hda_audio_base_class_init(ObjectClass *klass, void *data)
 883 {
 884     DeviceClass *dc = DEVICE_CLASS(klass);
 885     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
 886
 887     k->exit = hda_audio_exit;
 888     k->command = hda_audio_command;
 889     k->stream = hda_audio_stream;
 890     set_bit(DEVICE_CATEGORY_SOUND, dc->categories);
 891     dc->reset = hda_audio_reset;
 892     dc->vmsd = &vmstate_hda_audio;
 893     dc->props = hda_audio_properties;
 894 }
 895
 896 static const TypeInfo hda_audio_info = {
 897     .name          = TYPE_HDA_AUDIO,
 898     .parent        = TYPE_HDA_CODEC_DEVICE,
 899     .class_init    = hda_audio_base_class_init,
 900     .abstract      = true,
 901 };
 902
 903 static void hda_audio_output_class_init(ObjectClass *klass, void *data)
 904 {
 905     DeviceClass *dc = DEVICE_CLASS(klass);
 906     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
 907
 908     k->init = hda_audio_init_output;
 909     dc->desc = "HDA Audio Codec, output-only (line-out)";
 910 }
 911
 912 static const TypeInfo hda_audio_output_info = {
 913     .name          = "hda-output",
 914     .parent        = TYPE_HDA_AUDIO,
 915     .instance_size = sizeof(HDAAudioState),
 916     .class_init    = hda_audio_output_class_init,
 917 };
 918
 919 static void hda_audio_duplex_class_init(ObjectClass *klass, void *data)
 920 {
 921     DeviceClass *dc = DEVICE_CLASS(klass);
 922     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
 923
 924     k->init = hda_audio_init_duplex;
 925     dc->desc = "HDA Audio Codec, duplex (line-out, line-in)";
 926 }
 927
 928 static const TypeInfo hda_audio_duplex_info = {
 929     .name          = "hda-duplex",
 930     .parent        = TYPE_HDA_AUDIO,
 931     .instance_size = sizeof(HDAAudioState),
 932     .class_init    = hda_audio_duplex_class_init,
 933 };
 934
 935 static void hda_audio_micro_class_init(ObjectClass *klass, void *data)
 936 {
 937     DeviceClass *dc = DEVICE_CLASS(klass);
 938     HDACodecDeviceClass *k = HDA_CODEC_DEVICE_CLASS(klass);
 939
 940     k->init = hda_audio_init_micro;
 941     dc->desc = "HDA Audio Codec, duplex (speaker, microphone)";
 942 }
 943
 944 static const TypeInfo hda_audio_micro_info = {
 945     .name          = "hda-micro",
 946     .parent        = TYPE_HDA_AUDIO,
 947     .instance_size = sizeof(HDAAudioState),
 948     .class_init    = hda_audio_micro_class_init,
 949 };
 950
 951 static void hda_audio_register_types(void)
 952 {
 953     type_register_static(&hda_audio_info);
 954     type_register_static(&hda_audio_output_info);
 955     type_register_static(&hda_audio_duplex_info);
 956     type_register_static(&hda_audio_micro_info);
 957 }
 958
 959 type_init(hda_audio_register_types)